| 00:01.21 | fugitivo | I'd never used another port, but the syntax you're using seems correct |
| 00:01.25 | fugitivo | not the double = |
| 00:01.27 | fugitivo | => |
| 00:01.46 | *** join/#asterisk supjigatr (n=syslod@152.53.17.26) |
| 00:02.25 | zahid | fugitivo: double = was just a typo here |
| 00:02.48 | fugitivo | what asterisk version? |
| 00:03.11 | zahid | 1.2.0 |
| 00:04.35 | X-Files | ppls, please help ! I use asterisk 1.2.2 and Windows Messenger 5.1 , why i can't see users status online in Messenger ??? |
| 00:06.32 | fugitivo | does osx run on amd64? :) |
| 00:06.38 | fugitivo | only intel? |
| 00:08.18 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:08.23 | Mavvie | fugitivo: OS/X will not be a generic available OS. |
| 00:08.36 | Mavvie | fugitivo: you will need to buy the Apple hardware to run it. |
| 00:08.43 | fugitivo | really? |
| 00:08.46 | Mavvie | yes |
| 00:08.51 | aldsf | is the svn server down? |
| 00:08.51 | aldsf | [root@dhcp90 src]# svn co http://svn.digium.com/svn zaptel libpri asterisk |
| 00:08.51 | aldsf | svn: PROPFIND request failed on '/svn' |
| 00:08.51 | aldsf | svn: PROPFIND of '/svn': 403 Forbidden (http://svn.digium.com) |
| 00:08.56 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-2-97-231.mia.bellsouth.net) |
| 00:09.30 | fugitivo | Mavvie: you can buy it using the webpage |
| 00:09.36 | fugitivo | http://www.apple.com/macosx/techspecs/ |
| 00:09.36 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-2-97-231.mia.bellsouth.net) |
| 00:09.43 | *** join/#asterisk Guggemand (i=Guggeman@tester2.har-tabt.dk) |
| 00:09.45 | fugitivo | $129 for single user |
| 00:10.21 | Mavvie | fugitivo: yes, and that is for... (see Requirements on that page) |
| 00:10.24 | *** join/#asterisk R3DB0x (i=nobody@66.142.28.36) |
| 00:10.26 | fugitivo | or is that for apple hardware not intel only? |
| 00:10.46 | fugitivo | it is |
| 00:10.47 | fugitivo | well |
| 00:10.51 | *** join/#asterisk BillinOffice (n=bill@dsl092-234-029.phl1.dsl.speakeasy.net) |
| 00:10.58 | fugitivo | i'll keep my desktop with linux then |
| 00:11.03 | *** join/#asterisk iKale (n=kizzale@ip70-174-157-198.dc.dc.cox.net) |
| 00:16.12 | R3DB0x | what are some good providers that you guys are using to connecting your * boxes to? |
| 00:16.46 | *** join/#asterisk dimmik (n=dimmik@static217244.dsl.hol.gr) |
| 00:17.17 | *** join/#asterisk fndude (i=sobeit@127-48.124-70.tampabay.res.rr.com) |
| 00:18.17 | dimmik | Hi all. Is there a way to restrict the numbers that a sip phone can transfer to (using sip refer)? |
| 00:19.04 | X-Files | eh :( |
| 00:19.07 | justinu | isn't that handled in the dialplan? |
| 00:19.08 | rob0 | by putting it in a limited / restricted context? |
| 00:19.25 | rob0 | was I right? |
| 00:19.35 | dimmik | not really |
| 00:19.57 | X-Files | ppls, please help ! I use asterisk 1.2.2 and Windows Messenger 5.1 , why i can't see users status online in Messenger ??? Please, answer ... |
| 00:20.16 | dimmik | as it is, the user may transfer to any number available in the context |
| 00:20.33 | dimmik | I want to restrict this |
| 00:20.38 | Pegger | what do people use for t1 fail over (for when machine A dies the t1 is then routed to machine B) |
| 00:21.56 | justinu | dimmik: the call gets bounced back into the same context the SIP device uses |
| 00:21.59 | justinu | on a local channel |
| 00:22.39 | dimmik | is there a way to restrict this for transfers? |
| 00:23.11 | supjigatr | Pegger: NFAS PRI or SS7 |
| 00:23.37 | R3DB0x | what are some good providers that you guys are using for you voip connection? |
| 00:23.49 | justinu | dimmik: good question |
| 00:23.59 | dimmik | :) |
| 00:24.00 | fndude | Been using telasip for a week. so far so good. |
| 00:24.39 | supjigatr | nufone works good. |
| 00:25.38 | FuriousGeorge | ${DIALSTATUS}=UNAVAIL == chanisavail($CHAN)=true |
| 00:25.40 | FuriousGeorge | ? |
| 00:25.47 | FuriousGeorge | i mean FALSE |
| 00:25.48 | FuriousGeorge | FLASE |
| 00:25.53 | FuriousGeorge | er |
| 00:25.55 | FuriousGeorge | :) |
| 00:26.01 | iCEBrkr | FLASE? |
| 00:26.02 | iCEBrkr | haha |
| 00:26.25 | *** join/#asterisk jpablo (n=jpablo@dsl-201-128-19-21.prod-infinitum.com.mx) |
| 00:26.37 | FuriousGeorge | true or false: ${DIALSTATUS}=UNAVAIL == chanisavail($CHAN)=FALSE |
| 00:26.50 | iCEBrkr | FuriousGeorge: I'd have to disagree there |
| 00:26.51 | jpablo | hi, anyone can recomend a simple thing to generate asterisk statitics out of the cdr ? |
| 00:26.52 | R3DB0x | ya nufone was the one i was trying to thinkg of...ty |
| 00:27.03 | jpablo | nothing complicated and big, with pricing and stuff. just a simple thing |
| 00:27.04 | iCEBrkr | jpablo: There's a PHP app that does that.. |
| 00:27.26 | fndude | Is there a way to convert multiline pots phones to work in asterisk, some pci interface card, switch, etc? |
| 00:27.28 | jpablo | iCEBrkr, if you can give me the name and/or url that would rock :) |
| 00:27.39 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 00:27.44 | jpablo | fndude, lots of atas, or the new digium card, check www.digium.com |
| 00:28.09 | FuriousGeorge | iCEBrkr: lets say chan is not avail. what is ${DIALSTATUS} gonna come back as? |
| 00:28.10 | fndude | jpablo, nice, ty. |
| 00:28.20 | jpablo | fndude, TDM2400P card. or lots of FXO atas, check www.voipsupply.com |
| 00:28.24 | *** join/#asterisk angler (n=angler@gateway.digium.com) |
| 00:28.55 | iCEBrkr | FuriousGeorge: It'll attempt to dial and then return UNAVAIL |
| 00:28.56 | dimmik | jpablo: http://areski.net/asterisk-stat-v2/about.php |
| 00:29.08 | jpablo | dimmik, thanks i love you. |
| 00:29.10 | iCEBrkr | IsChanAvail() turns TRUE/FALSE if there's an available channel to make the call. |
| 00:29.21 | FuriousGeorge | so in that case, its true |
| 00:29.22 | iCEBrkr | jpablo: I'm looking |
| 00:29.25 | iCEBrkr | FuriousGeorge: It's not the same thing |
| 00:29.29 | FuriousGeorge | thats the case im worried about |
| 00:29.47 | FuriousGeorge | i guess a sip peer can be unvail and yet the channel is avail |
| 00:29.57 | iCEBrkr | jpablo: CDR Analyser |
| 00:30.01 | FuriousGeorge | oh wait, no it cant |
| 00:30.04 | FuriousGeorge | whatever |
| 00:30.09 | iCEBrkr | FuriousGeorge: Read the Wiki, it explains it |
| 00:30.13 | X-Files | chan_sip.c:3469 process_sdp: Unknown SDP media type in offer: message 5060 sip sip:lala@10.0.0.2 |
| 00:30.14 | iCEBrkr | FuriousGeorge: I was just reading about this today |
| 00:30.17 | X-Files | what this it ? |
| 00:30.37 | jpablo | iCEBrkr, thanks, great. |
| 00:30.43 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 00:30.52 | FuriousGeorge | i just wanna make sure that if my internet connection goes down dialstatus will be unavail and the dialplan will work |
| 00:31.13 | FuriousGeorge | im fairly convinced at this point, gonna give it a whirl :) |
| 00:31.15 | iCEBrkr | FuriousGeorge: That will most likely be the case |
| 00:33.51 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 00:34.06 | *** join/#asterisk wizard545 (n=wizard@tor/session/x-f6464ec356d33cfa) |
| 00:35.17 | wizard545 | hey guys, I tried EVERYTHING before coming here to ask this... I have a problem with a2billing with asterisk... it all works.. except when I try and make an outgoing call it says "number is not available" I've tried almost everything, what is stopping it from dialing out? I can forward calls fine. |
| 00:36.45 | iCEBrkr | wizard545: have you 'set verbose 9' in the CLI? |
| 00:36.50 | iCEBrkr | and watched the call progress? |
| 00:37.13 | fndude | wow the Wildcard TDM400P is like 400+. Whats the budget solution for asterisk + pots phones, if there is one? |
| 00:37.26 | Math` | fndude: how many phones? |
| 00:37.28 | wizard545 | I have the a2billing logs, but I can't make any sense of it |
| 00:37.33 | fndude | just 4. |
| 00:38.03 | Math` | fndude: buy ATAs, a PAP2 costs around 70$ and has 2 fxs ports |
| 00:38.12 | Math` | (canadian dollars) |
| 00:38.31 | fndude | Math`: cool, thanks, thats a little better. |
| 00:38.47 | Math` | just a little cheaper |
| 00:39.32 | Math` | but... do you already have the phones ? |
| 00:39.56 | iCEBrkr | wizard545: Are you using Asterisk@Home? |
| 00:40.12 | wizard545 | no i'm using regular asterisk |
| 00:40.37 | iCEBrkr | ok, so type asterisk -r |
| 00:40.39 | iCEBrkr | and get into the CLI |
| 00:40.51 | wizard545 | ok |
| 00:40.52 | *** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 00:40.56 | *** join/#asterisk riddlebox (n=james@24-171-11-166.dhcp.stls.mo.charter.com) |
| 00:41.03 | smallb | hi all |
| 00:41.07 | wizard545 | I turned on sip debugging.. hoping i would see it |
| 00:41.13 | smallb | need some help with passing dtmf |
| 00:41.14 | iCEBrkr | wizard545: no |
| 00:41.20 | Math` | smallb: just ask |
| 00:41.20 | iCEBrkr | wizard545: just 'set verbose 9' |
| 00:41.47 | wizard545 | ok now try a call? |
| 00:41.50 | Math` | yup |
| 00:41.51 | smallb | my * is routing calls via sip/g29 directly to a cisco router |
| 00:41.51 | iCEBrkr | yeah |
| 00:42.03 | Math` | smallb: ok |
| 00:42.14 | Math` | smallb: use rfc2833 |
| 00:42.14 | smallb | but dtmf is not working, I call dell 1 800 www dell and try the options to no avail |
| 00:42.16 | X-Files | ppls, please help ! I use asterisk 1.2.2 and Windows Messenger 5.1 , why i can't see users status online in Messenger ??? Please, answer ... |
| 00:42.30 | smallb | i am using rfc2833 |
| 00:42.34 | wizard545 | got the output |
| 00:42.38 | smallb | let me get the config |
| 00:42.50 | Math` | smallb: I'm gonnected to a cisco in g729 and dtmf are working fine |
| 00:42.57 | Math` | outbound tho |
| 00:43.05 | smallb | [4237] |
| 00:43.05 | smallb | <PROTECTED> |
| 00:43.05 | smallb | <PROTECTED> |
| 00:43.05 | smallb | <PROTECTED> |
| 00:43.05 | smallb | <PROTECTED> |
| 00:43.05 | smallb | <PROTECTED> |
| 00:43.07 | smallb | <PROTECTED> |
| 00:43.09 | smallb | <PROTECTED> |
| 00:43.11 | smallb | <PROTECTED> |
| 00:43.13 | smallb | <PROTECTED> |
| 00:43.15 | smallb | <PROTECTED> |
| 00:43.17 | Math` | ~pb |
| 00:43.19 | jbot | hmm... pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 00:43.24 | smallb | oops |
| 00:43.31 | wizard545 | Jan 19 19:42:33 WARNING[3609]: file.c:584 ast_readaudio_callback: Failed to write frame |
| 00:43.31 | wizard545 | have anyhting to do with it? |
| 00:43.34 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 00:43.47 | Math` | wizard545: just pastebin the whole output |
| 00:43.57 | wizard545 | ok thanks |
| 00:44.11 | Math` | smallb: your cisco gateway is host=dynamic? |
| 00:44.31 | smallb | no, its an ip address |
| 00:44.34 | Math` | cuz that looks more like your phone config than the cisco config |
| 00:44.36 | *** join/#asterisk jeffgus_ (n=jeffgus@greengables.zimage.com) |
| 00:44.36 | iCEBrkr | wizard545: Didn't you see it execute Dial()? |
| 00:44.47 | smallb | yes that was the client - x-lite |
| 00:45.17 | smallb | the cisco is type=peer,host=192.168.93.1, dtmfmode=rfc2833 etc |
| 00:45.54 | wizard545 | http://pastebin.com/513937 |
| 00:46.03 | Math` | have you checked the cisco's config? |
| 00:46.14 | wizard545 | at the end it just kept giving me the "number is not available" |
| 00:46.30 | Math` | ah thats a calling card app |
| 00:46.37 | wizard545 | No i didn't see a "Dial" |
| 00:46.48 | smallb | hmm, i don't have that, it belongs to a telco |
| 00:47.21 | Math` | smallb: then try asking your telco, or dtmfmode=info... maybe they use that |
| 00:47.24 | wizard545 | Math yes... it's pretty awesome if i just could get this Dial working |
| 00:47.35 | iCEBrkr | wizard545: Um at the top. |
| 00:47.54 | smallb | i guess i will have to try that, because * configs are fairly straightforward |
| 00:47.59 | iCEBrkr | wizard545: You see it executed Answer() |
| 00:48.03 | smallb | not hard to try the various options |
| 00:48.15 | wizard545 | yea |
| 00:48.34 | Math` | iCEBrkr: its a calling card script, it needs to answer to ask for the pin and the number to dial.. |
| 00:48.47 | jpablo | grrr, cdr analizer doesn't support sqlite |
| 00:49.03 | iCEBrkr | Math`: Umm, yeah, and you see how it says DEADAGI() that means it hung up |
| 00:49.08 | wizard545 | <PROTECTED> |
| 00:49.19 | wizard545 | it asks for pin.. and asks for the number |
| 00:49.24 | iCEBrkr | jpablo: You asked for something to log CDR.. I gave it to you :) |
| 00:49.37 | wizard545 | then says "number not available" and doesn't do a dial |
| 00:49.38 | iCEBrkr | DeadAGI() is executed when hanging up |
| 00:51.05 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 00:51.11 | jpablo | iCEBrkr, yeah, i would probably roll my own php script using sqlite, i basically just need someone to know how is hammering the phone. |
| 00:51.25 | jpablo | s/how/who |
| 00:51.57 | jpablo | i could also hack cdr analizer .. |
| 00:52.11 | riddlebox | I am trying to use python with AGi scripting, but none of the howtos seem to work, can anyone offer me any help? |
| 00:52.18 | Beirdo | hehe |
| 00:52.37 | Beirdo | "analizer" sounds like something completely different |
| 00:52.43 | iCEBrkr | Beirdo: lol |
| 00:53.14 | jpablo | jeje, sorry |
| 00:53.22 | jpablo | not english native speacker :P |
| 00:53.29 | Beirdo | that's OK |
| 00:53.38 | Beirdo | your English is way better than my Spanish |
| 00:53.48 | Beirdo | although that will change given time |
| 00:54.12 | *** join/#asterisk tainted_ (n=somewher@mail.k2usa.com) |
| 00:54.14 | Beirdo | my fiancee will make sure of that |
| 00:55.19 | tainted_ | any tcpdump gurus? |
| 00:55.26 | jpablo | spanish or LA girl, eh? |
| 00:55.35 | jpablo | tainted_, not really, but what do youneed? |
| 00:55.36 | tainted_ | i need to pinpoint an audio problem and could use some advice |
| 00:55.56 | Beirdo | Puerto Rican |
| 00:56.14 | shmaltz | tainted_, shoot |
| 00:56.20 | Beirdo | BANG! |
| 00:56.45 | rob0 | you missed |
| 00:56.55 | tainted_ | well audio cuts out intermittently during phone conversations |
| 00:57.09 | tainted_ | and i'm trying to figure out if it's dropped RTP packets |
| 00:57.14 | jpablo | tainted_, is it lan, dls or something ? |
| 00:57.40 | tainted_ | it's cogent fiber |
| 00:58.12 | shmaltz | tainted_, whats the speed, and latency? |
| 00:58.26 | shmaltz | ping for around 1000 times and report the avg |
| 00:58.29 | *** join/#asterisk usam (n=usam@203.156.61.204) |
| 00:58.38 | tainted_ | 100Mbps, we have SLA for 50ms avg |
| 00:58.55 | tainted_ | it's around 50ms |
| 00:59.01 | tainted_ | in practice as well |
| 01:00.18 | shmaltz | tainted_, and on the other side? what is the speed? |
| 01:00.22 | shmaltz | is this point to point? |
| 01:00.30 | tainted_ | yes |
| 01:00.32 | shmaltz | IAX? SIP? TDMoE? |
| 01:00.48 | tainted_ | SIP |
| 01:01.30 | jpablo | humm |
| 01:01.31 | jpablo | codec? |
| 01:01.43 | jpablo | equipment ? |
| 01:01.45 | tainted_ | 729 |
| 01:02.10 | tainted_ | polycom 301s, grandstream 488s, dell poweredge 2850s |
| 01:02.30 | tainted_ | mixed environment of 1.0.7 and 1.2.0 |
| 01:03.02 | jpablo | is asterisk in the media path ? have you tried doing with out it ? |
| 01:03.13 | tainted_ | yes it is |
| 01:03.26 | tainted_ | hmm.. i will try it w/o asterisk in the middle |
| 01:03.37 | jpablo | try putting it out of the media path, to discard some failiure there |
| 01:03.49 | shmaltz | tainted, you have debuggin on in the logs? |
| 01:04.06 | tainted_ | i have default settings for debug |
| 01:04.06 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 01:04.24 | tainted_ | i watch console with sip debug and verbose at 6 |
| 01:04.39 | tainted_ | no warnings in particular |
| 01:04.54 | tainted_ | and not isolated to any particular provider |
| 01:04.56 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 01:04.57 | shmaltz | tainted, try this: |
| 01:04.59 | shmaltz | cd /var/log/asterisk |
| 01:05.00 | shmaltz | grep frame * |
| 01:05.02 | shmaltz | do you get anything? |
| 01:05.41 | tainted_ | yes |
| 01:05.56 | shmaltz | what? paste bin it please |
| 01:05.57 | *** join/#asterisk JMcA (n=jmcadams@71.31.33.169) |
| 01:06.03 | tainted_ | bunch of old errors |
| 01:06.15 | shmaltz | like channle.c or something else? |
| 01:06.28 | X-Files | please, check this error : http://pastebin.ca/37488 |
| 01:06.35 | tainted_ | messages: Jan 18 23:18:02 WARNING[3885]: Received mini frame before first full voice frame |
| 01:06.58 | shmaltz | X-Files, thats a warning, not an error |
| 01:07.10 | Mavvie | that reminds me, who knows a video-conferencing client which works with the MeetMe conferences? |
| 01:07.12 | X-Files | why i cant use stream video ? |
| 01:07.17 | shmaltz | tainted, comment out the line: |
| 01:07.19 | shmaltz | ;debug => debug |
| 01:07.20 | shmaltz | in /etc/asterisk/logging.conf |
| 01:07.33 | X-Files | no, Microsoft Messenger |
| 01:07.38 | X-Files | shmaltz: ok wait |
| 01:07.44 | shmaltz | <PROTECTED> |
| 01:07.55 | Zodiacal- | how can i unlock my cisco 7960g so that i can change the network settings? |
| 01:08.01 | tainted_ | shmaltz okay |
| 01:08.08 | Zodiacal- | under network configration it has a little padlock locked |
| 01:08.13 | Zodiacal- | it only seems to display the info |
| 01:08.13 | tainted_ | asterisk -rx reload ? |
| 01:08.48 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 01:09.05 | shmaltz | tainted_, now do in the CLI a logger reload and wait until it happens again, when it does check that grep again like this: |
| 01:09.06 | shmaltz | grep frame /var/log/asterisk/debug |
| 01:09.08 | shmaltz | if you get anything let me know |
| 01:09.19 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com) |
| 01:09.30 | tainted_ | shmaltz will do thanks a bunch! |
| 01:10.01 | *** join/#asterisk denon23532 (i=denon@synapse.subneural.net) |
| 01:11.00 | X-Files | check it : http://pastebin.ca/37489 |
| 01:11.02 | X-Files | please |
| 01:11.20 | shmaltz | tainted_, if you get something like this: |
| 01:11.21 | shmaltz | Jan 19 05:32:12 DEBUG[31465] channel.c: Didn't get a frame from channel: SIP/211-7d72 |
| 01:11.23 | shmaltz | then let me know |
| 01:11.51 | tainted_ | ok |
| 01:11.58 | Zodiacal- | any ideas? |
| 01:12.00 | shmaltz | there has been some reports of this, but nobody to help decide why and where from, so we need somehelp overhere |
| 01:12.11 | tainted_ | what version of asterisk though? |
| 01:12.15 | shmaltz | tainted_, when you have that problem, what happens with the call? |
| 01:12.23 | tainted_ | the audio cuts out for a few seconds |
| 01:12.27 | tainted_ | then comes back |
| 01:12.30 | shmaltz | tainted_, I tried it on: 1.0.10, 1.2.1, and 1.2.2 |
| 01:12.31 | Zodiacal- | all the dox say to select the unlock option, but i don't see that, i only have: contrast, ring type, network configuration, status. |
| 01:12.32 | JMcA | tainted_: what codec? |
| 01:12.37 | tainted_ | 729 |
| 01:12.53 | tainted_ | call is never disconnected.. just audio cut outs |
| 01:13.05 | shmaltz | tainted_, thats a littel different then what we've seen, we see that it stops bridging and then calls again using the ring |
| 01:13.21 | tainted_ | haven't see that one yet |
| 01:13.26 | tainted_ | seen |
| 01:13.30 | Zodiacal- | nm figured it out.. |
| 01:13.33 | *** join/#asterisk iq|tablet (n=iq@71-38-74-41.omah.qwest.net) |
| 01:14.39 | JMcA | Zodiacal-: yeah, all the way at the bottom |
| 01:15.35 | shmaltz | gtg guys |
| 01:15.37 | shmaltz | c ya |
| 01:15.43 | tainted_ | thx shmaltz |
| 01:15.51 | X-Files | ppls, what me need for asterisk and microsoft messenger I can use chat and Camera translate ? |
| 01:16.16 | shmaltz | X-Files, where you from? |
| 01:16.41 | X-Files | shmaltz: latvia |
| 01:16.46 | Zodiacal- | jmca it wasn't there i had to press **# |
| 01:16.47 | shmaltz | oic |
| 01:16.58 | JMcA | Zodiacal-: ah...funky |
| 01:17.00 | Zodiacal- | jmca it now has an unlocked padlock icon, but it doesn't give me options to change anything |
| 01:17.21 | Zodiacal- | i.e. i highlight dhcp server, cuz i wanta put an ip in there... and theres only a save and cancel soft key options |
| 01:17.46 | *** join/#asterisk annonimous (n=annonimo@dsl-201-129-251-131.prod-infinitum.com.mx) |
| 01:17.49 | annonimous | hello |
| 01:17.49 | JMcA | uhm...its kinda counter-intuitive to input a dhcp server address |
| 01:18.13 | JMcA | that would tend to be a read-only option, typically |
| 01:18.17 | Zodiacal- | or even a manual ip address |
| 01:18.24 | Zodiacal- | i can't change any of the network configuration options in here |
| 01:18.26 | annonimous | good afternoon, i have a "demo" of a hardware called "audiocode" and i want to know if its compatible with asterisk =S |
| 01:18.30 | wizard545 | anyone know how to unlock the settings on a cisco 7940? |
| 01:18.49 | JMcA | Zodiacal-: see if there's an unlock option at the very bottom of the menu? that's how I do it on my 7960 |
| 01:18.50 | X-Files | Jan 20 03:25:08 WARNING[949]: chan_sip.c:3469 process_sdp: Unknown SDP media type in offer: message 5060 sip sip:user@10.0.0.1 <<--- i write message |
| 01:19.15 | Zodiacal- | jmca to the main menu after you press the check box button? |
| 01:19.18 | Zodiacal- | or the network configuration menu? |
| 01:19.23 | X-Files | Jan 20 03:25:57 WARNING[949]: chan_sip.c:3469 process_sdp: Unknown SDP media type in offer: video 14498 RTP/AVP 34 31 <<---- i see this in call mode video :( |
| 01:19.26 | JMcA | the main menu, I think |
| 01:19.37 | X-Files | why this not working and warning SDP ? |
| 01:19.49 | Zodiacal- | jmca theres only four options on it |
| 01:20.00 | Zodiacal- | jmca: contrast, ring type, network configuration, status. |
| 01:20.12 | JMcA | I dunno, then...I don't think the menus on mine are laid out the same way |
| 01:20.13 | Zodiacal- | i havn't installed any other firmware |
| 01:20.18 | Zodiacal- | its the default brand new phone |
| 01:20.26 | JMcA | oh...it has a skinny load on it? |
| 01:20.37 | Zodiacal- | i guess |
| 01:20.40 | Zodiacal- | im new obvously |
| 01:20.46 | Zodiacal- | :P |
| 01:21.01 | JMcA | I assume they default to skinny...mine has a sip load |
| 01:21.02 | Zodiacal- | i want to tell it where my tftp server is so i can update to sip |
| 01:21.38 | JMcA | you may need to set up a dhcp server and have the dhcp server tell it...I've never done the update, so I don't know for sure what they do |
| 01:21.57 | JMcA | I just borrowed one from work that already had a sip load on it |
| 01:22.01 | Zodiacal- | ic |
| 01:22.06 | Zodiacal- | i have a dhcp server running |
| 01:22.09 | Zodiacal- | it doesn't find it |
| 01:22.36 | Zodiacal- | omg it just found it, after like 10 mins |
| 01:22.38 | Zodiacal- | hehe :) |
| 01:22.54 | JMcA | welcome to the world of Cisco |
| 01:22.54 | Zodiacal- | jmca Thanks! |
| 01:23.01 | Zodiacal- | jmca thats not good to hear.. |
| 01:23.42 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 01:23.42 | Mavvie | have the same here with a 7970 on a C831 router. |
| 01:23.44 | JMcA | yeah...took us a while to figure out how to turn off VAD on our AS5850's...we had to turn off some random fax setting |
| 01:23.56 | Mavvie | keeps DHCP discovering for hours, and then suddenly accepts the offer. |
| 01:24.31 | JMcA | typical Cisco...if they made home appliances, pressing the clock button on the microwave would start the clothes dryer |
| 01:24.38 | Zodiacal- | its not letting me set the tftp server address tho |
| 01:24.40 | Zodiacal- | grr.. |
| 01:24.48 | Mavvie | Zodiacal-: "use alternate tftp server" |
| 01:24.48 | Ariel_ | Hello everyone |
| 01:25.14 | X-Files | http://pastebin.ca/37491 |
| 01:25.16 | X-Files | opz |
| 01:25.32 | Zodiacal- | mavvie i see that option but its set to no, theres no way to change it to yes.. :/ |
| 01:25.39 | Zodiacal- | the soft keys only say: save, cancel |
| 01:25.45 | Mavvie | Zodiacal-: press **# to unlock the settings. |
| 01:26.05 | Zodiacal- | yep, just did, it locks automaticly quickly :P |
| 01:27.55 | FuriousGeorge | the tar inside asterisk-sounds has all these funny sounds in it, then i make install and cant find them in var/lib/asterisk/sounds |
| 01:28.15 | FuriousGeorge | "all your base are belong to us" she says |
| 01:29.30 | brockj49464 | A X100P can only be used as FXO and not as FXO the line side and FXS on the phone side correct? |
| 01:29.44 | co-bdg^-^ | if i have one account in let say voipulse and want to use that one account to call out for 5 simultans calls out, is that possible ? |
| 01:29.53 | *** join/#asterisk zu (n=raz@89-pool1.ras14.floca.alerondial.net) |
| 01:29.56 | mog_work | you can only use it as a single fxo device brockj49464 |
| 01:30.00 | Ariel_ | X-Files, your trying to run video with msn on asterisk? |
| 01:30.04 | mog_work | she is pretty funny FuriousGeorge |
| 01:30.09 | zu | hy all |
| 01:30.19 | X-Files | Ariel_: yes |
| 01:30.25 | FuriousGeorge | "weasels have eaten our phone system" |
| 01:30.28 | *** part/#asterisk zu (n=raz@89-pool1.ras14.floca.alerondial.net) |
| 01:30.30 | co-bdg^-^ | so it can't used for 5 calls out ? |
| 01:30.43 | Ariel_ | X-Files, last I knew msn did not support normal sip/video calls |
| 01:30.49 | *** join/#asterisk zu (n=raz@89-pool1.ras14.floca.alerondial.net) |
| 01:31.05 | Ariel_ | co-bdg^-^, which voicepulse type of account. |
| 01:31.08 | X-Files | Ariel_: ok, but messeges support ? |
| 01:31.08 | zu | hy all |
| 01:31.35 | Ariel_ | msn 4.7 does no video but sip call yes but not 5.0 and above |
| 01:31.44 | mog_work | thats my favorite sound FuriousGeorge |
| 01:32.10 | zu | I prefer devices that do mpeg4 video codecs for high resolution H.264 |
| 01:32.16 | X-Files | Ariel_: i use 5.1 |
| 01:32.35 | zu | X-Files, what device? |
| 01:32.36 | Ariel_ | co-bdg^-^, voicepulse I have there inbound did only and they allow up to 4 calls. outbound since you pay by the minute 2.4 cents they don't restrict it. |
| 01:32.57 | Ariel_ | X-Files, use eyebeam |
| 01:33.11 | X-Files | zu: ? |
| 01:33.25 | zu | for your video cam |
| 01:33.51 | FuriousGeorge | mog_work: how come i cant find all_your_base in the sounds dir? where did that get make installed to? |
| 01:33.58 | X-Files | zu: notecam 300 usb |
| 01:34.23 | wizard545 | anyone know how many simultaneous connects my provider will allow? telesip? i've called the same number with 5 lines and it's never rang busy or anything |
| 01:34.28 | mog_work | is it not in that dir? |
| 01:34.33 | *** join/#asterisk Reverend (n=owned@68-169-204-147.agstme.adelphia.net) |
| 01:34.35 | FuriousGeorge | did it get renamed? |
| 01:34.48 | brockj49464 | Any idea how I configure * to correctly figure out which account a call is coming in from when I peer to the same server 3 times? |
| 01:34.53 | Ariel_ | wizard545, do they charge by the minute or unlimited |
| 01:35.03 | mog_work | i dont know |
| 01:35.05 | co-bdg^-^ | Ariel_: yes ofcourse ... we pay for it |
| 01:35.13 | *** join/#asterisk Jabron1 (n=Hercules@red-corp-200.76.249.142.telnor.net) |
| 01:35.18 | Reverend | can mpg123 play mp3 streams? |
| 01:36.47 | co-bdg^-^ | Ariel_: so if we buy one account it can used by unlimited caller out in our office ? |
| 01:36.47 | Ariel_ | co-bdg^-^, if you get the one that you pay by the minute correct. But if you get there unlimite no it's does not allow more then a few calls outbound. |
| 01:36.47 | zu | wizard providers usualy dont care about multiple connects because it means more minutes and more money |
| 01:36.55 | FuriousGeorge | mog_work: the asterisk-sounds tar contains 1300 files /var/lib/asterisk/sounds contains only 348 |
| 01:37.22 | X-Files | Ariel_: u have eyebeam ? i can't find where download |
| 01:37.31 | Ariel_ | X-Files, xten |
| 01:37.53 | FuriousGeorge | X-Files: eyebeam costs money |
| 01:38.01 | X-Files | ;(((((((( |
| 01:38.31 | co-bdg^-^ | Ariel_: can you more precise to explain a few ... ? how much calls out ? |
| 01:38.47 | Ariel_ | FuriousGeorge, last I knew you had to run make install in the asterisk-sounds download to get the files installed. |
| 01:38.48 | mog_work | heh someone needs to fix make install i see.... |
| 01:39.05 | [iPBX]Reverend | co-bdg^-^ which provider? i just joined... who has the 'unlimited' you're talking about? |
| 01:39.10 | mog_work | did you do make install in the asterisk sounds dir |
| 01:39.28 | Ariel_ | co-bdg^-^, if you pay for each minute of the call it's only not limited |
| 01:39.29 | FuriousGeorge | mog_work: i didnt |
| 01:39.41 | mog_work | heh |
| 01:39.43 | mog_work | there you are |
| 01:39.44 | zu | usualy its unlimited untill you use a ton of minutes |
| 01:40.02 | co-bdg^-^ | Ariel_: ok ... thanks |
| 01:40.10 | FuriousGeorge | mog_work: no target in that dir |
| 01:40.13 | *** part/#asterisk stdio (n=stdio@pcp01473275pcs.lncstr01.pa.comcast.net) |
| 01:40.28 | X-Files | Ariel_: X-Lite support video and messege ? |
| 01:40.33 | Ariel_ | voicepulse will allow you only about 2 or 3 calls outbound on there unlimite account. But there pay per minute they don't care since your paying for every call. |
| 01:40.47 | Ariel_ | X-Files, no xlite no video only eyebeam |
| 01:40.50 | FuriousGeorge | oh wait, i did do it in the dir of the tarball i extracted |
| 01:40.55 | X-Files | ;( |
| 01:40.58 | FuriousGeorge | i guess thats what you meant Ariel_ |
| 01:41.05 | Ariel_ | FuriousGeorge, yes |
| 01:41.47 | FuriousGeorge | yeah, so i did do that, its mog_work 's fault till he fixes the make install |
| 01:41.55 | X-Files | Ariel_: have alternative eyebeam ? |
| 01:41.55 | mog_work | lol |
| 01:42.04 | mog_work | its always my fault isnt it ^_^ |
| 01:42.16 | FuriousGeorge | Ariel_: they got great creams pills and lotions for hairloss these days |
| 01:42.24 | zu | Ariel_: ya its a pain in the ass webcams unless its working hours for conferencing |
| 01:42.40 | FuriousGeorge | hey mog_work now you have an excuse to call it asterisk sounds 1.2.2 right |
| 01:42.57 | mog_work | heh |
| 01:42.59 | mog_work | indeed |
| 01:43.05 | Ariel_ | FuriousGeorge, yes but I am a poor consultant can't afford them. |
| 01:43.07 | mog_work | im working on dtmf issue at the moment |
| 01:43.30 | FuriousGeorge | mog_work: you're gonna have to dedicate the upgrade to furious george now |
| 01:43.37 | FuriousGeorge | for his tireless debugging |
| 01:43.46 | zu | dtmf issues can be a pain in the but |
| 01:43.50 | X-Files | Ariel_: last question, in microsoft messenger have status contacts, why status not worked ? |
| 01:44.09 | Ariel_ | X-Files, I don't use msn it sucks |
| 01:44.11 | *** join/#asterisk penghb (n=penghb@202.108.130.138) |
| 01:44.26 | X-Files | Ariel_: status online have in eyebeam ? |
| 01:44.33 | rob0 | ~seen [TK]D-Fender |
| 01:44.46 | jbot | [tk]d-fender <n=joe@toronto-HSE-ppp4122655.sympatico.ca> was last seen on IRC in channel #asterisk, 4h 45s ago, saying: '[av]bani : Will look at one I'm home. ALter all!'. |
| 01:44.46 | JMcA | X-Files: I believe msn uses SIMPLE for its presense...dunno if asterisk supports that |
| 01:44.47 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 01:45.21 | Ariel_ | only one that works for sip calls via asterisk is the older one 4.7 not the newer ones 5.x and above |
| 01:45.52 | X-Files | JMcA: only voice protocol work in 5.1 , not video and messenge |
| 01:46.12 | X-Files | p.s. and status online |
| 01:46.21 | JMcA | Ariel_: I just want it all to work together...and its getting closer to that all the time |
| 01:46.38 | mog_work | bye |
| 01:47.31 | Ariel_ | isn't msn at version 7.5 now. Last I looked it did not support sip at all for voice it has it's own thing. |
| 01:48.13 | Ariel_ | JMcA, not everything will work together. But that is a good wish. |
| 01:49.05 | *** join/#asterisk Flauto (n=zhao@c-71-194-194-48.hsd1.il.comcast.net) |
| 01:49.09 | Flauto | hi people |
| 01:49.22 | JMcA | Ariel_: no, but the more that does, the better |
| 01:49.29 | Flauto | is there anyone can give me the asterisk cvs download information? |
| 01:49.37 | Flauto | would not find it any more |
| 01:49.41 | zu | its on asterisk.org |
| 01:50.02 | Flauto | zu, there is svn but no cvs |
| 01:50.15 | zu | get svn |
| 01:50.26 | zu | cvs is depreciated into oblivion |
| 01:50.34 | Math` | lol |
| 01:50.34 | Flauto | but i dont' see asterisk-addons and asterisk-sounds there |
| 01:50.35 | Nugget | "deprecated", not "depreciated" |
| 01:50.53 | *** join/#asterisk Lurr (n=pr0ph3t@adsl-2-97-231.mia.bellsouth.net) |
| 01:50.55 | JMcA | its depreciated too...its not worth anything anymore :) |
| 01:51.00 | zu | lol |
| 01:51.01 | *** part/#asterisk Lurr (n=pr0ph3t@adsl-2-97-231.mia.bellsouth.net) |
| 01:51.10 | *** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it) |
| 01:51.19 | zu | God dam cvs and its nasty ass repository locks out of the blue |
| 01:51.29 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 01:51.33 | zu | I would rather have subversion with a berklydb backend |
| 01:51.53 | Flauto | where can i get asterisk-addons and asterisk-sounds then? |
| 01:52.06 | zu | from asterisk.org |
| 01:52.13 | javier | Need help fixing CallerID using X100P |
| 01:52.17 | zu | get the 1.2.1 branch |
| 01:52.19 | Flauto | svn does not have that listed |
| 01:52.19 | dily | anyone can explane me how to build res_sqlite3? |
| 01:52.24 | Math` | zu: you mean 1.2.2 |
| 01:53.08 | zu | Math`: Ill up to that once I move my patch code too it |
| 01:53.15 | Ariel_ | last I saw there was still a cvs up and running |
| 01:53.15 | *** join/#asterisk wizard545 (n=wizard@it-hluchnik.de) |
| 01:53.22 | Math` | zu: which patch code |
| 01:53.32 | zu | fuzz with a diff is okay but hunks failing can cause issue |
| 01:53.33 | Flauto | there are two options |
| 01:53.37 | *** join/#asterisk nvrs (i=RUR@Kitchener-HSE-ppp3565498.sympatico.ca) |
| 01:53.47 | Flauto | from cvs, i was always downloading the development |
| 01:53.48 | Ariel_ | cvs co r1.2 should work. |
| 01:54.02 | JMcA | Flauto: why do you want the absolute latest? |
| 01:54.03 | Ariel_ | cvs plain yes but you can put r1.2 |
| 01:54.08 | zu | Math`: code Im planing to release into the code base, since I got a fax on file ;) |
| 01:54.16 | Flauto | but jmca, not really |
| 01:54.22 | Math` | zu: doing what |
| 01:54.23 | Flauto | so i will go with 1.2 |
| 01:54.26 | Flauto | thanks guys |
| 01:54.45 | zu | Math`: stuff that I need to do |
| 01:55.03 | hugo-v6 | is it correct that the * mysql backend stores extensions sip and voicemail configs but the * sqlite backend stores only extensions and cdr data? |
| 01:55.04 | Flauto | i liked cvs because i did not need to tar all the files |
| 01:55.32 | Ariel_ | export CVSROOT=:pserver:anoncvs@cvs.digium.com:/usr/cvsroot |
| 01:55.41 | Ariel_ | I just did that and it's working.. |
| 01:55.42 | javier | Has anyone had any success with Caller ID using X100P |
| 01:56.12 | Ariel_ | cvs login then password anoncvs |
| 01:56.23 | Ariel_ | cvs checkout -r v1-2 zaptel asterisk asterisk-addons asterisk-sounds |
| 01:56.30 | Ariel_ | hay it's still all there. |
| 01:56.44 | zu | cvs originated from a set of shell scripts |
| 01:56.49 | Ariel_ | javier, depending on your location it works |
| 01:57.12 | *** join/#asterisk Zand3r (n=Zand3r@spc2-bolt7-3-0-cust141.bagu.broadband.ntl.com) |
| 01:57.38 | javier | Ariel, I am in the US. But I cannot get it going. |
| 01:57.38 | zu | Math`: If I have a problem with asterisk I can usualy go through the code and find the problem |
| 01:58.13 | *** join/#asterisk Druken (n=druken@out.clearnet.com) |
| 01:58.20 | Ariel_ | javier, I have not used one in years but they work fine. I know a few customers using the orginal x101p from digium do you have that or a clone? |
| 01:59.09 | javier | I thought I was getting an original but got a clone, I think. It is from www.x100p.com |
| 01:59.38 | Ariel_ | digium no longer sells them... the switched to the tdm400p instead. |
| 01:59.44 | zu | Thats probably why I have a love/hate relationship with asterisk's ael, and sometimes new modules do need to be imported. since ael is just a scripting language for asterisk's backend c modules |
| 01:59.45 | javier | Call come in and out but no Caller Id. I get UNKOWN. |
| 02:00.30 | Ariel_ | javier, how did you set the board up. whats your zapata.conf look like for it. use pastebin.ca |
| 02:00.40 | zu | that could be a tx/rx level problem or other phones on the pots line causing other signal degridation issues |
| 02:00.57 | *** join/#asterisk znoG_ (n=gs@33-138-114-200.fibertel.com.ar) |
| 02:01.01 | Druken | god rogers tech support are suck dweebs.... |
| 02:01.26 | Druken | not ALL of the world are stupid... just 90% of it |
| 02:01.27 | javier | HOw can i send you the file. This is the first time I use chat. What is pastebin.ca, sorry. |
| 02:01.39 | Flauto | ariel what was the name for development from cvs |
| 02:02.04 | zu | Im interested in trying the san's new multiport cards for pots line with hardware echo supression anyone try them yet? |
| 02:02.08 | javier | I thought it was an RX/TX problem, but don't know what setting to try. |
| 02:02.08 | Ariel_ | Flauto, the development one is without the version included |
| 02:02.21 | Ariel_ | ~pastebin |
| 02:02.23 | jbot | pastebin is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 02:02.41 | Flauto | just zaptel asterisk .....etc? ariel? |
| 02:02.50 | Ariel_ | javier, do you know how to copy and paste??? |
| 02:02.52 | *** join/#asterisk iccomputing (n=Wireless@cpe-69-133-109-130.woh.res.rr.com) |
| 02:03.02 | zu | do you have your local miliwatt test line to test the x100 javier |
| 02:03.11 | Ariel_ | Flauto, yes |
| 02:03.12 | javier | Yes, I do I am very computer savy.. but not here. |
| 02:03.12 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 02:03.23 | Flauto | tnanks |
| 02:03.33 | javier | What is a miliwat test? |
| 02:03.39 | iccomputing | so does anyone know why I keep getting a 401 Unauthorized error on my sip debug? |
| 02:03.40 | *** join/#asterisk calennert (n=calenner@66-191-55-096.dhcp.gnvl.sc.charter.com) |
| 02:03.49 | zu | its used to set your tx and rx levels |
| 02:04.05 | Ariel_ | javier, post your settings for the zapata.conf in the /etc/asterisk directory on to the pastebin and tell us the link for it so we can look at it |
| 02:04.18 | zu | if you have two tdm lines you can loop back your milliwat test to get the levels just right |
| 02:04.32 | Ariel_ | callerID works and it's very simple is your telco supports it. |
| 02:04.39 | zu | javier: check out this site http://www.sineapps.com/news.php?rssid=297 |
| 02:04.48 | javier | OK... |
| 02:04.52 | zu | thats a article on how to setup a tdm line |
| 02:05.53 | Flauto | ariel it worked |
| 02:05.55 | iccomputing | SIP/2.0 401 Unauthorized ......i know the user/pass is right...anyone got any ideas? |
| 02:05.56 | Flauto | i got the 1.2 |
| 02:05.59 | Flauto | thanks |
| 02:06.13 | javier | I think I got the hang of the pastbin. http://pastebin.com/514016 |
| 02:06.49 | zu | iccomputing: 401's can be alot of things, its like if your girlfriend is pissed off for no reason |
| 02:06.50 | javier | I will look at that zu. |
| 02:07.07 | zu | first remove the authentication and see if it works |
| 02:07.21 | Ariel_ | Flauto, great. glad to help |
| 02:07.23 | zu | then check if you device needs something like a md5 secrect |
| 02:07.41 | zu | make sure the firmware is upto date too |
| 02:09.30 | Ariel_ | javier, ok the settings look ok. but your using amp or asterisk@home. I see your tx and rx gains high try them at 0.0 to test out with. |
| 02:09.32 | zu | annonimo is anonymous in spanish I guess |
| 02:09.40 | Ariel_ | and let the phone ring 2 times before picking it up |
| 02:10.02 | zu | ya alot of systems the callerid comes between the 1-2 ring |
| 02:10.18 | zu | why I have no idea |
| 02:10.22 | JMcA | zu: because the data isn't sent until then |
| 02:10.23 | Ariel_ | zu, it needs to send it between the rings |
| 02:10.45 | zu | Yes I know, I was being sarcastic |
| 02:10.56 | Ariel_ | pri lines send it at the same time due to it has a d channel and bchannels for voice |
| 02:11.04 | zu | I would have put it before the 1st ring, but that makes to much fucking sence |
| 02:11.15 | javier | I am using Asterisk@home, the setting were at 0. You want me to put them back. |
| 02:11.17 | zu | </end rant> |
| 02:11.25 | Ariel_ | zu it was done may years ago by the big bells |
| 02:11.36 | JMcA | Ariel_: channelized T1's send it during call setup too....pots is about the only telco technology that doesn't send it during initial call setup |
| 02:11.53 | javier | How can I let the X100P ring more than twice. |
| 02:11.55 | Ariel_ | javier, the tx and rx don't do much for callerID they are used for volume and echo |
| 02:12.03 | zu | Ariel_: yes I know and they are the telephone company they dont have to care, I remember when there were alot of Xbars that used inband signaling |
| 02:12.21 | JMcA | zu: I think because the phone needs to be ready for it |
| 02:12.27 | Ariel_ | zu, same here |
| 02:12.43 | Ariel_ | javier, just don't pick it up till after the 2nd full ring |
| 02:12.59 | zu | thats about 12 seconds |
| 02:13.06 | zu | Wait(11) |
| 02:13.10 | zu | Answer() |
| 02:13.12 | JMcA | there's a WaitRing() right? |
| 02:13.13 | Ariel_ | hummmm |
| 02:13.28 | zu | yea |
| 02:13.33 | Ariel_ | 20 is 4 to 5 rings.... |
| 02:14.00 | *** join/#asterisk reza (i=reza@abort.boom.net) |
| 02:14.04 | reza | ,docs |
| 02:14.07 | reza | ,doc |
| 02:14.12 | Ariel_ | ~doc |
| 02:14.17 | jbot | Documentation can be found at http://digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk |
| 02:14.18 | JMcA | sneezy |
| 02:14.21 | reza | thnx |
| 02:14.22 | zu | ~beer |
| 02:14.23 | jbot | beer is, like, ummm, ummm good!, or good for you!... not just for breakfast anymore |
| 02:14.43 | Ariel_ | ~weather ktmb |
| 02:14.45 | zu | hehehehe |
| 02:14.58 | reza | i'm going to get this fucking thing to work if it's the last thing i do.. |
| 02:15.03 | zu | hey im in fort liquerdale |
| 02:15.08 | reza | actually, how hard is it to get astrisk running? |
| 02:15.09 | Ariel_ | get wat thing to work |
| 02:15.14 | Ariel_ | zu homestead |
| 02:15.52 | zu | cool I used to do satellite dishes down there in 92 after andrew |
| 02:16.04 | Ariel_ | I just think people get the wrong Idea of homestead since the program invasion is out... |
| 02:16.10 | zu | tvro=television recieve only, |
| 02:16.19 | JMcA | wish it were close to 70 here |
| 02:16.29 | zu | whats that, ive been up north for the last 4 years |
| 02:17.18 | zu | I was in florida city, I went there from pembroke pines because they thought it would hit broward, it took me three days to get home |
| 02:17.18 | *** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net) |
| 02:17.32 | FuriousGeorge | <PROTECTED> |
| 02:17.34 | *** join/#asterisk mgoh (n=goh@60.49.6.190) |
| 02:17.46 | De_Mon | with autofallthrough = yes, how do I tell my dialplan to wait for the next extension? |
| 02:17.48 | FuriousGeorge | i tried a local radio station but wouldnt you know it, there not busy |
| 02:17.58 | Ariel_ | n |
| 02:18.05 | *** join/#asterisk kio (n=kio@ool-4577adba.dyn.optonline.net) |
| 02:18.12 | Ariel_ | FuriousGeorge, pizza hut |
| 02:18.31 | zu | yea but I went through wilma in broward it, it took me 6 hours to cut a path to the road and another 4 to find warm beet |
| 02:18.36 | zu | s/beet/beer |
| 02:18.47 | co-bdg^-^ | what;s the difference beetwen e1 and t1 ? |
| 02:18.57 | Ariel_ | co-bdg^-^, us and eu |
| 02:18.59 | zu | I was like shit, I have enough supplies for 2 weeks and Im out of beer |
| 02:19.02 | JMcA | co-bdg^-^: 6 channels |
| 02:19.09 | zu | e1=30 channels t1=24 |
| 02:19.10 | Ariel_ | 24 channels 31 channels |
| 02:19.40 | zu | that was it I borrowed a neighboors chainsaw and 6 hours later I had cut a path for my car to the road |
| 02:19.48 | Ariel_ | E1 31 don't for get the dchannel |
| 02:20.24 | Ariel_ | zu, I fell off the a tree cutting it down after wilma. I was layed out for almost 2 weeks... it sucks |
| 02:21.15 | *** join/#asterisk kart_179 (n=kart@200.103.160.41) |
| 02:21.31 | kart_179 | Hey i have problem with my R2, anyone can help me ? |
| 02:21.46 | zu | wow where I am in broward it took me with one chainsaw another neighbor with a chainsaw, 6 guys and 1 truck and 6 hours to clear a path to a main road |
| 02:22.03 | Ariel_ | strange the asterisk user list is so slow... I normally get about 200 plus emails every day from it. |
| 02:22.10 | kart_179 | Hey i have problem with my R2, anyone can help me ? |
| 02:22.38 | Ariel_ | R2 hummmm now there is something I have not worked with.... sorry kart_179 |
| 02:23.29 | javier | I just tried it, let the phone ring but I don't see the caller ID. I sounds like Asterisk is picking up the analog trunk on the first ring. |
| 02:24.03 | Ariel_ | asterisk will let it ring normally. |
| 02:24.21 | kart_179 | Ariel_ :( |
| 02:24.26 | kart_179 | Hey i have problem with my R2, anyone can help me ? |
| 02:24.26 | Ariel_ | you have usecallerid=yes which tells zaptel to let it ring |
| 02:24.28 | co-bdg^-^ | isdn e1 and g.703 adapter can connect pbx system to asterisk system ... is that true ? |
| 02:24.48 | *** join/#asterisk EriSan (n=erisan@81-174-42-154.f5.ngi.it) |
| 02:25.05 | Ariel_ | co-bdg^-^, well yes like a TE110p from digium |
| 02:25.44 | zu | yea e1 cards have hardware echo supression, no crappy software algorithms |
| 02:26.05 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 02:26.17 | iccomputing | does anyone have any experience troubleshooting mysql errors on "asterisk reload" ?? |
| 02:26.40 | zu | iccomputing: do you have paypal |
| 02:26.45 | iccomputing | yop |
| 02:26.50 | iccomputing | what do you charge? |
| 02:26.57 | Ariel_ | what is the error your getting |
| 02:27.11 | iccomputing | a few...can i pastebin? |
| 02:27.24 | Ariel_ | sure |
| 02:27.25 | kart_179 | Hey i have problem with my R2, anyone can help me ? |
| 02:27.45 | Ariel_ | kart_179, the more you post it the less people will respond. give it some time... |
| 02:28.25 | Ariel_ | kart_179, and for R2 you should try the user list first since there are more people there from overseas. |
| 02:29.12 | iccomputing | http://pastebin.com/514038 |
| 02:29.54 | Coccyx | terminology might help too, r2 is a signalling protocol, e1 is the actual circuit |
| 02:30.00 | Coccyx | framing actually |
| 02:30.05 | *** join/#asterisk YoMama (n=tchen@pcp02689850pcs.roylok01.mi.comcast.net) |
| 02:30.08 | YoMama | bah |
| 02:30.40 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 02:30.42 | tengulre11 | HI,all |
| 02:30.44 | reza | why are the x100p cards so expensive? i bought mine for $15 |
| 02:30.45 | YoMama | hi |
| 02:30.46 | Ariel_ | iccomputing, it's telling what's wrong |
| 02:30.47 | Math` | YoMama: hi |
| 02:30.53 | *** part/#asterisk iccomputing (n=Wireless@cpe-69-133-109-130.woh.res.rr.com) |
| 02:30.56 | YoMama | reza: yours is probably a clone |
| 02:30.59 | YoMama | hey Math! long time no see |
| 02:31.01 | *** join/#asterisk iccomputing (n=Wireless@cpe-69-133-109-130.woh.res.rr.com) |
| 02:31.08 | iccomputing | whoops!! i clicked close!! |
| 02:31.11 | reza | how can you tell? |
| 02:31.14 | YoMama | Math`: I ended up buying a locked PAP2 and converting that :) |
| 02:31.39 | iccomputing | Ariel, did you see my pastebin? |
| 02:31.40 | YoMama | reza: does it actually say Digium on it? |
| 02:31.42 | reza | pfn - there you are, been looking for you |
| 02:31.47 | tengulre11 | reza: how to buy? I m in china! |
| 02:31.54 | tengulre11 | I m point X100p card! |
| 02:32.08 | YoMama | anyone here good with fixing echo problems? |
| 02:32.15 | Coccyx | I see one available for $25 |
| 02:32.15 | reza | teng- i bought it off digium's website |
| 02:32.23 | *** join/#asterisk xxi (i=foobar@cpe-70-112-73-77.austin.res.rr.com) |
| 02:32.24 | reza | 1 x OEM X100P - FXO PCI Card (DIGITX100P) = $14.50 |
| 02:32.27 | Coccyx | YoMama: dunnoa bout good, but I just spent 4 days doing it |
| 02:32.42 | Coccyx | echocan preload patch finally got it for me |
| 02:32.42 | reza | yomama- dont see any markings |
| 02:32.46 | Ariel_ | iccomputing, it's telling what's wrong |
| 02:32.46 | YoMama | Coccyx: analog port? |
| 02:32.47 | Coccyx | it's a brand new patch |
| 02:32.54 | YoMama | reza: i dunno..maybe it's real |
| 02:32.58 | Coccyx | YoMama: yeah, had echo problems on 3 analog CO lines |
| 02:33.07 | YoMama | i just know there's lots of X100P clones out there |
| 02:33.13 | Coccyx | I was pretty frustrated, I'll tell you that |
| 02:33.16 | YoMama | Coccyx: and...how did u solve it? |
| 02:33.29 | *** join/#asterisk _cleric_ (n=dacleric@p5482974C.dip0.t-ipconnect.de) |
| 02:33.29 | reza | either way, glad i bought it. what does it do, btw? is it an fxs? |
| 02:33.30 | Coccyx | YoMama: why don't you pastebin me your zapata.conf? |
| 02:33.37 | Coccyx | reza: FXO |
| 02:33.46 | Coccyx | it takes 1 CO line |
| 02:34.04 | reza | so what could i use it for? (i bought it because it was cheap) |
| 02:34.04 | Coccyx | YoMama: that and I applied the new echocan preload patch, which is what really helped. |
| 02:34.19 | Coccyx | reza: plug your phone line from the phone company into it and then use a SIP hardphone or softphone to answer it |
| 02:34.30 | Coccyx | using asterisk as the softpbx obviously. |
| 02:34.41 | Coccyx | build a voicemail system, whatever, lots of stuff you can do. |
| 02:34.44 | reza | ah.. thnx |
| 02:34.58 | YoMama | Coccyx: http://pastebin.ca/37494 |
| 02:34.58 | reza | ok, now to figure out how to get this tdm400p to work |
| 02:35.05 | reza | *grumble* *grumble* |
| 02:35.16 | YoMama | Coccyx: i know the rxgain is quite high..but if i don't turn it up..i can barely hear my voicemails |
| 02:35.17 | Coccyx | reza: why did you buy a x100p if you already had a tdm400p? |
| 02:35.23 | reza | bought them at the same time |
| 02:35.27 | tengulre11 | reza: what 's difference between OEM x100p and gernic 100p! |
| 02:35.31 | YoMama | reza: I'll buy your TDM400P off of you for $5 :-P |
| 02:35.36 | reza | i wanted the tdm400p, but the x100p was only $15 |
| 02:35.48 | reza | 1 x DigiumTM TDM400P () = $180.00 |
| 02:35.48 | Coccyx | YoMama: yeah, it's going to be hard to solve echo through attenuation with a rxgain that high |
| 02:35.52 | reza | i paid a bit more for that |
| 02:35.56 | Coccyx | what echocan algorithm are you using for zaptel? |
| 02:36.05 | YoMama | Coccyx: umm..lemme see |
| 02:36.22 | Coccyx | YoMama: I recommend MG2 with this new patch, let me find the list posting... |
| 02:36.28 | Qwell | tengulre11: There is no such thing as an "oem digium" card |
| 02:36.32 | Ariel_ | KB1 or MG2 for me |
| 02:36.35 | YoMama | Coccyx: whatever is standard..i think it's mark2 |
| 02:36.54 | Ariel_ | last it was kb1 |
| 02:37.00 | Ariel_ | if you have 1.2 |
| 02:37.10 | YoMama | i do |
| 02:37.14 | YoMama | umm..lemme look |
| 02:37.17 | Coccyx | YoMama: 1.2 is KB1 |
| 02:37.23 | Coccyx | you'll want to go to MG2 |
| 02:37.28 | Ariel_ | so edit zconfig.h |
| 02:37.30 | Coccyx | and you'll want to apply this patch for zaptel: http://www.sineapps.com/news.php?rssid=1190 |
| 02:37.35 | tengulre11 | Qwell: nice! I need building a simple VOIP platform, what hardware me need? |
| 02:37.56 | Coccyx | tengulre11: for one line, a X100P would do you fine. |
| 02:37.57 | tengulre11 | s/need/ want to/ |
| 02:38.16 | Ariel_ | hardware simple one pc hdd memory and at least one phone |
| 02:38.16 | *** join/#asterisk ptiggerdine (n=ptiggerd@c220-237-93-88.rochd1.qld.optusnet.com.au) |
| 02:38.23 | tengulre11 | Coccyx: where can buy it? |
| 02:38.30 | tengulre11 | I m in china! |
| 02:38.30 | reza | pfn - you here? |
| 02:38.32 | Coccyx | tengulre11: google is a good place to start looking :) |
| 02:38.37 | tengulre11 | :( |
| 02:38.40 | Coccyx | tengulre11: ah, dunno about china. |
| 02:38.49 | Coccyx | tengulre11: your guess is as good as mine. |
| 02:38.54 | Coccyx | probably better actually. |
| 02:39.02 | Coccyx | YoMama: have you compiled zaptel yourself yet? |
| 02:39.19 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 02:39.24 | reza | ok, i'm confused. i got 2 of the green modules fo rthe tdm card - that's what i plug phones into, right? and they're called fxs's? |
| 02:39.39 | Coccyx | yes |
| 02:39.41 | Ariel_ | tengulre11, http://www.openvox.com.cn/products.php?genre_id=13 |
| 02:39.47 | Ariel_ | there in your location |
| 02:39.53 | *** part/#asterisk zahid (n=chatzill@user-0cdf50g.cable.mindspring.com) |
| 02:40.01 | reza | so why are they called fxoks in the zaptel conf file? |
| 02:40.14 | tengulre11 | Ariel: do u have MSN ? |
| 02:40.21 | Ariel_ | Rez, yes correct red ones are for phone lines |
| 02:40.28 | Ariel_ | tengulre11, yes but I don't give it out |
| 02:40.28 | YoMama | Coccyx: but of course |
| 02:40.31 | Qwell | reza: because fxs uses fxo signalling |
| 02:40.46 | reza | qwell - do they try to make this as hard as possible? |
| 02:40.49 | *** join/#asterisk dijit0 (n=dijit0@69.106.48.57) |
| 02:40.50 | YoMama | Coccyx: my phone company sucks..i know hte gains aren't helping...but without those gains..i can't hear shit |
| 02:40.54 | YoMama | yo Qwell |
| 02:40.56 | Qwell | no, it's simple |
| 02:41.00 | Ariel_ | Reza yes |
| 02:41.05 | reza | :P |
| 02:41.06 | tengulre11 | Ariel_: are you OPENVOX@xxxx.com? |
| 02:41.07 | Coccyx | YoMama: you go to that URL I pasted above? |
| 02:41.14 | Ariel_ | tengulre11, no |
| 02:41.14 | YoMama | Coccyx: yeah..i'm gonna give it a shot |
| 02:41.17 | Ariel_ | I am in the us |
| 02:41.20 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 02:41.22 | Coccyx | YoMama: you need to apply that patch. It solved my problems, really. |
| 02:41.24 | dijit0 | quick question, how can i login to my linux box from windows? |
| 02:41.30 | Coccyx | YoMama: if you hang on the line about 30 seconds does the echo go away? |
| 02:41.36 | reza | so the exampels on digium's site all assume at least one fxo card. is there anything that needs to be changed other than removig the fxsks= line? |
| 02:41.53 | reza | dijit0 - download putty (google putty ssh) |
| 02:42.11 | reza | sshd should be running on your linux box |
| 02:42.22 | dijit0 | ahh nice... thx |
| 02:42.35 | dijit0 | and as far as doing a remote desktop connection, is that possible in windows? |
| 02:42.40 | YoMama | Coccyx: yup |
| 02:42.46 | YoMama | Coccyx: it has to retrain every single time |
| 02:42.50 | Coccyx | YoMama: then this will solve it for you. |
| 02:42.52 | YoMama | Coccyx: it goes away after 10-20 seconds |
| 02:42.56 | Ariel_ | dijit0, no |
| 02:43.07 | Coccyx | YoMama: you know how to apply patches? |
| 02:43.19 | YoMama | Coccyx: yeah..this looks like it might work...question though...how come u don't specify the file you preload? |
| 02:43.28 | YoMama | Coccyx: step #4 doesn't seem right |
| 02:43.36 | zu | cd /usr/src/asterisk ; patch -p1 < ./patchname |
| 02:43.42 | YoMama | Coccyx: yeah..i'm Unix smart..just Asterisk stupid |
| 02:43.52 | dijit0 | grr... i wouldnt need to do it in windows if my wireless card was supported... everything works but my wireless card... and i dont know enough to properly set that up, since there are no linux drivers for it |
| 02:43.55 | Coccyx | YoMama: awesome |
| 02:43.57 | Ariel_ | dijit0, ssh or putty will give you a cli prompt from the asterisk box.. no xwindow which should not be loaded on an asterisk box |
| 02:43.57 | Coccyx | YoMama: it is wrong |
| 02:44.08 | Coccyx | you need to do a zt_ec_preload -d <channel> < datadumpfile |
| 02:44.24 | SkramX | anyone ben able to stream a meetme conference to shoutcast? |
| 02:44.31 | Coccyx | so whatevber file you dump the results from the other into then load it in step 4 from that file |
| 02:44.36 | Coccyx | it solved my problems, this patch was a godsend. |
| 02:44.39 | *** join/#asterisk shido (n=shido@d221-68-216.commercial.cgocable.net) |
| 02:44.59 | Coccyx | YoMama: you'll have to reload that data on reboot though |
| 02:45.00 | YoMama | Coccyx: that's what i thought...thanks man..i hope they build this into the source..why wouldn't u wanna preload your training data? |
| 02:45.11 | YoMama | Coccyx: nothing a little script won't fix |
| 02:45.17 | Coccyx | YoMama: it wouldn't work very well on digital lines... |
| 02:45.24 | Coccyx | YoMama: it only helps with analog where the echo is consistent. |
| 02:45.28 | YoMama | Coccyx: i doubt i'd be having these types of problems on digital lines |
| 02:45.36 | *** join/#asterisk zimdog (n=zimdog@c-24-9-24-165.hsd1.co.comcast.net) |
| 02:45.39 | Coccyx | on a PRI you're going to get different echo depending on where the call comes in |
| 02:45.51 | Coccyx | YoMama: you wouldn't :) |
| 02:46.03 | YoMama | Coccyx: true..i'm actually more verse at handling PRIs than stupid analog lines |
| 02:46.03 | Coccyx | YoMama: unless your PRI is running more than about 100 miles to the CO |
| 02:46.10 | Coccyx | which is very unusual if you're not a phone company already. |
| 02:46.19 | YoMama | Coccyx: and those people live out in the woods and should know better |
| 02:46.21 | Coccyx | I used to work for a cellular carrier... echo on digital spans is totally different. |
| 02:46.38 | Coccyx | of course then I could afford TEllabs gear :) |
| 02:46.51 | Coccyx | I'm really disappointed in asterisk's echocans... they're totally unacceptable. |
| 02:47.03 | Coccyx | but this patch is really helping. |
| 02:47.20 | Coccyx | I've heard digium's cards are the only ones that have this kind of problem though. |
| 02:47.33 | Coccyx | other ATAs have better hardware echocan. |
| 02:47.39 | *** join/#asterisk bweschke (n=bweschke@pcp09754274pcs.narlington.nj.comcast.net) |
| 02:48.08 | Ariel_ | Coccyx, well actually others have it as well. But dialogic firs board were not full duplex due to the echo problem. |
| 02:48.16 | Coccyx | I'm concerned because I want to sell this system in town and I really need something thats as clean as a traditional key system. |
| 02:48.30 | YoMama | Coccyx: it makes sense to deal with it on the hardware anyway |
| 02:48.44 | zimdog | Hello room, What dies SIP response 604 "Does not exist anywhere" mean? |
| 02:48.47 | Coccyx | YoMama: yeah, I'm disappointed the TDM400 doesn't have it in hardware. |
| 02:48.50 | YoMama | Coccyx: I hear ya...the handset support is iffy at times too |
| 02:49.09 | Coccyx | it's pretty processor intensive actually... DSPs do much better, which is why most have it in hardware. |
| 02:49.16 | reza | coccyx - what board are you using? |
| 02:49.23 | Coccyx | reza: TDM400 |
| 02:49.27 | YoMama | Coccyx: yeah...but the whole point of asterisk is not to have DSPs :) |
| 02:49.35 | reza | same one then; what problem did you have and what fixed it? |
| 02:49.41 | FuriousGeorge | the difference b/w congestion and busy is that congestion means all my resources are tied up? |
| 02:49.42 | Coccyx | YoMama: the phones have DSPs in them, so do most channel banks :) |
| 02:49.49 | Ariel_ | the tdm400 work well but you have to make sure irq is not shared as well |
| 02:50.00 | Coccyx | just no DSPs at the core, all the transcoding logic is at the edge. |
| 02:50.01 | YoMama | Coccyx: yeah..they really ain't that expensive anymore |
| 02:50.04 | reza | hmm. how do you do that w/ linux? |
| 02:50.06 | Ariel_ | congestion is busy |
| 02:50.22 | Coccyx | reza: I had a serious echo problem... fixed it with this echocan preload patch. |
| 02:50.27 | FuriousGeorge | Ariel_: SO WHY CAN ${dialstatus} be either |
| 02:50.28 | Coccyx | scroll up :) |
| 02:50.31 | *** join/#asterisk _cleric_ (n=dacleric@p5482974C.dip0.t-ipconnect.de) |
| 02:50.38 | Coccyx | Ariel_: yeah, that didn't make a difference in my case. |
| 02:50.41 | reza | can you mail me a copy? |
| 02:50.44 | Coccyx | neither did the PC, I switched to another PC. |
| 02:51.11 | reza | or paste a url |
| 02:51.30 | Ariel_ | Coccyx, I have setup over 50 systems with TDM board in them some do get echo some don't it's also dependent on the telco side. |
| 02:52.02 | zu | yea echo=set it to agressive |
| 02:52.08 | *** join/#asterisk dav0_ (n=dcb@CPE-144-131-46-226.vic.bigpond.net.au) |
| 02:52.16 | Ariel_ | but I do belive that digium could have done better on there boards with echo can. But they wanted to do this cheaply. |
| 02:52.16 | YoMama | dammit..i can't count today |
| 02:52.28 | *** join/#asterisk _cleric_ (n=dacleric@p5482974C.dip0.t-ipconnect.de) |
| 02:52.45 | FuriousGeorge | Ariel_: er, i mean if busy and congestion are one and the same why does the dialstatus variable differentiate |
| 02:53.10 | Ariel_ | congestion is sent out. busy is detected |
| 02:53.19 | FuriousGeorge | gotcha |
| 02:53.37 | zu | yea sangoma has tdm cards with hardware echo suppresion optionally |
| 02:53.46 | YoMama | Coccyx: what area are u in? |
| 02:53.48 | FuriousGeorge | Ariel_: was it you saying not to use busy detextion in zapata.conf yesterday |
| 02:53.56 | Ariel_ | no |
| 02:54.15 | FuriousGeorge | i meant to ask why :) |
| 02:54.16 | Ariel_ | sangoma seems to have a better echo can on there boards. |
| 02:54.16 | Coccyx | zu: aggressive echo can causes a lot of chop for me |
| 02:54.21 | Coccyx | YoMama: Fort Smith, ARkansas. |
| 02:54.23 | reza | ariel - is the echo mostly on receiving calls or making voip calls? |
| 02:54.30 | YoMama | Coccyx: lots of small businesses down there i bet |
| 02:54.51 | Ariel_ | echo is you hearing yourself at either a few seconds after you speak |
| 02:54.54 | Coccyx | YoMama: yeah, a lot... this system could be very successful in town if I can clear up the problems. |
| 02:54.58 | *** join/#asterisk _mountie (n=mountie@trb229.travel-net.com) |
| 02:55.06 | Coccyx | YoMama: lots of benefits for businesses with multiple locations in different towns as well. |
| 02:55.09 | FuriousGeorge | i thought echo can on a tdm was all software |
| 02:55.17 | Ariel_ | Coccyx, arkansas... I used to live up in the Cabot area |
| 02:55.25 | Coccyx | lots of local phone companies around here that charge LD for a town 5 miles away. |
| 02:55.36 | *** join/#asterisk seeeexy_girl_06 (n=seeeexy_@c-67-181-117-151.hsd1.ca.comcast.net) |
| 02:55.39 | Coccyx | Ariel_: I've been there. Used to work for a company called IPA that had a Internet POP in a bank there. |
| 02:55.45 | YoMama | Coccyx: yeah..i see the big success of Asterisk for small businesses...'cause small businesses can't afford a PBX with voicemail, conferenceing, etc...plus they charge an arm and a leg for the stupid handsets to those systems |
| 02:56.01 | Coccyx | YoMama: usually $150-$200 for a full featured digital handset. |
| 02:56.08 | Coccyx | plus about $2 to $3k for the "magic box" |
| 02:56.14 | Coccyx | if it includes voicemail |
| 02:56.19 | Ariel_ | Coccyx, I was in the Air Force back then... |
| 02:56.19 | Coccyx | generally flash based voicemail can store 3 hours of data. |
| 02:56.21 | YoMama | Coccyx: that's barebones |
| 02:56.27 | Coccyx | YoMama: yeah. |
| 02:56.32 | reza | this 400mhz celeron was never meant to compile anything; so painful to watch |
| 02:56.42 | Coccyx | YoMama: I'm building my systems on VIA EPIA boards booting asterisk off of flash |
| 02:56.49 | Coccyx | it's pretty badass... |
| 02:57.03 | YoMama | Coccyx: what distro of linux u using? |
| 02:57.06 | Coccyx | 1 Ghz is plenty for most small systems, if they don't do a lot of transcoding. |
| 02:57.09 | YoMama | Coccyx: yeah..that'd be the right way to do it |
| 02:57.13 | Coccyx | YoMama: right now gentoo, but I'm moving to astlinux. |
| 02:57.21 | Ariel_ | via epia sometimes have problems with there internal p/s and the digium board are you using the one with the external p/s |
| 02:57.29 | reza | astlinux? |
| 02:57.37 | Qwell | astlinux is cool |
| 02:57.43 | reza | hmm.. google time |
| 02:57.44 | Qwell | not complete crap like aah :) |
| 02:57.46 | Coccyx | Ariel_: yeah, external PS... not grounded though because it's DC, you have to ground the box manually |
| 02:57.59 | Coccyx | otherwise you can end up with a hum on the lines. |
| 02:58.09 | Coccyx | important to have access to the building ground. |
| 02:58.18 | Coccyx | right now i"m running groundless until I can rewire some stuff. |
| 02:58.34 | Ariel_ | I started to use a larger box from msi and going to put together a combo firewall/asterisk pbx for small biz in the next few weeks on the market. |
| 02:58.52 | Coccyx | Ariel_: funny that's what we're going to be selling. |
| 02:59.00 | Coccyx | openvpn on the boxes too for point to point VPN tunnels. |
| 02:59.06 | seeeexy_girl_06 | hi i need help with my asterisk/nufone delema |
| 02:59.13 | Libila | Qwell: whats wrong with aah |
| 02:59.20 | Ariel_ | what a name |
| 02:59.21 | *** join/#asterisk nrl[digium] (n=nlewis@12.158.129.130) |
| 02:59.22 | Coccyx | I left a job at a major cellular company to move to arkansas and sell this shit. That's how much I believe in asterisk. |
| 02:59.28 | Ariel_ | nufone works fine what is the problem |
| 02:59.32 | Qwell | Libila: too many things to mention... |
| 02:59.42 | Ariel_ | Coccyx, great to hear it. |
| 02:59.43 | Coccyx | I'm selling other stuff too... I"m selling zimbra as a replacement for exchange as well. |
| 02:59.45 | Math` | seeeexy_girl_06: are you coming for the same problem as yesterday? |
| 02:59.56 | seeeexy_girl_06 | i use it for displaying caller id of my choice but out of no where it stops working... |
| 03:00.02 | *** join/#asterisk masked (n=masked@static-203-87-16-192.vic.chariot.net.au) |
| 03:00.08 | Libila | Qwell: I'm curious, I normally use gentoo/bsd for my os's. then one of my friends said to use aah, so instead of installing gentoo like normal I installed aah a few hours ago. |
| 03:00.09 | Ariel_ | I have been on my own doing this for 1.5 years now I used to setup asterisk for another co before that. |
| 03:00.26 | Coccyx | Ariel_: any telephony experience prior to that? |
| 03:00.26 | Ariel_ | seeeexy_girl_06, did you upgrade asterisk ??? |
| 03:00.43 | iCEBrkr | seeeexy_girl_06: A/S/L? |
| 03:00.46 | seeeexy_girl_06 | im using the windows one on here sadly |
| 03:00.47 | Math` | lol |
| 03:00.49 | reza | astlinux & aah? |
| 03:00.51 | iCEBrkr | Math`: I had to. |
| 03:00.53 | seeeexy_girl_06 | 18/f/cali |
| 03:00.54 | Ariel_ | Coccyx, no I was into ivr's for hotels and phone networks for them all privite |
| 03:00.56 | Coccyx | I've deployed and been schooled on Nortel Meridian systems and Lucent Definitey systems. |
| 03:01.00 | iCEBrkr | seeeexy_girl_06: LOL |
| 03:01.03 | Math` | argh cali, too far |
| 03:01.08 | Math` | no help for you |
| 03:01.12 | mgoh | got any dialer for asterisk that only allow agent to dial from address books? |
| 03:01.25 | Coccyx | so I'm pretty picky when it comes to phone systems, I've used the ones you pay big bucks for, but asterisk is pretty close to being there. |
| 03:01.37 | Math` | mgoh: no but you can create a context with only the numbers allowed to be dialed and set the agent into that context |
| 03:01.38 | Ariel_ | Coccyx, I know them well. since we were dealing with hotels like marriot and others like that. |
| 03:01.50 | seeeexy_girl_06 | what do you think the problem could be ariel? |
| 03:01.59 | Coccyx | It's really mostly a handset issue... the handsets on the market are pretty poor... I'm hopeful for the new Linksys/Sipura phones to come close to Cisco quality on the handsets. |
| 03:02.21 | Ariel_ | seeeexy_girl_06, what version are you using 1.0.9 or 1.2? |
| 03:02.23 | Coccyx | configuring asterisk sure beats the hell oughta learning Nortel Meridian... all that load shit and 4 letter abbreviations is for the birds. |
| 03:02.24 | mgoh | math': where can we create a context insdie asterisk? |
| 03:02.26 | Math` | Ariel_: little question since you know "old" systems, if I get a PRI card for a Meridian system and plug it into a TDM card, will it work fine? |
| 03:02.33 | Math` | mgoh: in extensions.conf |
| 03:02.41 | masked | i have a little problem installing zaptel, i've been trying to nut it out myself but i can't, i'm hoping one of you may have a better idea whats wrong/missing - /bin/sh: -c: line 0: syntax error near unexpected token `;' |
| 03:02.42 | masked | /bin/sh: -c: line 0: `if -- etc etc until make: *** [install] Error 2 |
| 03:02.50 | Ariel_ | Math`, not witout so changes on the meridian |
| 03:02.58 | Ariel_ | so/some |
| 03:03.01 | Coccyx | Ariel_: you'll have a problem unless one of them is serving the D Channel... |
| 03:03.05 | iccomputing | zu++ |
| 03:03.09 | Coccyx | err Math` |
| 03:03.13 | seeeexy_girl_06 | its asteriskwin32 0.5.2 |
| 03:03.24 | Ariel_ | Coccyx, aslo need to the right software load |
| 03:03.24 | Coccyx | you'll be easier running channelized T1 between them. |
| 03:03.31 | Ariel_ | seeeexy_girl_06, argh |
| 03:03.32 | zu | ~karma zu |
| 03:03.32 | jbot | zu has neutral karma |
| 03:03.40 | iCEBrkr | ASSWIND? |
| 03:03.47 | seeeexy_girl_06 | lol i know but i cant install linux on this comp.... |
| 03:03.47 | Coccyx | most systems support generating robbed bit signalling but not ISDN PRI signalling. |
| 03:04.05 | Coccyx | I don't know if Asterisk can serve a PRI or not. |
| 03:04.20 | masked | seeeexy_girl_06: install linux into a virtual machine. |
| 03:04.20 | iccomputing | zu i closed the chat!! |
| 03:04.22 | Math` | can a meridian serve a pri? |
| 03:04.28 | Ariel_ | seeeexy_girl_06, nufune has updated there setups that could be your problem and if you call them they will tell you to upgrade asterisk |
| 03:04.39 | Ariel_ | Coccyx, yes |
| 03:04.40 | mgoh | Math': can the extensions.conf real time reflect back when I make a change? |
| 03:04.44 | iCEBrkr | Math`: I think a lot of PBX's are able to 'provide' PRI service *Shrug* |
| 03:04.44 | Qwell | Ariel_: it's a dialplan thing |
| 03:04.46 | seeeexy_girl_06 | come on guys help me... my ex boyfriend is an asshole and i need to call his girlfriends cell phone with his other ex-girlfriends cell |
| 03:04.50 | Qwell | Ariel_: we went over this last night |
| 03:05.04 | Qwell | She's dumb. That's all... |
| 03:05.11 | Ariel_ | argh cell phone hell... |
| 03:05.16 | YoMama | sexy_girl: huh? |
| 03:05.25 | iCEBrkr | seeeexy_girl_06: LOLOLOLOL |
| 03:05.35 | *** join/#asterisk jef_ (i=fischer@p54847385.dip.t-dialin.net) |
| 03:05.39 | seeeexy_girl_06 | http://profiles.yahoo.com/seeeexy_girl_06 |
| 03:05.41 | seeeexy_girl_06 | stfu |
| 03:05.44 | reza | does anyone know how traditional answering services work? |
| 03:05.45 | zu | hya Qwell |
| 03:05.49 | Ariel_ | seeeexy_girl_06, post your dial string on pastebin.ca so we can see it. |
| 03:05.52 | Math` | mgoh: extensions's realtime is.... how can I say.... realtime! |
| 03:05.56 | seeeexy_girl_06 | alright |
| 03:05.58 | YoMama | my best friend's sister's cousin's brother-in-law told his father's sister's grandfather that her mother's grand aunt's daughter was quite upset |
| 03:06.06 | reza | somehow they get a clone of the same phone number or somesuch... |
| 03:06.07 | masked | seeeexy_girl_06: get the ex's cell and ring her! |
| 03:06.14 | Qwell | Ariel_: she did yesterday. She's using some UGLY hack...and is never even setting cid |
| 03:06.23 | Qwell | We already told her this last night, but SOMEBODY didn't pay attention |
| 03:06.43 | Ariel_ | ok |
| 03:06.47 | Qwell | and the whole astwin32 thing |
| 03:06.53 | Math` | somebody? |
| 03:06.55 | Qwell | her |
| 03:06.58 | Math` | :P |
| 03:07.10 | Ariel_ | Math`, I don't use realtime due to it has issues |
| 03:07.14 | Math` | last thing I remember is trying to explain how to copy paste from the CLI |
| 03:07.17 | iCEBrkr | Astwind should work just like Asterisk on Linux. |
| 03:07.22 | iCEBrkr | I got it running, but never tinkered with it. |
| 03:07.24 | Ariel_ | besides I think asterisk does very well with reloads |
| 03:07.27 | Math` | Ariel_: it's becoming more and more stable |
| 03:07.28 | Qwell | Ariel_: What issues have you seen with realtime? There have been a bunch of fixes lately |
| 03:07.38 | YoMama | u know what i love about chix who come on IRC with nicknames like sexy_girl? |
| 03:07.40 | *** join/#asterisk inv_Arp (n=junya@c-66-176-211-109.hsd1.fl.comcast.net) |
| 03:07.48 | Qwell | YoMama: They're usually fat, balding, old men? |
| 03:07.57 | Qwell | (or women...) |
| 03:07.58 | FuriousGeorge | maybe someone knows the answer to this: i got 5 asterisk boxes. one cant register with the other. it just times out. the ip is right and everything |
| 03:08.07 | Ariel_ | Math`, fine but I don't play games with my systems. They are for real customers and I can't have them down. |
| 03:08.09 | YoMama | Qwell: hahahaha...nah..they're usually women, but they're rarely what most men would consider sexy |
| 03:08.17 | Qwell | YoMama: see above |
| 03:08.21 | mgoh | Math': if I intergrate with others app that update extension.con. Can the agents dial after tat app done updating. |
| 03:08.26 | FuriousGeorge | heres the catch, they all have dynamic ips so i use box1.dyniptodns.com service |
| 03:08.40 | Ariel_ | dns cache is off |
| 03:08.50 | FuriousGeorge | if i do iax show registry the ip is correct, but it just times out |
| 03:08.53 | masked | FuriousGeorge: tried static dhcp? |
| 03:09.18 | FuriousGeorge | static dhcp? there is nat between all these boxes, they arent on the same network. |
| 03:09.34 | Math` | uh oh |
| 03:09.42 | Math` | you said the Bad Word(tm) |
| 03:09.43 | Math` | (nat) |
| 03:09.50 | rob0 | What's *really* funny is to see the lonely horny boys stepping on their tongues trying to help a "sexy_girl". |
| 03:09.56 | reza | is there any advantage other than simplifying configs to using aah? |
| 03:09.57 | Ariel_ | 5 boxes and nat and none with real IP's.... and your having only one with issues... |
| 03:10.00 | FuriousGeorge | but ports are forwarded. all of em have sip clients logged into them, they are all logged into eachother and to their respective ips. its just these two boxes that wont connect via iax |
| 03:10.29 | Ariel_ | rob0, I try to help all does not need to be a girl.... but it does help. |
| 03:10.35 | seeeexy_girl_06 | rob0 you may make your own oppinion on my looks... i just seem to have what you lack... and that is SELF CONFIDENCE |
| 03:10.53 | rob0 | :) |
| 03:10.56 | Qwell | and she's still not paying attention to the people are ARE trying to help... |
| 03:11.08 | seeeexy_girl_06 | i am... |
| 03:11.11 | seeeexy_girl_06 | its just running slow |
| 03:11.15 | seeeexy_girl_06 | hold on |
| 03:11.26 | seeeexy_girl_06 | http://pastebin.ca/37496 |
| 03:11.28 | seeeexy_girl_06 | there |
| 03:11.30 | Qwell | and you STILL never showed me a damn cli output from a call |
| 03:11.37 | Math` | lollll |
| 03:11.51 | seeeexy_girl_06 | I DONT HAVE LINUX REMEMBER... |
| 03:11.54 | YoMama | i'm horny..not too lonely though |
| 03:12.02 | seeeexy_girl_06 | at least not on here |
| 03:12.08 | Ariel_ | seeeexy_girl_06, you do have an cli with astwind as well |
| 03:12.08 | Qwell | seeeexy_girl_06: That's great...it has a fucking console |
| 03:12.09 | YoMama | whoever made astwin should be shot |
| 03:12.15 | Math` | oh and how about I tall you that..... windows ports of asterisk aren't supported here |
| 03:12.22 | Math` | s/tall/tell/ |
| 03:12.37 | rob0 | <== I got help here in this channel today :) |
| 03:12.39 | Math` | omg that bot is awesome |
| 03:12.41 | seeeexy_girl_06 | i do not know how to post it in here.. if someone would kindly tell me that would be great |
| 03:12.46 | Ariel_ | she want food at this time argh this baby all she wants to do is eat. |
| 03:12.59 | YoMama | <-- got help in here today too |
| 03:13.02 | YoMama | about to try it out |
| 03:13.09 | Qwell | seeeexy_girl_06: umm...the same way you've been posting the other shit...maybe? |
| 03:13.24 | Qwell | Ariel_: how old? |
| 03:13.33 | Ariel_ | 2 |
| 03:13.38 | seeeexy_girl_06 | qwell.. i can open a text file to find this out? |
| 03:13.39 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 03:13.46 | Qwell | seeeexy_girl_06: just give up |
| 03:13.47 | Qwell | seriously |
| 03:13.54 | Qwell | uninstall asterisk...and go play |
| 03:14.02 | zu | yay |
| 03:14.05 | javier | yeah |
| 03:14.05 | zu | windows sucks |
| 03:14.06 | Qwell | if you can't read SIMPLE documentation... |
| 03:14.09 | YoMama | it's so hot when a girl talks about opening up a text file... |
| 03:14.09 | seeeexy_girl_06 | qwell give up on insulting me |
| 03:14.15 | zu | windows is the main reason the inter net is slow |
| 03:14.18 | Coccyx | rob0: well, they could go to a mailing list, but at least with IRC they don't have to deal with 100s of emails a day :) |
| 03:14.22 | Coccyx | zu: that's shit. |
| 03:14.50 | YoMama | zu: as much as i hate windoze and microsoft...that's a pretty inaccurate generalization |
| 03:14.51 | Ariel_ | damm seeeexy_girl_06 that is one really messed up dial plan |
| 03:14.52 | reza | sexy - you would have less drama if you changed your nick. |
| 03:15.00 | Qwell | Ariel_: note that the top part... |
| 03:15.02 | zu | Coccyx: the amount of spam from zombie winblows machines have shutdown fortune 100 companies so shut the fuck up |
| 03:15.03 | Qwell | [globals] |
| 03:15.04 | javier | I agree |
| 03:15.10 | Qwell | she omitted that this time |
| 03:15.13 | zimdog | What does SIP response 604 "Does not exist anywhere" mean? |
| 03:15.19 | zu | I have a MCSE/MCSA/MCDBA and have built windows clusters how about you |
| 03:15.26 | Qwell | Ariel_: I'll give you one guess where the call is actually coming in |
| 03:15.32 | YoMama | zu: I'm Bill Gates |
| 03:15.33 | seeeexy_girl_06 | yes i have global variables |
| 03:15.34 | Math` | zimdog: I think it means the extension doesnt exist anymore |
| 03:15.36 | Ariel_ | I see it |
| 03:15.45 | seeeexy_girl_06 | but i figured that was self explanitary |
| 03:15.48 | Qwell | and note this little gem |
| 03:15.49 | Qwell | exten => s,2,SetCallerID() |
| 03:15.50 | Math` | s/anymore/anywhere/ |
| 03:15.57 | Coccyx | zu: I've built windows machines attached to petabyte SANs serving gigabytes of data... my dick is longer. |
| 03:15.58 | *** join/#asterisk iccomputing (n=Wireless@cpe-69-133-109-130.woh.res.rr.com) |
| 03:15.59 | Ariel_ | just a sec baby is now here let me get her at least some snacks. |
| 03:16.11 | Qwell | seeeexy_girl_06: here, you want this fixed? |
| 03:16.11 | YoMama | Coccyx: thansk for the help man..i'm gonna haveta get someone to call |
| 03:16.17 | seeeexy_girl_06 | yes qwell s,2 string is like that because i am able to set it remotely |
| 03:16.18 | zu | YoMama: good your software sucks and allways have I used billgates os on a zorba cpm in 1979 and it still sucks today |
| 03:16.20 | Coccyx | YoMama: I'll call you if you want :) |
| 03:16.21 | Qwell | paypal.com, put in my email address, drop a donation |
| 03:16.25 | zimdog | Thanks Math |
| 03:16.29 | Qwell | then maybe I'll do it for you |
| 03:16.39 | Qwell | or...learn |
| 03:16.39 | zu | </end rant> |
| 03:16.53 | seeeexy_girl_06 | tell me exactly where the hell i can learn more about this... |
| 03:16.56 | Qwell | </end rant> is the same as <rant> |
| 03:16.58 | Qwell | seeeexy_girl_06: google |
| 03:16.59 | seeeexy_girl_06 | ive reached a stand still |
| 03:17.01 | Qwell | ~docs |
| 03:17.02 | jbot | docs is, like, probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 03:17.05 | Qwell | ~wikis |
| 03:17.06 | jbot | from memory, wikis is http://www.voip-info.org |
| 03:17.07 | Qwell | ~mailinglists |
| 03:17.11 | Qwell | ~mailinglist |
| 03:17.13 | jbot | rumour has it, mailinglist is Search Asterisk mailing lists by prepending site:lists.digium.com to your Google search. Browse the mailing list archive at http://lists.digium.com/ or search through it at http://www.asteriskguru.com/archives/search.php, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.html |
| 03:17.17 | Qwell | have fun |
| 03:17.22 | seeeexy_girl_06 | ive read all that is in voip-info.org |
| 03:17.27 | zu | no Qwell its the variable global end of a rant |
| 03:17.40 | Qwell | zu: You ended an end rant tag |
| 03:17.40 | Math` | seeeexy_girl_06: good, now the only thing you have to do is understand it, and apply it |
| 03:17.49 | shido | heh |
| 03:17.49 | seeeexy_girl_06 | ha |
| 03:17.53 | javier | yeah |
| 03:17.56 | rob0 | does that mean the rant continues? |
| 03:17.57 | shido | coming from a person named Math - that was perfect |
| 03:17.58 | YoMama | Coccyx: i'll call my girl instead...hehe :-P |
| 03:18.03 | Qwell | rob0: it means a new one is started...or something |
| 03:18.09 | zu | Qwell: yes I know it was in a xml compatible format though |
| 03:18.12 | Coccyx | zu: point being that traffic as a result of zombie computers is still probably less than 25% of all Internet traffic... if it wasn't windows machines they'd be spending all their time 0wn3ning unix-like systems for their botnets. |
| 03:18.12 | Qwell | the whole double negative thing :p |
| 03:18.20 | Qwell | zu: You wanted a </rant> tag |
| 03:18.28 | zu | Coccyx: Im done dude |
| 03:18.42 | Qwell | wow, 391 people? |
| 03:18.46 | rob0 | come on zu, tell us what you REALLY feel. |
| 03:18.50 | m[afk]ed | can someone tell me more about this: /bin/sh: -c: line 0: syntax error near unexpected token `;' |
| 03:18.56 | zu | Coccyx: one thing I do love about windows is that for the last 16+years its supplied me with a good income |
| 03:19.03 | Qwell | m[afk]ed: not unless we see the line |
| 03:19.04 | seeeexy_girl_06 | well then im just going to give up...youd figure some of you would be willing to help a struggling asterisk user |
| 03:19.08 | seeeexy_girl_06 | HENSE THE CHANNEL NAME |
| 03:19.15 | Qwell | seeeexy_girl_06: struggling...not stupid |
| 03:19.17 | Qwell | no offense |
| 03:19.22 | zu | <rant> insert rant here </end rant> |
| 03:19.24 | maskEd | Qwell: im not sure what line it is because its says line 0 |
| 03:19.27 | Math` | Qwell: 391 here, 90% idle |
| 03:19.32 | Qwell | zu: <rant> insert rant </rant> |
| 03:19.35 | zu | enuff |
| 03:19.38 | mgoh | guy do u all test how reliable asterisk voice recording? Do we really need external voice recording? |
| 03:19.39 | Math` | lol |
| 03:19.40 | Qwell | You can't end a rant tag with an end rant tag! :P |
| 03:19.52 | rob0 | maskEd: to sh, the semicolon is a special token, a command separator |
| 03:19.54 | zu | Qwell: thats not compatibile with my patented rant system |
| 03:19.54 | Math` | mgoh: what do you mean by external voice recording? |
| 03:19.57 | Coccyx | Qwell: yeah, wouldn't be valid XML. |
| 03:19.59 | maskEd | </rant>!!! |
| 03:20.04 | maskEd | <end rant> |
| 03:20.05 | zu | dont make me call the dumbass patent office |
| 03:20.07 | maskEd | now im ending my rant |
| 03:20.09 | maskEd | </end rant> |
| 03:20.15 | mgoh | Math': VOIP recorder |
| 03:20.17 | YoMama | eh? |
| 03:20.21 | seeeexy_girl_06 | well bye |
| 03:20.26 | Math` | mgoh: to record calls? |
| 03:20.27 | YoMama | no..it'd be <rant>blah blah blah</rant> |
| 03:20.28 | Qwell | You can't rant in an end rant tag...it just doesn't work that way |
| 03:20.33 | *** part/#asterisk seeeexy_girl_06 (n=seeeexy_@c-67-181-117-151.hsd1.ca.comcast.net) |
| 03:20.33 | mgoh | Math':Yap |
| 03:20.36 | zu | Hey A polycom phone would load that xml rant without any complaints |
| 03:20.41 | Qwell | she'll be back |
| 03:20.41 | Math` | mgoh: use Monitor() or MixMonitor() |
| 03:20.41 | YoMama | seeexy_girl: sign off and sign back on as a dude...you'll get a lot more help :) |
| 03:20.49 | maskEd | rob0: well theres lots of ;'s in the zaptel Makefile, still not sure which one it is. |
| 03:20.57 | Qwell | YoMama: it's funny, because you're probably right |
| 03:21.01 | Math` | lol |
| 03:21.16 | YoMama | Qwell: her nick made me immediately wanna pick on her |
| 03:21.26 | Qwell | YoMama: her stupidity did it for me |
| 03:21.28 | mgoh | Math': I know asterisk got this function, but are it stable for long time recording purpose. I want all recording been recorded. |
| 03:21.28 | rob0 | maskEd: take it back a step, what did you do to start out? |
| 03:21.43 | *** join/#asterisk seeeexy_girl_06 (n=seeeexy_@c-67-181-117-151.hsd1.ca.comcast.net) |
| 03:21.47 | Qwell | maskEd: What are you typing? |
| 03:21.48 | Qwell | See? |
| 03:21.51 | maskEd | rob0: built zaptel, then make install |
| 03:21.53 | YoMama | i've met one good looking female unix geek in my entire life...that was about...10 years ago |
| 03:21.58 | Ariel_ | OK back... she might do better with vmware and then loading asterisk.. But I still would not do it. |
| 03:21.59 | Math` | mgoh: it is |
| 03:22.03 | maskEd | Qwell: make install to install zaptel |
| 03:22.10 | YoMama | and she actually knew what she was doing...ha |
| 03:22.13 | Qwell | maskEd: you should pastebin the whole thing |
| 03:22.22 | maskEd | Qwell: at pastebin now |
| 03:22.35 | rob0 | looks don't matter much anyway :) |
| 03:22.41 | maskEd | Qwell: the whole lot or just the error? |
| 03:22.45 | Qwell | the whole thing |
| 03:22.49 | seeeexy_girl_06 | hey shido reach me on yahoo messager if youd like to help me |
| 03:23.00 | zimdog | Does anyone here connect to a broadsoft switch? |
| 03:23.04 | *** join/#asterisk FastJack_ (i=fastjack@p5091E315.dip.t-dialin.net) |
| 03:23.14 | Qwell | shido: She's trying to abuse CID with nufone! :D |
| 03:23.17 | Ariel_ | broadsoft switch.... |
| 03:23.22 | nrl[digium] | exit |
| 03:23.27 | mgoh | Math': Do you think asterisk able handle 60 extension voice recording conccurently. |
| 03:23.33 | YoMama | Qwell: u can set your callerid with nufone? |
| 03:23.37 | Qwell | YoMama: num |
| 03:23.40 | seeeexy_girl_06 | qwell i am not... |
| 03:23.50 | seeeexy_girl_06 | i am trying to do the same shit they did to me... |
| 03:23.56 | YoMama | Qwell: really...that can't be too good for business |
| 03:24.18 | seeeexy_girl_06 | other than that ill be using it to spoof my own cell number in order to use my cell number when im in the house... which doesnt work inside currentkly |
| 03:24.18 | YoMama | seeeexy_girl_06: you're not quite over this guy are you? he dumped you didn't he? |
| 03:24.26 | Libila | lol |
| 03:24.28 | Ariel_ | seeeexy_girl_06, you could load vmware then put a real asterisk setup there |
| 03:24.44 | seeeexy_girl_06 | he cheated on me with some ho! |
| 03:24.54 | seeeexy_girl_06 | anyway |
| 03:24.54 | Libila | Ariel_: Whats your definition of a "real" asterisk system? |
| 03:24.55 | seeeexy_girl_06 | im out |
| 03:24.59 | Ariel_ | seeeexy_girl_06, you could do it simple like use a 9 for a different id 8 for normal... |
| 03:25.12 | YoMama | hmm |
| 03:25.15 | *** join/#asterisk oppie (n=oppie@12-217-44-162.client.mchsi.com) |
| 03:25.33 | YoMama | actually..i'm startin' to think she's probably pretty hot...she ain't too bright and she's a bit nuts..and u know what they say about the crazy ones..they're usually hot |
| 03:25.42 | zimdog | Arial Broadsoft server? |
| 03:25.49 | seeeexy_girl_06 | aint too bright? |
| 03:25.59 | seeeexy_girl_06 | excuse me im at a state college unlike you |
| 03:26.00 | Ariel_ | zimdog, yes I know it. it's a very strange system to use. |
| 03:26.09 | oppie | Wait, I thought we were suppose to talk about asterisk in here. |
| 03:26.15 | Qwell | oppie: usually |
| 03:26.15 | YoMama | seeeexy_girl_06: are u going to tell me now that "ain't" ain't a word? |
| 03:26.26 | oppie | OK, I have a question on compiling. |
| 03:26.34 | YoMama | seeeexy_girl_06: it took u all the way to state college to figure that one out? |
| 03:26.57 | oppie | I did make rpm and at the very end, it gives me this error. |
| 03:26.58 | zimdog | Arial_: Have any pointers to get outbound trunk working? |
| 03:27.03 | oppie | make[1]: Leaving directory `/home/oppie/asterisk-1.2.2' |
| 03:27.03 | oppie | error: Legacy syntax is unsupported: copyright |
| 03:27.03 | oppie | error: line 6: Unknown tag: Copyright: Linux Support Services, inc. |
| 03:27.03 | oppie | make: *** [__rpm] Error 1 |
| 03:27.05 | seeeexy_girl_06 | im out |
| 03:27.10 | YoMama | 1.2.2??? |
| 03:27.13 | seeeexy_girl_06 | reach me on yahoo if you want to help |
| 03:27.15 | oppie | YEs |
| 03:27.18 | seeeexy_girl_06 | same name as this |
| 03:27.19 | Math` | oppie: what the hell |
| 03:27.24 | maskEd | http://pastebin.com/514101 |
| 03:27.32 | YoMama | Ariel_: Michigan? |
| 03:27.45 | Ariel_ | Miami the canes rule |
| 03:27.46 | YoMama | Ariel_: or Minnesota? |
| 03:28.00 | YoMama | Ariel_: AUGH..you go to a Florida school..curse you..I'm a Wolverine |
| 03:28.03 | maskEd | Qwell: http://pastebin.com/514101 |
| 03:28.15 | Math` | echo "alias wcfxs wctdm" >> ; |
| 03:28.19 | Math` | thats not really good |
| 03:28.25 | YoMama | Ariel_: although..the girls are about 1000x hotter at U of Miami |
| 03:28.36 | Ariel_ | YoMama, I got out of school back in 1979 so I am sure it was before you started. |
| 03:28.39 | YoMama | so hats off to you there |
| 03:28.49 | maskEd | Math`: its not? |
| 03:28.50 | YoMama | Ariel_: yeah...a bit |
| 03:29.00 | YoMama | Ariel_: sure the chix were still pretty hot back then too :-P |
| 03:29.01 | reza | ok |
| 03:29.05 | *** join/#asterisk linlin (i=linlin@c-67-184-231-154.hsd1.il.comcast.net) |
| 03:29.12 | Ariel_ | ohh yes they still are |
| 03:29.20 | YoMama | Ariel_: and the Michigan rivalry with Florida teams goes well before 1979 |
| 03:29.28 | Ariel_ | yes I know |
| 03:29.30 | Math` | maskEd: not really, it expects a filename for IO redirection... like echo "alias wcfxs wctdm" >> /etc/modules.conf |
| 03:29.48 | YoMama | Ariel_: I hate Florida State even more |
| 03:29.53 | maskEd | Math`: now why wouldn't it have done that? |
| 03:29.55 | Ariel_ | but the main school that was our rivalry back then was Ohio state. |
| 03:29.56 | rob0 | line 151: if [ -f ] ... no argument for -f |
| 03:29.57 | oppiet30 | So, should I go and look through the bugs? |
| 03:30.07 | YoMama | Ariel_: well, we'll hate them together... |
| 03:30.14 | YoMama | OHOWIHATEOHIOSTATE |
| 03:30.15 | Math` | maskEd: dunno it works pretty fine on my box |
| 03:30.27 | maskEd | Math`: would that have something to do with me not having a modules.conf? |
| 03:30.36 | *** join/#asterisk Trazz (i=Trazz@c-67-163-92-37.hsd1.il.comcast.net) |
| 03:30.39 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 03:30.43 | Math` | you don't have a /etc/modules.conf? |
| 03:30.55 | maskEd | rob0: pardon? |
| 03:30.58 | maskEd | nope |
| 03:31.03 | *** part/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 03:31.06 | reza | what does this mean : ZT_CHANCONFIG failed on channel 1: No such device or address (6) |
| 03:31.06 | Ariel_ | I have to go and see if I can get my baby back to sleep |
| 03:31.10 | Math` | maskEd: lets try.... touch /etc/modules.conf |
| 03:31.11 | reza | when i modprobe'd wctdm? |
| 03:31.14 | maskEd | Math`: i have a /etc/sysconfig/modules |
| 03:31.29 | YoMama | reza: it means it woucldn't find the tdm device |
| 03:31.35 | YoMama | reza: errr..couldn't |
| 03:31.53 | YoMama | reza: probably goofed your zapata.conf file |
| 03:32.06 | reza | i did a second modprobe and it didn't err the second time |
| 03:32.19 | rob0 | maskEd: I don't know :( |
| 03:32.22 | reza | and now they are in modprobe |
| 03:32.23 | reza | er |
| 03:32.24 | reza | lsmod |
| 03:32.34 | Qwell | reza: It'll only give that error the first time |
| 03:32.40 | Qwell | it still loads the module, is why |
| 03:32.49 | Qwell | so you have to unload it, then load it again to fix it |
| 03:32.55 | Qwell | but...you need to figure out why it's erroring |
| 03:32.57 | reza | [root@dhcp90 asterisk]# cat /etc/zaptel.conf |
| 03:32.58 | reza | fxoks=1-2 |
| 03:32.58 | reza | defaultzone=us |
| 03:32.58 | reza | loadzone=us |
| 03:32.58 | rob0 | oh, look at 161, ">> ;" |
| 03:32.58 | Qwell | ztcfg -vvvv |
| 03:33.24 | reza | it's got two of the green modules in it |
| 03:33.33 | maskEd | Math`: well it didn't help install.. |
| 03:33.40 | rob0 | actually not just there, it's at every > or >> operator |
| 03:34.08 | Trazz | qwell, can you have * transfer to an extension and if your not there have it call your cell, your house, etc ? |
| 03:34.22 | maskEd | still no file there.. |
| 03:35.01 | maskEd | well how can i change that....? |
| 03:35.04 | *** join/#asterisk wilymage (n=wily@funkmunch.net) |
| 03:35.11 | Qwell | Trazz: sure |
| 03:35.21 | Trazz | qwell, how do i do that type of stuff? |
| 03:35.35 | Qwell | Trazz: simple failover...search the wiki for failover |
| 03:35.41 | Trazz | ok |
| 03:36.10 | YoMama | Coccyx: u there? |
| 03:37.00 | reza | should the TDM cards show up in lspci? |
| 03:37.05 | Qwell | reza: yes |
| 03:37.09 | Qwell | tigerjet or something |
| 03:37.13 | reza | it's not |
| 03:37.17 | reza | *grumble* |
| 03:38.27 | reza | i wonder if i got a bad card |
| 03:38.31 | reza | do these cards die? |
| 03:38.37 | Qwell | reza: call digium |
| 03:38.43 | rob0 | reza: power connector? |
| 03:38.45 | maskEd | any idea why make install isn't supplying a place to output those entries to? |
| 03:38.46 | *** part/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 03:38.49 | reza | it's plugged in |
| 03:38.50 | Qwell | rob0: it should still show up |
| 03:38.53 | oppiet30 | I googled the error, adn that didn't tell me anything. http://lists.digium.com/pipermail/asterisk-users/2005-October/128558.html |
| 03:38.54 | rob0 | reza: PCI 2.1? |
| 03:39.06 | reza | rob - ? |
| 03:39.19 | *** join/#asterisk r1ddl3r (n=blah@24-171-11-166.dhcp.stls.mo.charter.com) |
| 03:39.19 | reza | it's an old old system |
| 03:39.25 | reza | 400mhz celeron |
| 03:39.37 | rob0 | aha |
| 03:39.44 | javier | gotta go thanks FuriousGeorge...... |
| 03:40.07 | rob0 | <== couldn't get my TDM400P to work in a PII-400 box |
| 03:40.14 | oppiet30 | I am running a 500 Mhz Celeron on this linux box. |
| 03:40.31 | FuriousGeorge | np |
| 03:42.06 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 03:42.09 | maskEd | what package is update-modules contained in? |
| 03:42.12 | reza | rob - think it's the mb then? |
| 03:42.37 | rob0 | reza: that is my guess, yes. Do you have a newer system you could try? |
| 03:42.40 | r1ddl3r | if I am writing a script in AGI and use the GET DATA function, but when I enter any dtmf digits, it doesnt say 200 result = 1234 or whatever? |
| 03:42.46 | mgoh | how do u ppl over come if IP-PBx fail suddently. Whole company will stop working with it. |
| 03:45.36 | reza | robo - yeah, i'll test it out in a bit; god that would suck. |
| 03:46.54 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 03:47.39 | *** join/#asterisk jjhall (n=chatzill@94-253.69-92-cpe.cableone.net) |
| 03:47.54 | *** join/#asterisk SwK (n=SwK@12-219-147-107.client.mchsi.com) |
| 03:50.20 | rob0 | I also tried in a Via C3 800 box, whose manual claimed PCI 2.1 compliance. They lied. |
| 03:50.41 | rob0 | I ended up buying another motherboard for the job. |
| 03:50.51 | *** join/#asterisk bmg505 (n=leon@c1-40-1.rndf.isadsl.co.za) |
| 03:51.12 | inv_Arp | thats why I use external fxo |
| 03:51.16 | PoWeRKiLL | since asterisk > 1.2.0 I got Invalid or unknown command when getting value from DIALSTATUS any idea ? |
| 03:52.53 | jjhall | Is it possible to register and unregister a SIP connection from the dialplan? For example, #11 to register as a user to an account, #12 to unregister and no longer take calls. |
| 03:53.20 | *** join/#asterisk Pegger (n=peg@pool-68-163-139-134.bos.east.verizon.net) |
| 03:54.57 | *** join/#asterisk ke4qqq (n=chatzill@srv.fgp.com) |
| 03:56.46 | inv_Arp | jjhall: prob... a key combination to sign off |
| 03:57.05 | De_Mon | If I change extensions within the same context, do I have to re-set my timeouts? |
| 03:57.48 | De_Mon | when I change xtensions within the same context, my timeouts appear to reset... |
| 03:58.02 | jjhall | inv_Arp: Basically what I want to do is login to a call queue and log back out, only the queue doesn't operate with agents, it is just a SIP account where the first person to answer gets the call. |
| 03:58.35 | jjhall | I'm just using a softphone logged in right now and exit it when I am done. I'd rather have Asterisk handle it so I can use my hardware phones. |
| 03:58.37 | Qwell | jjhall: use dynamic members |
| 03:58.47 | Qwell | then AddQueueMember and RemoveQueueMember |
| 03:58.59 | FuriousGeorge | im reading about this snom 360 on voip-info and it makes no mention of how well if at all the leds and the intercom work w/ * |
| 03:59.49 | jjhall | If I'm reading that correctly, I need control over the server end of the call. I do not have that ability, I am a use only on the originating system. |
| 04:00.17 | SkramX | does ,r in musiconhold work for anyon? |
| 04:00.23 | inv_Arp | Qwell: damn you have gotten good at this |
| 04:00.28 | SkramX | it doesnt actually happen when i use it |
| 04:00.30 | Qwell | gotten? |
| 04:00.38 | inv_Arp | ha |
| 04:01.49 | inv_Arp | I remember starting before you ... |
| 04:02.18 | SkramX | default => quietmp3:/var/lib/asterisk/mohmp3/Funk/,r |
| 04:02.21 | SkramX | i have that.. |
| 04:02.27 | inv_Arp | but alas... I got drowned at work doing *nix administraion/php |
| 04:02.30 | jjhall | Qwell: So do you think I am out of luck? |
| 04:04.22 | De_Mon | Qwell you were born that good? |
| 04:04.57 | De_Mon | inv_Arp boooriiiing |
| 04:05.21 | inv_Arp | ha |
| 04:05.27 | oppiet30 | I think I fixed the make rpm problem. It has to do with the asterisk.spec file. Change Copyright to License on line 6 |
| 04:07.37 | mgoh | Are Asterisk able to do load balancing? |
| 04:08.00 | oppiet30 | Change Are to Is. Grammer cops on patrol |
| 04:08.09 | oppiet30 | :) |
| 04:13.42 | *** part/#asterisk xtrvd (n=j@d209-121-36-44.bchsia.telus.net) |
| 04:15.53 | *** join/#asterisk MnxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 04:16.07 | FuriousGeorge | omg, i can use devstate to make led's on phones correspond to parking spots and conferences! |
| 04:16.29 | Math` | wow digium charges 175$usd per hour of support |
| 04:16.31 | zu | really |
| 04:16.41 | zu | thats cool you just made a key system |
| 04:16.53 | Math` | http://store.digium.com/products.php?category_id=11 |
| 04:16.58 | Qwell | Math`: it's worth it |
| 04:17.07 | Math` | its a lot |
| 04:17.23 | Math` | I wonder how much they pay their techs |
| 04:17.30 | rob0 | I don't think US$175 is a lot. |
| 04:17.38 | justinu | whatever a 6 pack of redbull costs/hr |
| 04:17.49 | rob0 | for specialized support ... it's cheap. |
| 04:18.50 | MnxPower | That's only a little more than I charge for consulting |
| 04:19.03 | Math` | how much do you charge for consulting |
| 04:19.54 | MnxPower | My official rate is $120/hr, but various discounts can apply depending on the customer. |
| 04:20.07 | Math` | and for how long have you been in business? |
| 04:20.14 | *** join/#asterisk javier (n=javier@adsl-64-219-154-129.dsl.hrlntx.swbell.net) |
| 04:20.38 | justinu | i'm doing an asterisk/legacy pbx integration... trying to set up an E&M tie trunk... PBX doesn't support PRI |
| 04:20.39 | zu | mine is 200-300 for security and 100-200 for unix/linux/aix/rpg/as400/vaxvms |
| 04:20.43 | MnxPower | You can get %20 off if 1) the invoice is more than $1,000 AND you pay with 15 days of the date of the invoice (which is sent via e-mail) |
| 04:20.53 | justinu | couldn't figure out why the PBX wouldn't respond to anything I sent it |
| 04:20.58 | MnxPower | Oh, I also have to like you. |
| 04:21.13 | justinu | found out today the friggen PBX doesn't have DTMF receivers!!! |
| 04:21.23 | *** join/#asterisk Peggerr (n=peg@pool-68-163-232-33.bos.east.verizon.net) |
| 04:21.35 | MnxPower | justinu, ROFL! you need to collect digits via PULSE? |
| 04:21.41 | justinu | MnxPower: yes |
| 04:21.46 | MnxPower | justinu, that's pretty common on PBXs. |
| 04:21.57 | justinu | and the PBX can't reliably receive what ast is sending |
| 04:22.03 | *** join/#asterisk wizard545 (n=wizard@cpe-65-25-136-96.columbus.res.rr.com) |
| 04:22.08 | *** join/#asterisk Chotaire (i=chotaire@chotaire.net) |
| 04:22.12 | justinu | it works 1 out of maybe 5 tries |
| 04:22.18 | MnxPower | The analog adapters on the Nortel don't have DTMF generators either. They also don't provide disconnect supervision |
| 04:22.21 | justinu | thinking of recompiling zaptel to reset the defaults |
| 04:22.29 | MnxPower | justinu, fiddle with the pulse lengths. |
| 04:22.31 | justinu | for pulse break timing |
| 04:22.45 | justinu | MnxPower: it looks like the default in zaptel is 50ms |
| 04:23.09 | justinu | i figured that was actually on the long side |
| 04:23.14 | *** join/#asterisk FranckM (n=franck@202.62.0.1) |
| 04:23.24 | [av]bani | awesome... got auto-generation of sip peers working |
| 04:23.24 | justinu | but maybe it should be 100 which is 10 pps |
| 04:23.34 | *** join/#asterisk alphaque (n=alphaque@218.111.24.41) |
| 04:23.36 | [av]bani | now just have to make it auto-reload in * :) |
| 04:23.50 | MnxPower | The "woman" that is the official phone person (she didn't get hired on merit) was yelling about a problem with the Asterisk/Nortel link having issues "again". This is the first report I've heard of this specific problem since the end of Aug. |
| 04:24.18 | MnxPower | justinu, there are telco stanaards for this. |
| 04:24.39 | *** join/#asterisk Peggerr (n=peg@pool-68-163-232-33.bos.east.verizon.net) |
| 04:24.48 | MnxPower | 50ms? 500ms is .5 second, so 50ms would be .05 second |
| 04:25.04 | justinu | yeah... so that would allow it to send 20 pulses per second |
| 04:25.23 | justinu | it could outpulse 10 in 500ms |
| 04:25.45 | Math` | heh |
| 04:25.54 | justinu | the pbx has options to send 10 or 20pps |
| 04:25.56 | MnxPower | justinu, do you know why area codes were originally assigned where they were? |
| 04:26.04 | justinu | yes |
| 04:26.07 | justinu | i told you guys :P |
| 04:26.37 | Qwell | I told my wife that yesterday...she didn't believe me. heh |
| 04:26.59 | MnxPower | so on a rotary dial phone places with larger populations (NYC = 212, LA = 213) would have a shorter "pull" |
| 04:27.08 | MnxPower | ugh, that made no sense. |
| 04:27.11 | justinu | yep |
| 04:27.21 | Qwell | My old house wins |
| 04:27.24 | justinu | lol |
| 04:27.24 | Qwell | 909 |
| 04:27.27 | Qwell | booyah |
| 04:27.40 | justinu | MnxPower: glad you've got your priorities straight |
| 04:27.43 | Qwell | longest to dial, besides 900 :p |
| 04:27.53 | FranckM | What are the main diff between 1.2.1 and 1.2.2? |
| 04:27.59 | Qwell | FranckM: bug fixes |
| 04:28.19 | FranckM | Qwell: no new features beside netsec? |
| 04:28.36 | justinu | nutsac? |
| 04:29.16 | Mavvie | hope it fixes my mysterious "calls come in, but don't go out" problem. |
| 04:29.25 | Mavvie | happened twice in a row, then never came back again. |
| 04:31.36 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 04:32.27 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 04:34.02 | MnxPower | justinu, The ultimate goal is to get OTHER stuff transfered over. |
| 04:34.31 | Mavvie | * = ※ |
| 04:34.49 | Mavvie | (See http://everything2.com/?node_id=1421689 for more information) |
| 04:35.52 | oppiet30 | FranckM: http://ftp.digium.com/pub/telephony/asterisk/ChangeLog-1.2.2 |
| 04:35.55 | *** join/#asterisk lrizzo (n=rizzo@151.52.4.6) |
| 04:36.52 | justinu | MnxPower: yeah... i'm just providing technical backup on this, someone else is calling those shots |
| 04:38.07 | FranckM | oppiet30: yeah... too much to read ;) |
| 04:39.43 | *** part/#asterisk lrizzo (n=rizzo@151.52.4.6) |
| 04:40.00 | *** join/#asterisk tainted- (n=identd@ppp-71-134-157-119.dsl.irvnca.pacbell.net) |
| 04:40.24 | *** join/#asterisk copantl (n=copantl@205.240.200.96) |
| 04:41.16 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 04:43.01 | r1ddl3r | what the heck do I have to do to get my AGI script to show the dtmf, on GET DATA? it used to say 200 result = whatever I pressed, but now it doesnt |
| 04:43.46 | *** join/#asterisk greendisease (n=greendis@fedora/greendisease) |
| 04:44.43 | FranckM | it seems I get every 10mn a chan_sip.c:5558 check_auth: stale nonce received |
| 04:44.49 | FranckM | how comes? |
| 04:48.02 | *** join/#asterisk bch (n=bch@CPE-70-92-133-175.mn.res.rr.com) |
| 04:48.40 | bch | how does one return values to asterisk in PHP? |
| 04:49.29 | *** join/#asterisk ManxPower (i=ewieling@137.sub-70-210-120.myvzw.com) |
| 04:49.54 | ManxPower | Apparently I have a screw loose |
| 04:50.22 | justinu | or two |
| 04:50.54 | ManxPower | I think the power cord into the router in the main house is not seated good. |
| 04:51.00 | ManxPower | I'll check on it tomorrow |
| 04:54.24 | tainted- | Jan 19 20:34:41 DEBUG[1801]: Didn't get a frame from channel: |
| 04:54.29 | tainted- | anyone know what that is? |
| 04:54.37 | tainted- | audio cuts out and i get that message |
| 04:54.44 | justinu | VAD? |
| 04:54.47 | justinu | CNG? |
| 05:01.54 | tainted- | is there anything i should delete when upgrading from 1.0.7 to 1.2.2? |
| 05:02.06 | tainted- | or will it overwrite the corresponding files for me |
| 05:02.47 | *** join/#asterisk shekhar (n=shekhar@221-128-138-173.exatt.net) |
| 05:03.12 | fugitivo | is any way to change the name of the agent monitor file? |
| 05:03.14 | ManxPower | tainted-, "make install" will tell you what modules should be removed. |
| 05:03.29 | tainted- | ManxPower thx! |
| 05:03.41 | shekhar | hi |
| 05:03.59 | *** join/#asterisk Qwell_ (n=north@24-205-180-81.dhcp.wsco.ca.charter.com) |
| 05:04.05 | fugitivo | no? |
| 05:05.03 | tainted- | ManxPower Loading chan_modem.so failed! |
| 05:05.10 | tainted- | can i just remove that module if i don't use it? |
| 05:05.21 | ManxPower | tainted-, usually yes. |
| 05:05.30 | ManxPower | certinally for the chan_modem* |
| 05:05.48 | *** join/#asterisk annonimous (i=annonimo@dsl-201-129-251-131.prod-infinitum.com.mx) |
| 05:05.52 | annonimous | hello |
| 05:05.59 | tainted- | <PROTECTED> |
| 05:06.03 | tainted- | that's a new file... |
| 05:06.11 | annonimous | anybody here knows if the Audiocodes fxs can run with asterisk? =/ |
| 05:06.34 | fugitivo | annonimous: the gateways? yes |
| 05:06.48 | annonimous | fugitivo yes the gateways audiocode |
| 05:06.52 | fugitivo | yes |
| 05:06.55 | fugitivo | they work |
| 05:06.55 | annonimous | fugitivo talk spanish? |
| 05:07.00 | fugitivo | yes |
| 05:07.02 | ManxPower | tainted-, it's optional |
| 05:07.04 | annonimous | jeje |
| 05:07.08 | annonimous | me too xD |
| 05:07.10 | ManxPower | tainted-, read UPGRADE.txt |
| 05:07.23 | fugitivo | how can i change the filename of an agent monitor file? |
| 05:07.23 | annonimous | fugitivo can u tell me when i can found a howto or mini-manual? |
| 05:07.48 | wasim | fugitivo: its a var, irrc |
| 05:07.56 | fugitivo | is it UNIQUEID? |
| 05:08.12 | tainted- | ManxPower Jan 19 21:05:22 WARNING[18224]: loader.c:554 load_modules: Loading module pbx_wilcalu.so failed! |
| 05:08.30 | ManxPower | tainted-, that should have been listed when you did a "make install" |
| 05:08.50 | tainted- | how do i remove it? |
| 05:08.54 | tainted- | noload? |
| 05:08.58 | tainted- | in modules.conf? |
| 05:09.23 | ManxPower | rm it |
| 05:09.27 | *** join/#asterisk dsfr (n=dsfr@pdpc/sponsor/digium/dsfr) |
| 05:09.40 | ManxPower | in fact, unless you are using custom modules or g729 codec just rm /var/lib/asterisk/modules |
| 05:09.46 | ManxPower | then re run make install |
| 05:10.04 | tainted- | i'm using g729 |
| 05:10.06 | wasim | fugitivo: /usr/src/cvs-src/asterisk/doc/README.variables |
| 05:10.21 | wasim | ${MONITOR_FILENAME} File for monitoring (recording) calls in queue |
| 05:10.22 | *** join/#asterisk tengulre11 (n=tengulre@221.11.5.180) |
| 05:10.26 | tainted- | so i rm ../modules and re-copy the 729.so back in? |
| 05:10.39 | wasim | tainted-: yeah, you have to pay $10 for every cp |
| 05:10.57 | fugitivo | wasim: hmm, no, that doesn't change agents recorded files |
| 05:10.57 | wasim | tainted-: so use mv, instead |
| 05:11.09 | wasim | fugitivo: ok, have it your way |
| 05:11.13 | tengulre11 | best wish to Mission to Pluto USA!! |
| 05:11.31 | fugitivo | wasim: that change queue's files, but not agents |
| 05:11.58 | tainted- | wasim thx |
| 05:12.54 | *** join/#asterisk kshumard (n=kshumard@gateway.digium.com) |
| 05:12.56 | *** join/#asterisk angler (n=angler@gateway.digium.com) |
| 05:12.58 | wasim | tainted-: that was a failed attempt at 1000 hours humour |
| 05:13.05 | *** join/#asterisk reni (n=nubb@gateway.digium.com) |
| 05:13.07 | tainted- | yea i know |
| 05:13.12 | wasim | i need more coffee |
| 05:13.15 | tainted- | :D |
| 05:13.42 | tainted- | i've always wondered why the licenses are so restrictive |
| 05:13.50 | tainted- | to prevent transfers? |
| 05:16.45 | fugitivo | i don't understand 100% monitor for queues |
| 05:16.53 | fugitivo | i have the option to record queues AND agents |
| 05:17.06 | fugitivo | but if i turn on monitoring for agents, queue is not monitored |
| 05:17.19 | FuriousGeorge | im reading about these snom phones. turns out they have an intercom feature and a shared line feature the latter allows multiple phones to pickup calls put on hold, and obviously they all ring together. will this work with asterisk? |
| 05:18.15 | FuriousGeorge | i mean i know i can make multiple phones ring at the same time, i wanna know if the "shared line" program-ability will work |
| 05:19.17 | DarkFlibble | FuriousGeorge, why not use call parking... you can mix your phone types then |
| 05:20.56 | FuriousGeorge | DarkFlibble: i just dont feel like explaining to people the difference b/t hold and park |
| 05:21.02 | tainted- | were call files changed between 1.0.7 and 1.2.2? |
| 05:21.16 | tainted- | i have setvars in my call file that no longer work after the upgrade |
| 05:21.49 | DarkFlibble | FuriousGeorge, kk |
| 05:21.50 | FuriousGeorge | ive always said there should be a reverse transfer feature with asterisk where you can "pull" a call from someone just by knowing their extension, but no one seems to agree with me |
| 05:24.39 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 05:25.03 | jjhall | FuriousGeorge: You can do that via Manager commands. Regular users shouldn't have the ability to do that for obvious security reasons. |
| 05:25.46 | *** join/#asterisk Storm (n=StorM@sgc91-2-82-237-76-2.fbx.proxad.net) |
| 05:26.55 | Coccyx | anyone done anything with GROUP_COUNT()? |
| 05:27.04 | Coccyx | for the life of me I can't get it to properly increment |
| 05:27.13 | Storm | hello, i have a trouble with "File size limit exceeded" when uptime of my * go to several day. ulimit said unlimited to fd, any idea what else to check ? thanks |
| 05:27.14 | fugitivo | is any way, to know which agent answered a queue? |
| 05:28.38 | FuriousGeorge | jjhall: how so? i could listen for when someone is put on hold then based on that allow another user to dial (for instance) their extension + star to transfer the call to them |
| 05:29.22 | FuriousGeorge | ~agi |
| 05:29.28 | jbot | it has been said that agi is the Asterisk Gateway Interface... similar to CGI for web applications AGI lets you script call control and access databases using your favorite language. AGI wrappers are available for Python (pyst), Perl (astperl?) and other languages |
| 05:29.42 | file | you gotta keep on holding on, it's about as bad as it could be |
| 05:29.45 | file | seems everyone's bugging me |
| 05:29.57 | file | like nothing wants to go my way, it just ain't been my day, nothing's coming easily |
| 05:30.55 | fugitivo | it seems i can't find a solution to this, and i don't know if i'm missing something |
| 05:31.16 | fugitivo | i want to monitor agent's calls, but with the possibility of changing the filename of the recorded call |
| 05:31.22 | FuriousGeorge | jjhall: is that what you meant? |
| 05:32.22 | wasim | fugitivo: are you using Monitor() for it? |
| 05:32.22 | fugitivo | it seems that the variable MONITOR_FILENAME only works for queues |
| 05:32.22 | fugitivo | wasim: no |
| 05:32.32 | wasim | how ya doing it? |
| 05:32.37 | fugitivo | wasim: monitor from agents.conf and queues.conf |
| 05:32.58 | fugitivo | but if i monitor queues, it doesn't record agents calls |
| 05:33.08 | fugitivo | and i want to know which agent answered the call |
| 05:33.15 | fugitivo | so i don't have that info recording queue's calls |
| 05:33.23 | shekhar | hi - apologies to interrupt - i typed a query into extensions.conf which involved retrieving a numeric value from ASTdb followed by SayNumber to speak out the number. The family/key pair does not exist. So technically it should go to priority n+101 (102 in my case), but it doesn't seem to be doing so. Instead, the CLI keeps telling me "SayNumber requires an argument (number)". Can anyone please help? |
| 05:33.25 | fugitivo | am I missing something? |
| 05:34.18 | fugitivo | wasim: recordagentcalls=yes for agents.conf and monitor_format=gsm for queues.conf |
| 05:34.44 | [TK]D-Fender | shekhar : the old priority jumping is depricated. There is a value to test to see if the lookup succeeded. There are many ways to prevent taht from disrupting your dialplan however |
| 05:35.01 | file | [TK]D-Fender: eep you're alive, run away! |
| 05:35.30 | justinu | deprecated :P |
| 05:35.41 | shekhar | any leads? |
| 05:36.09 | FuriousGeorge | can i use the manager interface to tell me when an incomming call is placed on hold? |
| 05:36.30 | shekhar | thanks |
| 05:39.29 | *** join/#asterisk drumkill1 (n=russell@host-12-179-65-65.nctv.com) |
| 05:41.07 | wasim | what a long hostname you have, my dear! |
| 05:41.32 | Corydon76-home | All the better to confuse you with, my dear. |
| 05:41.39 | justinu | lol |
| 05:41.39 | Qwell | You know what they say about guys with long hostnames, don't you? |
| 05:41.40 | DarkFlibble | lol |
| 05:41.59 | ManxPower | Qwell, someone is compensating for something? |
| 05:42.22 | Qwell | ManxPower: something like that |
| 05:42.55 | Corydon76-home | Heh, I've been known to shorten some of my subdomains by using airport codes... |
| 05:43.07 | Corydon76-home | Confusing as hell to try to remember |
| 05:43.26 | Coccyx | ok, this is really weird, I can't figure it out... following the examples and I just can't seem to get GROUP_COUNT working right |
| 05:43.33 | Coccyx | GROUP_LIST returns the groups its in |
| 05:43.54 | Coccyx | but I can't get it to return anything other than 1 if I do a GROUP_COUNT against the same exact string in the GROUP_LIST |
| 05:43.57 | Coccyx | strange. |
| 05:44.39 | Corydon76-home | Coccyx: you've got channels in multiple groups? |
| 05:45.14 | Coccyx | Corydon76-home: how do you mean multiple groups? pickup groups? call groups? they're all int he same group now |
| 05:45.24 | Coccyx | but I'm assigning a group when I dial out to the SIP extension |
| 05:45.26 | Corydon76-home | Neither of those |
| 05:45.36 | Corydon76-home | What group are you setting? |
| 05:45.40 | Coccyx | which is where I'm trying to maximize one inbound call at a time from the PSTN |
| 05:45.52 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 05:46.45 | Coccyx | http://pastebin.com/514204 |
| 05:46.51 | Coccyx | that's my macro so far |
| 05:47.23 | FuriousGeorge | i guess i see how i can put an incoming call in a meetme room, what i dont get is how i can then get that room to call extensions. is this possible? |
| 05:47.31 | ManxPower | Coccyx, you can do something like setvar=GROUP=1 in the Zap channel config |
| 05:47.48 | shekhar | can someone tell me the new method of checking if an ASTdb lookup succeeded or failed? |
| 05:47.49 | ManxPower | that will automagically set it anytime a call comes from that device. |
| 05:48.05 | Coccyx | ManxPower: yeah, I don't care about what groups the zap channels are in, it's the SIP calls I'm concerned about |
| 05:48.05 | ManxPower | I'm not SURE that feature is in chan_zap, but it's in chan_sip and chan_iax2 |
| 05:48.19 | Coccyx | I want to make sure any calls from the PSTN to a SIP extension don't exceed more than one call per extension |
| 05:48.30 | wasim | come on aussie come on |
| 05:48.40 | *** join/#asterisk imran (i=imran@cpe-68-206-53-16.houston.res.rr.com) |
| 05:48.45 | Coccyx | it's the leg from asterisk to the SIP extension I want to set a group on |
| 05:48.47 | Corydon76-home | Is ARG1 perhaps empty? |
| 05:48.55 | Coccyx | no, my noop output is good |
| 05:49.11 | Corydon76-home | Pastebin your output |
| 05:49.30 | Coccyx | ok, it's up at: http://pastebin.com/514208 |
| 05:49.59 | shekhar | will someone please tell me the new method of checking if an ASTdb lookup succeeded or failed? |
| 05:50.02 | Coccyx | see if you agree that it looks likeit should work |
| 05:50.25 | Corydon76-home | So what's wrong with it? |
| 05:50.53 | Coccyx | it doesn't work, it always returns 1 no matter if there's already a call to that SIP Channel |
| 05:51.01 | wasim | the problem is that now that the rest of the world has finally started making inroads against their blasted top and middle order batsmen, their tail now wags, brett lee scores 50s ... bah |
| 05:51.32 | swm_ | Anyone know of a home automation solution that mounts on a wall has a small touch screen lcd input and output audio a few usb ports and a network adapter wireless or cabled |
| 05:52.41 | Corydon76-home | Coccyx: Oh, I think I know why... |
| 05:52.54 | Corydon76-home | Why are you using the Local channel? |
| 05:53.17 | Coccyx | Corydon76-home: Because I need to ring multiple extensions that execute the Macros |
| 05:53.19 | Corydon76-home | The variables aren't persisting |
| 05:53.25 | Coccyx | ah, ic |
| 05:53.39 | Corydon76-home | You're going to need to find a different way to do that |
| 05:53.58 | Coccyx | Corydon76-home: hrm, ok, any ideas on how to Execute multiple macros simultaneously another way? |
| 05:54.18 | Coccyx | so I can run a macro for extension 100, 101, 102 etc simultaneously? |
| 05:54.22 | Coccyx | that's why I'm using LOCAL/ now |
| 05:54.45 | Corydon76-home | I'm not seeing the purpose of running all those extensions at once |
| 05:54.59 | Coccyx | on an incoming call, I need to ring about 4 extensions at once |
| 05:55.00 | FuriousGeorge | does anyone see what i mean? let's say a call comes in and i answer it, then dial a meetme extension. thatll put the calling party into a conference room. now how do i ring an extension or two and if the extension answers, put them in that conference with the caller |
| 05:55.02 | Coccyx | it's setup a like a key system |
| 05:55.12 | Coccyx | ring multiple phones at once... no autoattendant |
| 05:55.17 | Corydon76-home | I thought you're dialling out on the PSTN... |
| 05:55.24 | Coccyx | no, this is for inbound calls |
| 05:55.36 | wasim | fugitivo: use a call file |
| 05:55.41 | Corydon76-home | You want to limit inbound calls? |
| 05:55.56 | Coccyx | yes, if you're already on the phone, you shouldn't get a second call |
| 05:55.59 | znoG | linksys have *the* worst support monkeys ever |
| 05:56.07 | znoG | they don't even know the PAP2-NA exists! |
| 05:56.09 | Corydon76-home | So turn off call waiting |
| 05:56.17 | znoG | "sir, you have to talk to your VoIP provider" |
| 05:56.27 | znoG | "err i AM my own voip provider... i run a VoIP server!!!" |
| 05:56.36 | znoG | "sir please call Vonage if you need help" |
| 05:56.40 | znoG | *hangup" |
| 05:56.41 | Coccyx | Corydon76-home: I don't see any way to do that in the device config in sip.conf |
| 05:56.57 | Corydon76-home | Coccyx: turn it off on your phone |
| 05:57.40 | Coccyx | I'm looking, but I don't remember being able to do that on these phones. |
| 05:57.48 | Coccyx | but if that works it'd certainly save me some time. |
| 05:58.12 | FuriousGeorge | anyone available to answer a meetme question? |
| 05:58.30 | wasim | do we have to meetme to answer it? |
| 05:58.35 | FuriousGeorge | no |
| 05:58.36 | Coccyx | oh, heh, that is there |
| 05:59.21 | wasim | err ... btw s/fugitivo/FuriousGeorge above |
| 05:59.53 | Corydon76-home | You could also have used the M() parameter to Dial and skipped trying to use the Local channel entirely |
| 06:00.04 | FuriousGeorge | wasim: lets say i get a call on my did and i send it to a meetme room. can i then call an extension? |
| 06:00.15 | FuriousGeorge | oh, you were talking to me when you said "use a call file" |
| 06:00.27 | Coccyx | Corydon76-home: yeah, I was just looking at that too |
| 06:00.35 | Corydon76-home | by running a macro for the answering channel, you could just set the group at that point |
| 06:00.53 | wasim | <PROTECTED> |
| 06:01.03 | Corydon76-home | The problem is that you're setting the group for the calling channel |
| 06:02.10 | FuriousGeorge | wasim: MEETME_AGI_BACKGROUND? did you just make that up? where can i look at this thing? |
| 06:02.16 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 06:02.53 | wasim | FuriousGeorge: yes, yes and show application meetme |
| 06:03.11 | *** join/#asterisk freq (n=freq@bender.2600hz.net) |
| 06:03.54 | wasim | all i do is blab about cricket scores, to give y'all a well rounded view of the world |
| 06:04.16 | wasim | 6/203 :) |
| 06:04.31 | JamesDotCom | who's playing today? |
| 06:04.31 | wasim | au vs sa |
| 06:04.33 | JamesDotCom | which ground? |
| 06:04.44 | *** join/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 06:05.03 | wasim | melbourne |
| 06:05.41 | JamesDotCom | it's good when people blab about the scores, means i dont need to pay much attention :P |
| 06:06.01 | wasim | yeah, and the ulaws go ... huh? wha? |
| 06:06.24 | JamesDotCom | hahah |
| 06:06.36 | JamesDotCom | well, that's what you get for not appreciating only the best sport :P |
| 06:07.24 | JamesDotCom | whatever that last sentence was meant to be |
| 06:07.29 | JamesDotCom | man it's been a long week |
| 06:07.40 | wasim | no other game allows you to chill with a beer, laptop and sun for 5 days and get some serious work done |
| 06:09.18 | JamesDotCom | haha, amen |
| 06:09.34 | JamesDotCom | which country are you in? |
| 06:10.00 | wasim | good 'ol pk |
| 06:10.51 | mgoh | Where can find Asterisk GUI Client documentation? |
| 06:11.04 | wasim | eh, wazzat? |
| 06:11.27 | wasim | oh no, another config tool |
| 06:11.32 | Lots | hey all if five9 a decent company? |
| 06:11.37 | Lots | if = is |
| 06:11.58 | wasim | oh, its the vicidial people |
| 06:12.07 | Lots | ??? |
| 06:12.22 | wasim | Lots: not you, mgoh |
| 06:14.07 | Lots | anyone use asterisk here as a predictive dialer? |
| 06:18.45 | *** join/#asterisk Grubs (n=Miranda@c211-28-119-169.eburwd3.vic.optusnet.com.au) |
| 06:19.29 | Jameno123 | ack, wrong chan :) |
| 06:20.40 | Grubs | Can anyone tell me where to view debug messages logged by RxFax when using rxfax(${FAXFILE}|debug) |
| 06:21.16 | *** join/#asterisk jeffik (n=jeffik@CPE0050babf4cd5-CM014350000760.cpe.net.cable.rogers.com) |
| 06:22.23 | distortion | hmm, what would cause glibc detected *** double free or corruption (out)? seems to happen with chan_ooh323 after a second "reload" * ver 1.2.2? |
| 06:25.33 | wasim | anybody know of a good low fxo-g729/sip ata? |
| 06:32.43 | Peggerr | what do people use for t1 fail over (for when machine A dies the t1 is then routed to machine B) |
| 06:35.37 | wasim | Peggerr: junghanns |
| 06:35.49 | wasim | Peggerr: they make a pri failover switch |
| 06:36.04 | wasim | http://www.junghanns.net/en/ISDNguard_produkt.html |
| 06:36.56 | Peggerr | wasim, i found that before, I could not find out how much it cost, do you know |
| 06:37.43 | wasim | MSRP $599 or so |
| 06:37.52 | wasim | or EU500 or so |
| 06:38.36 | Peggerr | so it is $599 dollars or 500 euros? |
| 06:38.43 | wasim | not sure |
| 06:39.19 | Peggerr | wasim, do you use one? |
| 06:39.48 | wasim | no, none of my customers are important enough |
| 06:40.05 | justinu | lol |
| 06:40.09 | Peggerr | oha that is comforting |
| 06:40.26 | Peggerr | do you know how it determines if a box needs failover |
| 06:40.31 | wasim | and the ones that are handle their e1 through ss7 |
| 06:40.40 | wasim | Peggerr: heartbeat monitoring over a serial link |
| 06:41.31 | Peggerr | aha i see it kind of checks to see if asterisk is alive |
| 06:41.41 | wasim | or the underlying kernel even |
| 06:43.05 | justinu | wasim: how'd you get too smart for your own good? |
| 06:43.38 | wasim | justinu: long and tedious process |
| 06:43.59 | wasim | justinu: you start off by doing the absolute minimum to get by and then improve on that |
| 06:44.04 | justinu | lol |
| 06:44.11 | justinu | sounds like my life |
| 06:44.32 | *** join/#asterisk ThaZZa_Work (n=me@203.80.44.200) |
| 06:44.36 | Peggerr | what is the diffrence between isdn NET and isdn CPE |
| 06:44.38 | wasim | i think its a trait common to most astmasters |
| 06:44.47 | justinu | pegger: customer side, and network side |
| 06:44.54 | Peggerr | oha ok |
| 06:45.01 | justinu | time for bed |
| 06:47.52 | Peggerr | err it is not cool that the ISDNguard needs the serial connectin I wanted to connect it to the console server |
| 06:50.51 | *** part/#asterisk alephcom_ (n=alephcom@host75.net14.mcsnet.ca) |
| 07:03.30 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-122.claranet.co.uk) |
| 07:07.22 | Trazz | when i have softphone not connected and dial that extension is says the person is on the phone.. why is this and how cani fix it? |
| 07:14.19 | *** join/#asterisk duvalin (n=chatzill@c-24-1-207-176.hsd1.tx.comcast.net) |
| 07:15.55 | duvalin | dut.dut.dut |
| 07:16.10 | Trazz | when i have softphone not connected and dial that extension is says the person is on the phone.. why is this and how cani fix it? |
| 07:16.44 | Qwell | Trazz: try repeating it again, somebody might answer |
| 07:16.58 | Trazz | ehehehe |
| 07:17.39 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 07:19.11 | duvalin | question, what specific application(s) would you use an IAXy for? |
| 07:19.46 | duvalin | I believe it's a Digium - S101I |
| 07:19.57 | Qwell | duvalin: tons of situations |
| 07:20.16 | Qwell | anywhere you need a phone |
| 07:25.01 | Trazz | qwell, any examples of complex ivr's around i can view? |
| 07:25.12 | duvalin | so if i understand this right its like an IAX speaking ATA, basically |
| 07:26.42 | Qwell | duvalin: that's exactly what it is...an ata |
| 07:27.18 | Qwell | an iaxy actually works a lot better than most others, when it comes to nats, like at an airport/hotel |
| 07:29.39 | tzafrir_laptop | Trazz, the demo that comes with Asterisk |
| 07:29.58 | tzafrir_laptop | Just complicate it a bit more |
| 07:30.16 | Trazz | when i call my * box it seems sometimes the greeting has already played a few seconds before i hear it... any ideas? |
| 07:30.18 | tzafrir_laptop | Also have a look around the wiki for configuration examples. |
| 07:30.52 | tzafrir_laptop | Trazz, this is the reason for the Wait(1) you see in some samples :-( |
| 07:31.14 | Trazz | i have a wait there now |
| 07:31.16 | Trazz | at 6 |
| 07:31.39 | Trazz | exten => s,1,Wait,6 |
| 07:31.52 | tzafrir_laptop | Wait(6) |
| 07:32.04 | tzafrir_laptop | exten => s,1,Wait(6) |
| 07:32.16 | tzafrir_laptop | though maybe your syntax is actually valid |
| 07:32.32 | tzafrir_laptop | Trazz, look at the trace in the CLI: |
| 07:32.40 | tzafrir_laptop | asterisk -rvvv |
| 07:33.00 | tzafrir_laptop | (that is: set verbosity to at least 3) |
| 07:33.28 | duvalin | how is this done (nat traversal).. is this an inherent property of the IAX spec. |
| 07:33.35 | tzafrir_laptop | and then you see basically everything that happens in the dialplan |
| 07:33.46 | Qwell | duvalin: the problem with sip (and others), is the extra rtp stream |
| 07:33.56 | Qwell | since iax only uses one port...it's not a problem |
| 07:34.04 | *** join/#asterisk EriSan (n=erisan@151.8.109.91) |
| 07:34.04 | Qwell | duvalin: if you can use http, you can use iax |
| 07:34.09 | Qwell | (generally speaking) |
| 07:34.21 | tzafrir_laptop | duvalin, this is an inherent property of the fact that it uses just one UDP "stream", with the RTP data embedded in it |
| 07:35.05 | duvalin | ok, i think im understanding this better now |
| 07:35.06 | tzafrir_laptop | almost. There is still no way to tunnel IAX over an http proxy |
| 07:35.59 | tzafrir_laptop | Anybody up to the task :-) ? |
| 07:36.00 | *** join/#asterisk M-I-A-- (n=mai@wsp05975102wss.cr.net.cable.rogers.com) |
| 07:36.07 | Qwell | stunnel it |
| 07:36.22 | Qwell | any sane http proxy will allow https |
| 07:37.34 | tzafrir_laptop | but https is still tcp. And AFAIR asterisk does not support IAX over TCP |
| 07:38.21 | Qwell | stunnel on both sides |
| 07:38.29 | *** join/#asterisk lesouvage (n=lesouvag@cc341200-a.assen1.dr.home.nl) |
| 07:38.37 | tzafrir_laptop | can stunnel do UDP? |
| 07:38.45 | Qwell | dunno, why not? |
| 07:38.49 | Qwell | iax>stunnel>https proxy>stunnel>iax |
| 07:39.27 | DarkFlibble | ssh can act as a socks proxy... socks can do udp |
| 07:39.59 | Qwell | ssh over a telnet proxy |
| 07:40.01 | Qwell | heh |
| 07:40.29 | FuriousGeorge | Qwell: you know if this astjab Patch for head branch and 1.2rc1 will work with 1.2.1? |
| 07:40.34 | FuriousGeorge | hi, by the way |
| 07:40.40 | Qwell | FuriousGeorge: should...try it |
| 07:40.42 | Qwell | or ask mog |
| 07:40.59 | FuriousGeorge | was gonna but i aint seenem for a bit |
| 07:41.20 | Qwell | yeah, haven't seem him today |
| 07:41.39 | *** join/#asterisk shekhar (n=shekhar@221-128-138-134.exatt.net) |
| 07:42.29 | FuriousGeorge | pointed out to him that asterisk sounds werent making it from make install to sounds dir, maybe hes fixing |
| 07:42.41 | duvalin | was just reading specs on IAX at voip-info.org.. |
| 07:42.42 | tzafrir_laptop | ssh can do UDP??? |
| 07:43.08 | DarkFlibble | socks can do udp... ssh can do socks proxying... |
| 07:43.29 | DarkFlibble | so ssh should do udp |
| 07:43.40 | DarkFlibble | tunnelled via a tcp connection |
| 07:45.03 | duvalin | IAX2 seems to be a very well thought-out/developed protocol |
| 07:45.27 | DarkFlibble | IAX2 just works... |
| 07:45.30 | JamesDotCom | hahahah |
| 07:45.31 | JamesDotCom | ahahahah |
| 07:45.31 | JamesDotCom | hahah |
| 07:45.33 | JamesDotCom | *cough*\ |
| 07:45.54 | DarkFlibble | IAX1 though... heard a few people swear about it... |
| 07:46.14 | DarkFlibble | but that predates me a little |
| 07:47.35 | iDunno | morning |
| 07:47.53 | duvalin | seems grass-roots/collaborative/community-based efforts.. |
| 07:48.06 | duvalin | ...cut straight through all the BS and politics of standards bodies and orgs. |
| 07:50.07 | DarkFlibble | thats because generally those types of projects have 1 person develop the initial concept... and only when its working (in some form) do others start to contribute.... |
| 07:50.21 | DarkFlibble | most of the bs is in planning in bigger projects |
| 07:51.04 | DarkFlibble | but even open source projects can have a lot of in fighting once they get big... |
| 07:51.20 | DarkFlibble | look at the kernel or apache etc... |
| 07:51.30 | duvalin | yep |
| 07:51.34 | duvalin | true |
| 07:51.42 | DarkFlibble | even asterisk to some extent... |
| 07:51.56 | DarkFlibble | there are quite a few forks out there |
| 07:53.10 | Qwell | none that are successful |
| 07:53.54 | DarkFlibble | Xorg was a fork of a big project that was sucessful... so its not true that all forks are destined to fail... |
| 07:54.18 | DarkFlibble | but they did take a lot of the developers with them... |
| 07:54.22 | Trazz | i am trying to dial out and have some issues. if i use 18135551212 i get the person you are calling is unavilable. when i use 9 in front of it then broadvoice dont like it |
| 07:54.26 | Trazz | because of the 9 |
| 07:54.52 | DarkFlibble | Trazz, ${EXTEN:1} may help... |
| 07:55.42 | Trazz | exten => _91NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30) is my line now |
| 07:56.25 | DarkFlibble | Trazz, does it work without the 9? |
| 07:56.27 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 07:56.35 | DarkFlibble | hey fenlander |
| 07:56.39 | Trazz | with 9 it looks like it passes the 9 to broadvoice |
| 07:56.44 | Trazz | and they kick it back |
| 07:57.02 | Trazz | yet i have ignorepat => 9 |
| 07:57.02 | DarkFlibble | Trazz, exten => _91NXXNXXXXXX, 1, dial(SIP/${EXTEN:1}@sip.broadvoice.com,30) |
| 07:57.04 | fenlander | DarkFlibble: hi |
| 07:57.34 | DarkFlibble | Trazz, but have a look at your sip debugging |
| 07:57.40 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 07:57.49 | DarkFlibble | fenlander, hows the fens today? |
| 07:58.12 | Trazz | what is the command for that? |
| 07:58.15 | fenlander | misty, cold, pretty much as normal :) |
| 07:58.29 | DarkFlibble | Trazz, sip debug |
| 07:58.37 | DarkFlibble | and sip no debug to turn it off |
| 07:58.51 | DarkFlibble | just cold in Leicester... |
| 07:58.56 | DarkFlibble | fairly clear skys tho |
| 07:59.42 | Trazz | DarkFlibble, that fixed. it... what did that command do? |
| 07:59.57 | DarkFlibble | ignore the first digit |
| 08:00.09 | Trazz | nice..thanks |
| 08:00.40 | DarkFlibble | Trazz, you mean the ${EXTEN:1} or the sip debug? |
| 08:00.51 | DarkFlibble | the :1 is the one I referred to |
| 08:00.52 | Trazz | <PROTECTED> |
| 08:01.05 | Trazz | yep |
| 08:01.07 | DarkFlibble | have a look at the manual a little |
| 08:01.17 | Trazz | okie |
| 08:01.17 | DarkFlibble | might make it easier in future |
| 08:03.24 | *** join/#asterisk Bambr (n=Bambr@213-35-236-199-dsl.end.estpak.ee) |
| 08:04.26 | *** join/#asterisk cica (i=cica@81.30.249.241) |
| 08:08.12 | thazza | Hey does anyone here. know about the australian company Engin? |
| 08:10.44 | DarkFlibble | not me |
| 08:11.43 | thazza | darn.. they used to have a peer arrangement with fwd.. and it doesn't seem to be working |
| 08:12.06 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 08:12.13 | DarkFlibble | which direction? |
| 08:12.20 | DarkFlibble | FWD -> Engin? |
| 08:13.01 | thazza | yeah |
| 08:13.13 | thazza | well thats what it says on FWD's peer page |
| 08:13.32 | DarkFlibble | Often providers initially say yes and then find they start to lose money and block it |
| 08:13.44 | DarkFlibble | Vonage did iirc |
| 08:13.58 | DarkFlibble | and they weren't the first |
| 08:14.54 | thazza | i am guessing i will have to contact them and find out. :-( |
| 08:15.02 | thazza | Thanks DarkFlibble .! |
| 08:15.04 | DarkFlibble | probably |
| 08:15.06 | DarkFlibble | np |
| 08:15.08 | *** join/#asterisk zoa (n=kkk@pirus.securax.be) |
| 08:15.47 | thazza | DarkFlibble: Is there anyway you can see if the call is failing. or it will just not connect from fwd -> engin? |
| 08:16.02 | DarkFlibble | there are sip diagnostic tools... |
| 08:16.16 | DarkFlibble | you can essentially do a sip traceroot... |
| 08:16.49 | DarkFlibble | you might get some data by turning on SIP debugging... |
| 08:16.53 | DarkFlibble | but its not certain |
| 08:17.00 | thazza | ok.. so i could see the packet going from mine, to fwd and then trying to engin. |
| 08:17.33 | DarkFlibble | try sip debug |
| 08:17.42 | DarkFlibble | its free... its built into asterisk... |
| 08:18.00 | DarkFlibble | and its availible now... |
| 08:18.15 | DarkFlibble | sip no debug to turn it off again |
| 08:18.19 | DarkFlibble | :P |
| 08:18.50 | thazza | DarkFlibble: okie thanks mate.. once again.. i tried it a while ago.. yet without debug mode.. will have another play.. :-) |
| 08:19.09 | DarkFlibble | sipsak is a tool for debugging sip |
| 08:19.21 | DarkFlibble | but not sure how much extra info it would give you |
| 08:19.44 | *** join/#asterisk aurelien_ (n=aurelien@lev92-1-81-57-180-153.fbx.proxad.net) |
| 08:19.50 | aurelien_ | Hi all |
| 08:20.14 | DarkFlibble | hi |
| 08:23.38 | DarkFlibble | http://sipsak.org/sipsak_mint_big.jpg <-- not quite the tool I was looking for... |
| 08:23.39 | DarkFlibble | :P |
| 08:23.54 | *** join/#asterisk usam (n=usam@203.156.61.204) |
| 08:24.10 | Trazz | DarkFlibble, when i call a sip phone extesion that is not logged in it says the person is on the phone... |
| 08:24.57 | DarkFlibble | when you execute a dial command it returns the status... you can use this status to do more fine grained control of messages and such |
| 08:25.20 | DarkFlibble | one sec... I'll find an example |
| 08:25.38 | Trazz | ok thanks |
| 08:25.39 | Trazz | Jan 20 02:37:40 NOTICE[32098]: app_dial.c:1010 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 08:25.48 | Trazz | thats whati get which is obvious |
| 08:26.05 | Trazz | i would rather it go to voice mail |
| 08:26.10 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
| 08:26.18 | Math` | Trazz: then make it go to voicemail |
| 08:26.29 | DarkFlibble | http://www.voip-info.org/wiki/index.php?page=Asterisk+variable+DIALSTATUS |
| 08:26.58 | DarkFlibble | http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+goto <--example half way donw here |
| 08:28.08 | aurelien_ | does anybody know how to configure kphone? i don't see any field to set my password for asterisk account |
| 08:28.10 | Trazz | got it.. so do i ignore the no route to destination? |
| 08:28.33 | DarkFlibble | Trazz, its up to you what you choose to do with each of the states... |
| 08:28.49 | DarkFlibble | what would be most logical for your users? |
| 08:29.12 | Trazz | get voicemail |
| 08:29.20 | DarkFlibble | so do that then... |
| 08:29.32 | Trazz | :) thanks |
| 08:29.42 | DarkFlibble | np |
| 08:30.39 | DarkFlibble | now if I could only find a job... I'd be happy... |
| 08:30.42 | DarkFlibble | :P |
| 08:30.46 | Trazz | heheh |
| 08:35.50 | *** join/#asterisk scanna (n=scannach@81-174-16-211.f5.ngi.it) |
| 08:37.59 | *** part/#asterisk FranckM (n=franck@202.62.0.1) |
| 08:42.50 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:43.10 | *** join/#asterisk yxa (n=diablo@58.185.90.98) |
| 08:43.44 | yxa | if i have 2 sip phones connected to * and when I dial one, it rings but when I pick up there's no voice. what could be a likely reason? |
| 08:45.40 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:48.22 | Mavvie | Module Size Used by |
| 08:48.23 | Mavvie | zaptel 224004 504 |
| 08:48.30 | Mavvie | do I see a problem there? |
| 08:48.47 | Mavvie | let's reboot this box first before I can upgrade to 1.2.2 |
| 08:49.22 | iDunno | yxa: you're not talking in to the phone? :) |
| 08:53.41 | *** join/#asterisk Alex1 (n=A@188.90.233.220.exetel.com.au) |
| 08:53.50 | kaldemar | yxa: check codec negotiation and your fw. asterisk uses ports 10000-20000 for rtp traffic by default. |
| 08:54.59 | FuriousGeorge | ever since i upgraded to 1.2 i cant get atxfer and blindxfer in features.conf working |
| 08:55.21 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 08:56.34 | Alex1 | hello all i am having this problem with astrix 3.0 its installed on a hdd when it boots he file system is ro so i make it rw mount -o rw,remount / i make some changes in the /etc/asterisk/sip.conf save them then check to see if its saved then i mount -o ro,remount / then reboot when it boots up my chages are not there ? why ? |
| 08:57.08 | *** join/#asterisk brettnem (n=brettnem@72.29.102.158) |
| 08:58.23 | FuriousGeorge | b/c there is no asterisk 3.0? |
| 08:58.46 | yxa | kaldemar cant be ports. all devices on same switch |
| 08:59.14 | Alex1 | how do i tell what ver it is |
| 09:00.22 | Alex1 | hello all i am having this problem with astrix latest ver from there web site the iso its installed on a hdd when it boots he file system is ro so i make it rw mount -o rw,remount / i make some changes in the /etc/asterisk/sip.conf save them then check to see if its saved then i mount -o ro,remount / then reboot when it boots up my chages are not there ? why ? |
| 09:01.08 | *** join/#asterisk fanguin (n=user@p548F6922.dip.t-dialin.net) |
| 09:01.36 | Peggerr | are most t1 lines unlimited in the us and canada or are there diffrent calling plans for t1 lines? |
| 09:04.24 | Alex1 | has anyone had this problem ? |
| 09:04.26 | *** join/#asterisk e3g (i=ee@u15157627.onlinehome-server.com) |
| 09:04.43 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 09:04.51 | e3g | I want to install RAWPLAYER.....how to uninstall mpg123? |
| 09:06.14 | Peggerr | Alex1, go into the asterisk cli and type show version |
| 09:06.23 | Peggerr | e3g, what distro |
| 09:06.33 | tzafrir_laptop | e3g, start by installing the rawplayer. after it works, simply tell astersik not to use mpg123 |
| 09:06.50 | tzafrir_laptop | only then worry about uninstalling it |
| 09:07.19 | tzafrir_laptop | s/installing rawplayer/configuring rawplayer/ |
| 09:07.25 | e3g | Peggerr: Red hat |
| 09:08.24 | tzafrir_laptop | hmm, jbot's s should only be activated if it actually substituted something... |
| 09:08.40 | e3g | how to tell asterisk not to use mpg123? |
| 09:09.14 | tzafrir_laptop | e3g, you can use the "custom" method on musiconhold.conf and use your own script |
| 09:09.15 | e3g | I'm fedup of this msg "monmp3thread: Request to schedule in the past?!?!" |
| 09:09.22 | tzafrir_laptop | I'm not sure about native musiconhold |
| 09:11.14 | e3g | does rawplayer play mp3 files? |
| 09:16.24 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 09:20.02 | *** join/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 09:20.13 | e3g | how to convert MP3 files to RAW Files????? SOX says "sox: Failed reading 082.MP3: Do not understand format type: MP3 |
| 09:20.13 | e3g | " |
| 09:22.43 | *** join/#asterisk Little-L_ (n=daniel@0x50a471db.svgnxx1.adsl-dhcp.tele.dk) |
| 09:22.53 | *** join/#asterisk srt (n=nobody@18.120.9.213.dsl.getacom.de) |
| 09:23.55 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 09:24.24 | e3g | how to convert MP3 files to RAW Files????? SOX says "sox: Failed reading 082.MP3: Do not understand format type: MP3" |
| 09:26.57 | *** join/#asterisk thosa (n=thosa@p54879AC3.dip0.t-ipconnect.de) |
| 09:28.23 | *** join/#asterisk wizhippo_ (n=wizhippo@Quebec-HSE-ppp237166.qc.sympatico.ca) |
| 09:28.24 | thosa | hi there. short question. can i use my PCI ISDN card with winbond W6692cf chipset to work in NT mode ? |
| 09:28.37 | aurelien_ | how can i use gizmoproject to connect to my asterisk with SIP? By default it connect to gizmo server..... |
| 09:28.48 | thosa | or is it better to have an ISDN board with the cologne chipset ? |
| 09:29.03 | thosa | i mean in case of supported hardware / software combination ? |
| 09:29.56 | thosa | aurelien_: as far as i know gizmo is "branded" and not able to connect to other networks than the gizmo servers |
| 09:30.07 | thosa | that is also a reason why it is free |
| 09:30.26 | aurelien_ | oh ok... |
| 09:30.43 | thosa | but that is only AFAIK |
| 09:31.00 | aurelien_ | and do you know good client for Mac OS X? |
| 09:31.37 | thosa | sorry not that i know of, but for that case it might make sense to use google |
| 09:32.13 | gaupe | thosa: only cologne chips supports NT-mode |
| 09:32.13 | thosa | i saw several comparings of different clients on different os |
| 09:32.27 | thosa | gaupe: are you really sure on that? if so, i have to send my two new cards back |
| 09:32.40 | aurelien_ | ok thosa, thx a lot for your answer |
| 09:32.50 | gaupe | e3g: convert it first with with lame og mpg321, you'll find the info on voip-info.org |
| 09:32.51 | zoa | aurelien_: is an iax2 client also ok ? |
| 09:32.56 | thosa | aurelien_: no prob |
| 09:33.07 | gaupe | thosa: not 100%, but 95% - just looked at it yesterday |
| 09:33.12 | aurelien_ | zoa: nope, only sip |
| 09:33.35 | thosa | gaupe: ok. thanks for help. i am sad now anyway. but the truth can be hard! :-) |
| 09:33.56 | gaupe | hehe, those isdn-cards are dirt cheap anyway |
| 09:34.24 | e3g | gaupe: thanks |
| 09:34.37 | thosa | gaupe: yes but that is my first test installation. i am running an AVM FRITZ!BOX if you know that one |
| 09:34.48 | thosa | gaupe: but i am sure, asterisk can do more.. :-) |
| 09:34.51 | gaupe | know of it - have to run, meeting :) |
| 09:35.00 | thosa | gaupe: thanks. bye |
| 09:36.36 | *** join/#asterisk redman (i=redman@dhcp-0-2-b3-9a-4a-5b.cpe.quickclic.net) |
| 09:42.58 | *** join/#asterisk Modcuts (n=sam@proporta.gotadsl.co.uk) |
| 09:42.59 | *** join/#asterisk Fraeggl (n=Fraeggl@rkom.r-kom.de) |
| 10:02.30 | zoa | goddamn crappy iax2 :( |
| 10:06.35 | thazza | whats up zoa ?? |
| 10:08.43 | zoa | experienced the problem with iax2 on links with large delays again |
| 10:08.49 | zoa | timestamps are broken |
| 10:08.54 | zoa | pfft |
| 10:08.55 | zoa | :( |
| 10:09.42 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 10:11.09 | *** join/#asterisk Peggerr (n=peg@pool-68-163-232-33.bos.east.verizon.net) |
| 10:19.08 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 10:19.23 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 10:19.57 | Mimmus | good morning, I'm getting a lot of 'h' as destination number in my CDR logs |
| 10:20.14 | Mimmus | Some time ago I solved this problem but now I'm not able to remember anymore... grrrr... |
| 10:20.50 | *** join/#asterisk fulgas (n=fulgas@209.8.233.242) |
| 10:21.20 | PoWeRKiLL | since asterisk > 1.2.0 I got Invalid or unknown command when getting value from DIALSTATUS any idea ? |
| 10:23.07 | *** join/#asterisk zeedo (n=zeedo@reboot-robot.net) |
| 10:26.59 | *** join/#asterisk apardo (n=apardo@62.97.121.93) |
| 10:31.17 | RoyK | PoWeRKiLL: prolly your own fault |
| 10:41.17 | *** join/#asterisk shekhar_ (n=shekhar@221-128-139-53.exatt.net) |
| 10:41.32 | *** join/#asterisk DoDo (n=Koc_asLa@213.186.176.25) |
| 10:48.12 | *** join/#asterisk fenlander (n=fenlande@82.152.81.57) |
| 10:50.29 | *** join/#asterisk apardo (n=apardo@62.97.121.93) |
| 10:53.14 | Modcuts | what would a line look like sending an extention straight to a digital recepitionist "exten => number,what,what"? |
| 10:56.50 | *** join/#asterisk sac|h0p|werk|afk (n=h0p@S01060002b3eb8fa7.ok.shawcable.net) |
| 10:56.52 | *** join/#asterisk Givur (n=mail@Ga8cf.g.pppool.de) |
| 10:56.53 | Givur | <PROTECTED> |
| 10:56.55 | Givur | Hi all |
| 10:57.29 | DarkFlibble | Modcuts, depends on exactly what you want to do |
| 10:58.22 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 10:58.51 | Modcuts | i want to send an external number to a digital receptionist i have setup called test and then i know how to make it ring extensions if the hold is a certain amount. |
| 10:58.57 | Modcuts | make sense? |
| 10:59.07 | *** part/#asterisk halorgium (i=tim@nuke.halorgium.net) |
| 11:00.57 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 11:02.17 | DarkFlibble | the hold? |
| 11:02.17 | DarkFlibble | I assume english isn't your fist language |
| 11:02.20 | DarkFlibble | k...first of all... get the external number hitting your asterisk box... |
| 11:02.26 | DarkFlibble | debugging is the key here.... |
| 11:02.30 | DarkFlibble | then make a context with your required prompts and such... |
| 11:02.34 | DarkFlibble | I assume that if they fail to type and extension you want it to fall through toa default... |
| 11:02.40 | DarkFlibble | WaitExten can help there |
| 11:02.42 | DarkFlibble | or setting an absolute timeout and t extension |
| 11:04.27 | Modcuts | i just need to know the context of the line that allows the digital recepitionist to be rung first |
| 11:04.35 | Modcuts | or is it a series of lines? |
| 11:04.49 | DarkFlibble | it depends which context you place it in... |
| 11:04.49 | Modcuts | and english is my first language just shit at explaining what i'm trying to do |
| 11:05.19 | Modcuts | well i have the exten => in extension.conf under the from-pstn context |
| 11:05.29 | Modcuts | but i can assign the incoming number to any context. |
| 11:06.27 | DarkFlibble | I normally dump each group of numbers into a seperate context then use a goto to jump to a main-main context... |
| 11:06.37 | Givur | I have a problem with my goto construction. I have setup a dialplan what use a Goto to dial a alternate line when the primary line is busy. This is working fine sofar, just that I have the problem that I lose the ${EXTEN} Informations and that cause trouble with my CDR. Is there any suggestion what I need todo for fix that? My Dialplan for that is avaible at http://pastebin.com/514396. |
| 11:06.40 | DarkFlibble | means I can redirect a group of incoming lines easily |
| 11:07.00 | DarkFlibble | Modcuts, okay to pm? |
| 11:07.35 | Modcuts | yep |
| 11:08.50 | tzafrir_laptop | Givur, so save that value in another local variable |
| 11:09.10 | tzafrir_laptop | Use Set |
| 11:09.52 | tzafrir_laptop | And later on use that variable instead of EXTEN, or restore its value to EXTEN |
| 11:11.23 | Givur | *nods* That I have do already in the dialplan, that works fine there(Line 2, Saving it as CallNo). I only have a problem with the cdr |
| 11:11.26 | Givur | *hmms* |
| 11:12.36 | Givur | Oh, restore the value |
| 11:13.53 | *** join/#asterisk Assid (n=assid@203.115.64.10) |
| 11:13.54 | Assid | heya |
| 11:14.54 | Assid | is it possible have a SIP/blah/18001111111|15&SIP/2020|30 ? |
| 11:15.59 | Assid | with 2 different timeouts |
| 11:16.58 | DarkFlibble | Assid, you could call the the longer one with the difference between the two then both... |
| 11:17.14 | DarkFlibble | but it might occassionally glitch... |
| 11:17.47 | Assid | well.. i need both to ring simultanously |
| 11:18.19 | Givur | tzafrir_laptop: Ok works. Thanks sofar :) |
| 11:18.55 | DarkFlibble | do the opposite to what I suggested then... |
| 11:19.39 | Assid | DarkFlibble: okay what if i want both on for say 15 seconds.. do i just change the parameter to 15 ? will that cause a glitch? |
| 11:21.17 | DarkFlibble | basicly the glitch I was talking about was that you essentially have race condition... once the first set of dials timesout and the second one kicks in its possible that they may pick up the phone.... in which case the call will fall through |
| 11:21.53 | Assid | okay how do i overcome it? |
| 11:22.01 | Assid | mention both at 30? |
| 11:22.07 | DarkFlibble | to my knowledge (most of which is over a year old) that is the only easy way to do it |
| 11:22.08 | Assid | or is there a global way? |
| 11:22.22 | DarkFlibble | but someone else may have a better solution |
| 11:28.23 | *** join/#asterisk jyukes_ (n=jameshot@pool-138-89-229-250.atc.east.verizon.net) |
| 11:28.49 | *** join/#asterisk Camisa (n=Camisa@c-67-186-94-173.hsd1.in.comcast.net) |
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| 11:57.37 | Assid | i guess it should od |
| 12:00.25 | *** join/#asterisk sysdebug (n=sysdebug@200.163.193.247) |
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| 12:04.29 | puzzled | morning |
| 12:04.47 | jbalcomb | morning |
| 12:04.54 | areski | g morning too |
| 12:07.31 | jbalcomb | time to monkey with rxgain & txgain.. |
| 12:09.57 | caio1982 | jbalcomb: welcome to the club |
| 12:10.45 | jbalcomb | caio1982 thanks. its fund to be here except the part where filing complaints takes up half my day.. every day. |
| 12:10.51 | jbalcomb | s/fund/fun |
| 12:10.59 | Camisa | good morning. |
| 12:11.42 | sivana | morning |
| 12:12.29 | caio1982 | jbalcomb: i meant to the club of people that goes nuts with it (rx/tx stuff) :) |
| 12:12.45 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
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| 12:14.20 | jbalcomb | caio1982 yes, well, hopefully going nuts with rx/tx will fix these joyous echo,jitter, pausing call problems |
| 12:15.25 | caio1982 | jbalcomb: share the results with us later, would be great to see good results :) |
| 12:19.52 | *** join/#asterisk A-jay (n=quirc@62.217.245.194) |
| 12:20.53 | *** join/#asterisk CALLER (n=Camisa@c-67-186-94-173.hsd1.in.comcast.net) |
| 12:23.33 | Modcuts | Where is the DIR CONTEXT varible defined in asterisk? |
| 12:23.42 | CALLER | dunno.. |
| 12:24.54 | jpk | Hi guys. I could use some help with app_sms. |
| 12:25.01 | jpk | Anyone here know how to handle that? |
| 12:25.25 | jpk | I would like * to send sms to my internal landline gigaset. |
| 12:25.31 | CALLER | nope.. I'm way too noob. |
| 12:25.36 | jbalcomb | ditto |
| 12:25.38 | CALLER | good luck. |
| 12:28.15 | CALLER | anybody here have experience setting up asterisk with a single sip phone? |
| 12:29.18 | caio1982 | let's search a bit for it hehe |
| 12:29.22 | CALLER | DarkFlibble: I'm on gentoo. |
| 12:29.31 | DarkFlibble | CALLER, poor you... |
| 12:29.33 | DarkFlibble | :P |
| 12:29.38 | CALLER | DarkFlibble: what's the best configuration tool for asterisk? nano? |
| 12:30.00 | DarkFlibble | any text editor works... vi is my pref... but nano should work |
| 12:30.10 | CALLER | cool. |
| 12:30.43 | *** join/#asterisk bofh42 (n=bofh42@p5482B912.dip0.t-ipconnect.de) |
| 12:30.49 | CALLER | Well, here's my story. I subscribe to a national voip serivice... and I have my sip settings written down. I tested them with twinkle and kphone... and was able to dial my cell phone without an outbound call from the computer. |
| 12:31.11 | CALLER | How do I setup asterisk so that whenever I call on the line, asterisk answers and immediately starts recording? |
| 12:31.57 | DarkFlibble | you will need to emulate a sip phone... |
| 12:32.08 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 12:32.13 | DarkFlibble | so a register line in sip.conf |
| 12:32.23 | DarkFlibble | and a block for the provider... |
| 12:32.30 | DarkFlibble | okay to pm? |
| 12:32.35 | CALLER | yes |
| 12:32.45 | CALLER | go ahead, it should work unless they need me to register my nick first. |
| 12:35.46 | DarkFlibble | not really... |
| 12:35.49 | jbalcomb | welp, with rxgain -6 & txgain -22 ztmonitor shows an average around 15% for rx and average 75% for tx |
| 12:35.50 | DarkFlibble | have a look |
| 12:36.18 | DarkFlibble | you wont be able to reply in a om without registering... but in channel should be fine |
| 12:37.46 | CALLER | Darn... so what I was typing in the PM you weren't able to read? |
| 12:37.55 | DarkFlibble | nope |
| 12:38.12 | *** join/#asterisk JooZoo (n=chatzill@82-203-171-162.dsl.gohome.fi) |
| 12:38.20 | DarkFlibble | I made that mistake the other day... |
| 12:38.34 | *** join/#asterisk OnuR (n=__Neo__@62.220.216.128) |
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| 12:39.23 | *** join/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net) |
| 12:39.41 | CALLER | so I haven't read the manual yet... what are the main config files? I'm editing sip.conf and it's talking about extensions.conf already too. |
| 12:40.10 | DarkFlibble | sip.conf iax.conf and extensions.conf are the most common |
| 12:45.11 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 12:45.11 | *** join/#asterisk viperdude (n=jon@borat.enta.net) |
| 12:45.42 | *** join/#asterisk astr (n=ts@59.93.65.229) |
| 12:46.55 | *** join/#asterisk fugitivo (n=ajf@201.255.177.156) |
| 12:47.02 | tzafrir_laptop | CALLER, extensions.conf basically wires things together. sip.conf, iax.conf, zap.conf etc. are configuration files of specific channels |
| 12:47.23 | tzafrir_laptop | extensions.conf is the dialplan: tells what to do in a call when it s in the PBX |
| 12:48.27 | tzanger | I'm baaaaaaaaaaaaaaack |
| 12:48.33 | DarkFlibble | although most of the old extensions.conf examples wont work on the CVS head atm due to changes in stuff like timeouts and DB access |
| 12:48.39 | astr | hello, we are trying to transcode GSM to G711 and vice versa? Do you think it is a good suggestion? The reason we are doing transcoding is because we are not able to find good PSTN providers for GSM codec |
| 12:51.54 | CALLER | astr: on the fly transcoding? |
| 12:52.09 | tzanger | CALLER: is there any other? |
| 12:52.24 | CALLER | tzanger: I'm new |
| 12:52.25 | astr | Caller: yes |
| 12:52.35 | DarkFlibble | astr, it shouldn't be too bad.. since G711 is almost uncompressed... since you get most problems with quality when going from one heavy compression scheme to another... ie g729 <-> gsm... |
| 12:53.05 | DarkFlibble | CALLER, asterisk will automatically transcode codecs if it can.... |
| 12:53.14 | tzanger | ulaw is PSTN quality. going from gsm to ulaw and back isn't bad. yes it's a quality hit but I've been running an office of 30 people with it for a year and they're fine with it |
| 12:53.22 | *** join/#asterisk znoG (n=gs@33-138-114-200.fibertel.com.ar) |
| 12:53.29 | tzanger | I received complaints when I used g729 and ilbc, so gsm it is. |
| 12:53.54 | sivana | heh |
| 12:53.56 | DarkFlibble | people are generally used to gsm quality if they use cell phones |
| 12:54.01 | sivana | cheap bastard |
| 12:54.03 | astr | I am trying to transcode GSM and G729 |
| 12:54.10 | DarkFlibble | ilbc can sound a little odd... |
| 12:54.10 | tzanger | well g729 is also used in cell phones |
| 12:54.12 | CALLER | for my SIP connection from AT&T... how do I know if it would be defined as a user, peer or friend connection? |
| 12:54.18 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:54.38 | DarkFlibble | tzanger, it is? you got an example? |
| 12:54.48 | tzanger | DarkFlibble: what do you mean? |
| 12:54.58 | DarkFlibble | of g729 in cell phones |
| 12:55.08 | *** join/#asterisk _deg_ (n=deg@200.163.193.247) |
| 12:55.14 | tzanger | DarkFlibble: CDMA phones typically use g729, whereas gsm phones use, well, gsm. |
| 12:55.21 | astr | GSM and G711 would be OK but it takes a lot of bandwidth and thats the reason we are planning to go with G729 but am not sure about the transcoding between gsm and G729 |
| 12:55.30 | DarkFlibble | that explains why I have never seen it... |
| 12:55.38 | DarkFlibble | Only really GSM in the EU |
| 12:55.48 | tzanger | yeah |
| 12:57.44 | JMcA | DarkFlibble: gsm is getting to be rather common in the US as well |
| 12:57.48 | astr | Guys, one more question about teh SIP.conf for Vonage. It registers fine but when a call is received, it does not send user name in the invite packet during the MDS hash challenge |
| 12:57.52 | JMcA | both Cingular and T-Mobile are GSM networks |
| 12:58.38 | tzanger | JMcA: it's definitely a strong battle between the two |
| 12:58.44 | tzanger | japan is cdma too I think |
| 12:58.50 | DarkFlibble | I still think everyone should use lpc10! :P |
| 12:59.23 | astr | was anyone successful with Vonage configuration with *? |
| 12:59.37 | tzanger | I was not aware that vonage was playing nice iwth asterisk |
| 12:59.42 | JMcA | I think, until we can push the PSTN into irrelevance, that most people should stick with g.711, personally |
| 13:00.36 | *** join/#asterisk SkalTura (i=none@a85-156-173-3.elisa-laajakaista.fi) |
| 13:00.41 | SkalTura | hiya |
| 13:00.47 | DarkFlibble | g.711 is nice as long as you have enough bandwidth... |
| 13:00.51 | astr | tzanger: I was able to register, everything seems to be fine but asterisk sends blank username when it sends nounce etc. and vonage replies back with 407 |
| 13:00.52 | SkalTura | damn didn't get aroudn to work with asterisk today much :( my testing server died... |
| 13:01.00 | SkalTura | atleast i got that one ready: http://mailx.artichost.net/ (or atleast almost) |
| 13:01.04 | *** part/#asterisk JooZoo (n=chatzill@82-203-171-162.dsl.gohome.fi) |
| 13:01.43 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
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| 13:03.23 | astr | guys, is there a good provider of PSTN minutes? Most of them I found were too small and unreliable. Is anybody using for mass productions like 5k mins per month |
| 13:03.46 | DarkFlibble | astr, where you based? |
| 13:04.26 | astr | dark..:US |
| 13:04.38 | tzanger | astr: as I don't use vonage, I don't know :-) |
| 13:04.39 | DarkFlibble | probably not me then |
| 13:04.42 | viperdude | hi guys |
| 13:05.22 | JMcA | 5k mins per month is not large to most real providers, I'm sorry to say |
| 13:05.27 | SkalTura | PSTN minutes? |
| 13:06.07 | JMcA | my company (not a telco) does in the ballpark of 3 million minutes per month, to give a counter-example |
| 13:07.33 | tzanger | http://www.theinquirer.net/?article=29131 |
| 13:07.36 | tzanger | unbelievable |
| 13:08.04 | astr | JMcA: mind sharing your company name |
| 13:08.45 | JMcA | Appriss |
| 13:09.30 | DarkFlibble | tzanger, the arguments about linux don't hold much water... since they *can't* have wizards in linux |
| 13:09.43 | DarkFlibble | daemons tho... thats another story |
| 13:09.58 | JMcA | we provide a service where we notify (primarily) victims of crime when the perpetrator of the crime gets out of jail...called VINE |
| 13:10.20 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 13:12.13 | skeffling | Has anyone had problems with 'clipping' (a second or 2 of one-way silence in a call) - I'm using a TE406, zaptel/asterisk 1.2.2. I'm gonna turn the gain down tonight to see if that helps. |
| 13:12.49 | DarkFlibble | Is linux a good buddhist OS then... since it has enlightenment! |
| 13:12.50 | fugitivo | JMcA: what do you say to the victims? "run!" ? |
| 13:13.08 | h3x | skeffling: its probably echo training |
| 13:13.15 | JMcA | fugitivo: hehe...we just give them the information...what they do with it is up to them :) |
| 13:13.36 | DarkFlibble | JMcA, do they subscribe to the info? |
| 13:14.20 | JMcA | darkskiez: yes, they have to register to be notified...we only gather enough information to get the notification to them, but they do have to do the registration to be notified |
| 13:14.33 | DarkFlibble | k |
| 13:14.45 | darkskiez | JMcA: Oh thanks, good to know. |
| 13:14.53 | skeffling | h3x, it happens mid-call and more than once sometimes - could this be traning? -even though I have echotraining off in zaptel.conf |
| 13:15.01 | *** join/#asterisk af_ (n=af@ip-142-84.sn1.eutelia.it) |
| 13:15.02 | JMcA | we also provide inbound capability, where they can call to get current status information about the offender |
| 13:15.19 | h3x | it could be the jitter buffer then |
| 13:15.47 | JMcA | we try very hard to avoid any appearance of spam or the telephony equivalent |
| 13:15.49 | fugitivo | JMcA: it's a nice service, is that a private company or government? |
| 13:16.13 | JMcA | we're private, but its contracted from state and local governments almost completely |
| 13:16.22 | JMcA | no cost to the end user |
| 13:16.26 | astr | Another thing about E911, do we have to support E911 if we are just providing PSTN minutes and no PSTN->our service? |
| 13:16.27 | fugitivo | great |
| 13:16.58 | *** join/#asterisk mobmob (n=mobmob@195.176.254.254) |
| 13:17.03 | JMcA | fugitivo: we've got statewide service in 19 states, plus a lot of others in other states...there's a reasonable chance the service is available in your area |
| 13:17.13 | h3x | astr: according to the FCC you have to provide it no matter what if its a "pc to phone" service |
| 13:17.15 | mistral | whats really funny about the E911 is that most pbx's if you kill the power to them they die |
| 13:17.16 | fugitivo | i'm not in the us :) |
| 13:17.17 | *** part/#asterisk secure75 (n=mic@host-82-135-62-14.customer.m-online.net) |
| 13:17.22 | JMcA | ah |
| 13:17.23 | DarkFlibble | brb - hitting shops for drink! |
| 13:17.33 | h3x | mishehu: the power dosent always go out when you need 911 heh |
| 13:17.38 | JMcA | then there's a very slim chance its available in your area :) |
| 13:17.42 | fugitivo | and thanks god i'm not a victim of any kind, hehe |
| 13:17.52 | mistral | h3x: i know |
| 13:18.00 | mistral | but i am uk and they are thinking of bring it in here |
| 13:18.15 | h3x | whats e911 have to do with the UK |
| 13:18.17 | h3x | isnt it 112 there |
| 13:18.26 | DarkFlibble | 999 and 112 in the UK |
| 13:18.26 | JMcA | absolutely...interestingly, we get a lot of non victims registering for notifications...family of offenders...police officers that arrested the offenders...all kinds of stuff like that beyond what the original vision of the service was |
| 13:19.11 | fugitivo | JMcA: i'm wondering what does the people do when they get the notification from your company |
| 13:20.34 | JMcA | most will not do much different other than just be aware and wary and watchful for a little while...some will make arrangements to not be at home for a couple of days...stay with friends or family for a couple of days to be harder to find |
| 13:20.36 | astr | h3x: I was reading the FCC doc and they say that we have to provide the E911 only if we are competitive to replace the regular POTS. we are simply providing PSTN minutes and not providing DIDs |
| 13:21.03 | h3x | where does it say taht |
| 13:21.50 | fugitivo | JMcA: does your company has a study on people's reaction? i'd be interesting |
| 13:21.52 | astr | http://hraunfoss.fcc.gov/edocs_public/attachmatch/FCC-05-116A1.pdf |
| 13:22.10 | h3x | maybe they did that coz i bitched at them about it |
| 13:22.11 | h3x | heh |
| 13:22.27 | astr | hsx: Also, I know a couple of companies which are big VoIP providers but they verify that we already have a POTS line before they give service |
| 13:22.29 | *** join/#asterisk razu (n=razu@80-235-90-19-dsl.prn.estpak.ee) |
| 13:22.45 | *** part/#asterisk mobmob (n=mobmob@195.176.254.254) |
| 13:23.25 | JMcA | fugitivo: nothing formal...we're really a fairly small'ish company and don't really have the spare revenue to do something like that....we do try to build strong connections with police departments, sheriffs, victims advocates, prosecutor's offices, and the like, so we have, informally, a pretty decent feel for the situations that people face |
| 13:23.35 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 13:23.40 | astr | There is also an In and Out service for some providers like SKype. If both CAN be used, then yes - you will have to provide but when used seperately |
| 13:23.45 | jaike | 1.2.2 released? yey! |
| 13:24.02 | h3x | astr: that thing is 91 pages long |
| 13:24.03 | h3x | where |
| 13:24.05 | h3x | what page |
| 13:24.49 | astr | hsx :) |
| 13:24.56 | astr | hsx: let me look |
| 13:25.02 | JMcA | ok...off to work |
| 13:25.51 | astr | h3x: http://www.techlawjournal.com/topstories/2005/20050603.asp - this is nother link - look for jeff pulver |
| 13:25.55 | jaike | i guess the voicemail bug is fixed in 1.2.2 |
| 13:27.21 | h3x | uh |
| 13:27.31 | h3x | the purpose of the skype comments is that skype as a business isnt in the US |
| 13:27.42 | h3x | well now it is |
| 13:27.51 | h3x | but they are talking about people getting US numbers in another country |
| 13:28.24 | *** join/#asterisk CALLER (n=Camisa@c-67-186-94-173.hsd1.in.comcast.net) |
| 13:31.28 | astr | h3x: you say that we will have to provide E911 even if we provide only PSTN minutes, no DIDs |
| 13:32.34 | h3x | well thats what i remember the fcc ruling saying |
| 13:32.34 | h3x | because they are retards |
| 13:32.41 | astr | h3x: page 12 scope |
| 13:32.42 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:33.26 | astr | read 23, 24 |
| 13:33.58 | *** join/#asterisk luis_ (n=luis@87.223.225.60) |
| 13:34.26 | h3x | alright well im glad they addressed that |
| 13:34.31 | h3x | but this document isnt the final rule |
| 13:34.32 | h3x | its just comments |
| 13:34.49 | astr | hsx: also look at teh stanaphone services about E911 - they split the services into IN and OUT et al. |
| 13:34.57 | *** join/#asterisk mountie (n=mountie@CPEdeaddeaddead-CM000a739acaa4.cpe.net.cable.rogers.com) |
| 13:35.13 | astr | Is there any good 'reliable' provider of minutes in US? |
| 13:35.34 | h3x | what kind of minutes |
| 13:35.35 | h3x | heh |
| 13:35.38 | cypromis | god |
| 13:35.39 | cypromis | :P |
| 13:35.40 | astr | PSTN |
| 13:35.47 | [TK]D-Fender | Yay, GnomeMeeting's next major beta is out for *nix! Just waiting for the Win32 version...... http://www.ekiga.org/ |
| 13:35.49 | h3x | pstn is everything that has to do with a phone |
| 13:35.58 | h3x | you mean 1+ termination? |
| 13:36.06 | astr | h3x: correct termination |
| 13:36.10 | [TK]D-Fender | SIP/H.323 audio/video w/ all the trimmings and free! |
| 13:36.14 | h3x | i own carrierone.net |
| 13:36.14 | h3x | heh |
| 13:36.45 | astr | :) |
| 13:37.09 | astr | how many mins does carrierone provide on a monthly basis? |
| 13:37.14 | h3x | ive got some customers using 2000+ channels of voip termination in my colo |
| 13:37.15 | DarkFlibble | I think a large proposion of this channel owns some kind of telecoms company... |
| 13:37.35 | h3x | 45 million |
| 13:37.36 | tzanger | DarkFlibble: I don't think so |
| 13:37.47 | tzanger | DarkFlibble: some of them do, that is certain, but a large proportion? nah |
| 13:38.37 | astr | h3x: compatible with SER + Asterix setup? |
| 13:38.38 | h3x | btw thats 2000 channels of g.711 fax |
| 13:38.46 | h3x | so if fax works i guess you would say its pretty damn reliable |
| 13:38.57 | h3x | yeah we use ser |
| 13:39.19 | astr | do you support GSM codec? :) |
| 13:39.41 | h3x | basically we have a few private ip connections to major carriers, and use ser as a b2bua so your box connects RTP directly to the underlying carrier |
| 13:39.41 | h3x | no |
| 13:39.45 | h3x | g.729 and g.711 |
| 13:39.54 | h3x | coz thats what all the carriers take |
| 13:40.18 | h3x | they have $2 Million voip gateways that work a lot better than if i used a bunch of asterisk boxes or max tnt's |
| 13:40.49 | h3x | the echo cans etc are really good |
| 13:41.06 | astr | is it built in echo cans? |
| 13:41.23 | h3x | most of the carriers are using Sonus or Telica switches |
| 13:41.43 | h3x | There are two carriers with shitty ass voip that i refuse to use such as Global Crossing and XO |
| 13:41.47 | h3x | so i would use TDM with them |
| 13:41.57 | h3x | and run that through a Max TNT |
| 13:42.11 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:42.13 | h3x | GX and XO use Sonus but however they implemented it sucks |
| 13:42.34 | h3x | Qwest uses Sonus but their call quality is awesome |
| 13:42.55 | *** join/#asterisk Defraz (n=t0tal@24-119-94-19.cpe.cableone.net) |
| 13:43.19 | astr | how do I get rate information of your offerings? I did not find on your website. We were contemplating commpartners as they look good and have billing supported |
| 13:43.20 | h3x | maybe its just their proxy |
| 13:43.37 | h3x | commpartners HQ is across the street from me |
| 13:43.51 | h3x | we're getting an optical cross connect up and going to exchange some traffic |
| 13:44.08 | h3x | the difference is, they are in a colo facility that they arent allowed to sublease |
| 13:44.11 | h3x | they send customers to me sometimes |
| 13:44.20 | h3x | i own my datacenter |
| 13:44.20 | jaike | h3x: which company you with? whats your site? |
| 13:44.25 | h3x | www.carrierone.net |
| 13:44.40 | *** join/#asterisk coppice (n=chatzill@93.155.17.210.dyn.pacific.net.hk) |
| 13:44.52 | h3x | we don't advertise rates because its all wholesale and everybodys got different needs |
| 13:45.12 | h3x | and the lower rate plans are volatile rates |
| 13:45.18 | astr | h3x: do you mind if I PM you? |
| 13:45.20 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 13:45.23 | h3x | thats fine |
| 13:45.43 | Peggerr | h3x where is the center located |
| 13:45.46 | h3x | basically what we're doing that others dont do is least cost route on OCN and LATA |
| 13:46.00 | h3x | we dip into SS7 databases and grab the actual carrier even if the numbers ported |
| 13:46.14 | h3x | big iron switches can do that but they dont pass the savings on to customers coz their billing and costs arent associated |
| 13:46.16 | h3x | las vegas |
| 13:46.38 | astr | h3x: do you have billing built for your resellers? |
| 13:47.02 | h3x | on our thin margin rate plans we make the customer pay for the lidb query, but the data is cached for a few days |
| 13:47.16 | h3x | not yet, maybe someday soon |
| 13:48.15 | *** join/#asterisk PoWeRKiLL (n=PoWeRKiL@84.205.154.241) |
| 13:48.45 | h3x | if you have something you want to do with commpartners instead, you can colo here |
| 13:49.03 | h3x | but if you use our services then the colo costs are cheap/free |
| 13:49.39 | h3x | the luxury of paying 0.50 a square foot instead of $30 a square foot like CP is doing :D |
| 13:49.41 | astr | h3x: we are more interested in a reliable service provider for the kind of business we do and my research proved them to be stable and cash ful :) |
| 13:49.59 | jaike | were doing business with commpartners too |
| 13:50.11 | jaike | we were with txlink, commpartners bought txlink |
| 13:50.22 | astr | <PROTECTED> |
| 13:50.31 | jaike | yup..pretty reliable |
| 13:50.40 | h3x | the one big problem i have with CP is they dont manage their channel limits very well |
| 13:50.53 | h3x | they will oversell channels of capacity |
| 13:51.00 | jaike | but they only support sip... |
| 13:51.01 | jaike | no iax |
| 13:51.03 | h3x | they were down for about 15 mins the other day coz some dialer company used all their ports |
| 13:51.15 | h3x | during a peak time of the day |
| 13:51.43 | coppice | the airlines oversell seats, and it never does them any ha..... oh, they're all in chapter 11, aren't they :-) |
| 13:51.46 | h3x | the main reason we are starting to use them for our stuff is just because we have some customers that may be 'fly by night' and i dont feel like ordering gobs of capacity from a major ixc |
| 13:51.50 | h3x | and then having to cancel orders |
| 13:52.14 | astr | jaike: we have SER setup with Asterisk so that we can route POTS call to Asterisk. Can we hook CP with SIP.conf in asterisk? |
| 13:52.21 | h3x | their OCN list also sucks wang |
| 13:52.30 | h3x | its a carbon copy of the global crossing OCN list |
| 13:52.39 | h3x | they only consider like 26 OCNs to be their lowest cost tier |
| 13:52.39 | jaike | astr: yup...were doing asterisk-sip with them |
| 13:52.45 | h3x | whereas qwest considers 78 LECs to be |
| 13:52.53 | h3x | so commpartners does a bill and keep |
| 13:53.06 | h3x | they charge you for a high rate tier when they are paying for a low cost one |
| 13:53.36 | jaike | astr: better do more research...but so far theyre ok |
| 13:53.50 | astr | hsx: yes, they do. But hte prices were around 0.00[8-10]c which was already low |
| 13:54.02 | h3x | not their flat rates |
| 13:54.08 | h3x | thats tier A |
| 13:54.12 | jaike | h3x: had a hard time reading their rates |
| 13:54.40 | h3x | at the end of the day their rates are gonna cost about the same or more than voipjet |
| 13:54.48 | astr | jaike: any places where I can research. My customers aall use GSM and if I get GSM provider, that will be excellent but I understand GSM is hard and I am ok with transcoding |
| 13:55.10 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 13:55.22 | astr | jaike: you are right. They have 4-5 rates and I did not understand anything. Any guidance would be appreciated. Why do we have 4 rates. RBOC 1 etc etc. |
| 13:55.23 | mutilator | ugh |
| 13:55.31 | mutilator | customers annoy me |
| 13:55.31 | h3x | commpartners is big though |
| 13:55.34 | mutilator | [07:46:32] <mdgraham-M33Access> we just got hooked up to DSL yesterday and now my McAfee Virus Scan has been running the whole time and I can't even restart or turn off my computer |
| 13:55.38 | jaike | they only do ulaw and g729 |
| 13:55.46 | mutilator | [07:48:05] <mdgraham-M33Access> I can't cancel it...it all started after we got hooked up |
| 13:55.49 | jaike | were using g729 with them to save bandwidth |
| 13:55.51 | h3x | they are so big that they got wiltel to construct fiber across the street to the colo building they are in |
| 13:56.01 | *** mode/#asterisk [+o drumkilla] by ChanServ |
| 13:56.09 | h3x | Yeah, comm partners uses telica switches |
| 13:56.11 | pimpwell | anyone from NY here? |
| 13:56.13 | h3x | g.729 baby |
| 13:56.31 | astr | jaike: yes, we are going to use ulaw or G729? any idea aboutGSM to 729 transcoding? I think it will be an overkill |
| 13:56.49 | *** join/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 13:56.53 | h3x | astr: what is your application |
| 13:56.55 | *** join/#asterisk oej (n=oej@199.227.185.35) |
| 13:56.59 | jaike | havent dont g729 - gsm |
| 13:57.31 | Katty | hi lads. |
| 13:57.37 | h3x | btw we're debt free. I paid for it all with cash out of pocket |
| 13:57.49 | astr | h3x: cannot reveal more. something to do with voip on small devices |
| 13:57.55 | h3x | i didnt spend many millions on it but its all here and its mine |
| 13:58.12 | h3x | your small devices dont support g.729? |
| 13:58.17 | h3x | Are you making a dick tracy voip watch? :D |
| 13:58.28 | Peggerr | when you buy a t1 line what kind of voice plans are ushilly on them? local, national? |
| 13:58.41 | astr | h3x: no - that will be OVERKILL for those devices |
| 13:58.49 | zoa | h3x what are you making ? |
| 13:58.58 | h3x | making? |
| 13:59.01 | astr | jaike: are you using any billing software from CP? |
| 13:59.11 | astr | h3x: lol |
| 13:59.24 | h3x | $ wise? i cant say that coz then everybody would figure out how much it cost me for minutes |
| 13:59.24 | h3x | haha |
| 14:00.04 | Peggerr | anyone purchased t1 lines before? |
| 14:00.32 | h3x | Peggerr: you can get t1s from a local equipment carrier (LEC) or inter exchange carrier (IXC) |
| 14:00.53 | *** join/#asterisk Ariel_ (n=Ariel@adsl-068-157-125-248.sip.mia.bellsouth.net) |
| 14:00.57 | Peggerr | h3x, what does tls stand for |
| 14:01.00 | h3x | LECs give you local service and expensive long distance coz its handed off to another carrier and they share revenue |
| 14:01.27 | jaike | t1s |
| 14:01.30 | h3x | or you can get dedicated long distance but if you called local it would be a intrastate long distance call |
| 14:01.34 | h3x | T1 not TL |
| 14:01.44 | *** join/#asterisk rculp (n=rculp@66.173.240.20) |
| 14:01.55 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 14:01.56 | astr | jaike, h3x: what is RBOC , CLEX , GTE and NECA mean. I only understood Sprint in their pricing structure |
| 14:02.26 | DarkFlibble | astr, you might want to buy youirself a nice Telecoms dictionary |
| 14:02.28 | h3x | RBOC = Bell South, SBC (not the AT&T part), Qwest (US West), and Verizon |
| 14:02.50 | astr | dark : yes - I should. Thanks H3x |
| 14:02.50 | h3x | GTE is the GTE portion of Verizon |
| 14:02.59 | h3x | CLEC are competitive local carriers |
| 14:03.02 | h3x | NECA is like, |
| 14:03.16 | h3x | the gougers of ILECs like rural telcos, indian reservations, etc. |
| 14:03.25 | jaike | astr: never got to understand them..we were txlink's client so we decided to continue with the old rate |
| 14:03.29 | jaike | flat rate |
| 14:03.43 | h3x | but their fancy titles that CP uses |
| 14:03.50 | h3x | isnt what those columns really are |
| 14:03.53 | jaike | were only doing 150,000+ mins per month |
| 14:03.55 | h3x | you have to ask for a OCN list |
| 14:04.07 | astr | jaike: flate rate for anywhere in US |
| 14:04.11 | h3x | of 4 character alphanumeric codes that defines what each rate tier is |
| 14:04.13 | jaike | yup |
| 14:04.25 | astr | jaike: how are they for international termination? |
| 14:04.36 | jaike | went dont do intl |
| 14:05.03 | *** part/#asterisk flok420 (i=nobody@keetweej.xs4all.nl) |
| 14:05.36 | h3x | so heres the deal |
| 14:05.40 | Peggerr | yaha is it possible to get a t1 with a flate rate anywhere in the us? |
| 14:05.53 | h3x | thanks to the FCC you can port numbers from anything to anything practically if its in the same rate cente |
| 14:05.53 | h3x | r |
| 14:06.03 | *** join/#asterisk edwin_ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 14:06.20 | h3x | Peggerr: there are some companies selling that, but since they are paying for it as minutes they make the price somewhat unattractive |
| 14:06.25 | *** part/#asterisk cpm (n=Chip@pdpc/supporter/sustaining/cpm) |
| 14:07.06 | *** join/#asterisk fuzza (n=andrew@ppp171-157.lns1.per1.internode.on.net) |
| 14:07.12 | fuzza | hi all |
| 14:07.31 | h3x | ive heard of $1200 + loop all you can eat t1s |
| 14:07.35 | h3x | but if you work the math on it |
| 14:07.46 | Peggerr | ok so you buy a t1 and you get two diffrent providers a local and a log distance, how do you make sure each call goes to the proper provider |
| 14:07.46 | h3x | its really hard to make that cheaper than per minute |
| 14:08.14 | astr | jaike: do you have their billing or yours? |
| 14:08.56 | fuzza | I've got a 1.0.2 install (yes I know it's old but upgrading isn't a high priority) with chan_capi and two Fritz cards on two BRIs in Australia. all works fine, except for incoming callerID... Telstra are "absolutely certain" that it's turned on, but no matter what settings I try (of which there don't seem many) it's always empty... any ideas? |
| 14:09.24 | tzanger | fuzza: use pri debug (I think that will work with BRI as well) and verify the q.931 signaling is getting caller ID |
| 14:09.51 | astr | jaike, h3x: another last question: what is this 10000.100013 10000.10015 etc. I have 4 xls with different rates. Are they peak and off peak? |
| 14:10.01 | fuzza | tzanger: there's capi debug, is that likely what you mean? |
| 14:10.44 | h3x | eh |
| 14:12.18 | *** join/#asterisk NewSole (n=dave@d38-53-48.commercial1.cgocable.net) |
| 14:12.19 | jbalcomb | [TK]D-Fender iCEBrkr you guys wanna make sweet IRC based Asterisk love? |
| 14:12.53 | fugitivo | fuzza: you'll not get much help if you don't upgrade |
| 14:13.48 | jbalcomb | fuzza: its true that you should upgrade |
| 14:14.00 | fuzza | fugitivo: true I guess... |
| 14:14.04 | *** join/#asterisk basta (n=basta@194.150.162.129) |
| 14:14.09 | jbalcomb | fuzza: you have the callerid settings turned on in zapata.conf? |
| 14:14.17 | fugitivo | trying to debug something old is a waste of time sometimes |
| 14:14.28 | basta | apart from nat, what can be other causes for one way audio ? (sip) |
| 14:14.43 | fuzza | jbalcomb: looks like it: usercallerid=yes |
| 14:14.44 | jbalcomb | fuzza: you have maybe a NoOp($CALLERID) or something so you can see the value you of it early in your calling plan? |
| 14:15.06 | fuzza | jbalcomb: yep, empty (also can see with show channels) |
| 14:15.08 | NewSole | your RTP ports not open basta> |
| 14:15.29 | *** join/#asterisk graab1 (n=gunnar@bkkb-gw.bitcon.no) |
| 14:15.33 | jaike | astr: youll have to ask a commpartners rep to explain that. gave me a headache trying to understand their spreadsheet |
| 14:15.36 | fuzza | jbalcomb: having said that, re zapata.conf, is there a particular module(s) that uses it? because I may have disabled them |
| 14:15.42 | jbalcomb | fuzza: have you confirmed with your telco that its available? |
| 14:15.45 | basta | mh, I'll take a look, thanks |
| 14:16.32 | fuzza | jbalcomb: they're "absolutely certain" (however sure that actually makes them...), and I was at a near-identical install today which worked fine |
| 14:16.33 | jbalcomb | fuzza func_callerid.so might be neccessary |
| 14:16.34 | jaike | basta: make sure your allowng UDP 10000-20000 |
| 14:17.08 | NewSole | and check on your iptables if you have it installed to allow it there too |
| 14:17.24 | fuzza | jbalcomb: don't seem to have that one, might it be a 1.2 feature? |
| 14:17.44 | fuzza | jbalcomb: I have app_setcallerid.so (which I already use) but that's obviously the other way |
| 14:17.57 | jbalcomb | fuzza how about app_setcallerid.so? |
| 14:18.05 | jbalcomb | fuzza ok |
| 14:18.53 | jbalcomb | fuzza i dont know that func_callerid.so is 1.2.x; you might check the wiki to be sure |
| 14:19.19 | fuzza | jbalcomb: trying to find it :-/ there's a reference in Slimming but that's about it |
| 14:19.40 | basta | in rtp.conf I'm allowing 5000-31000 now, anyway is the called who can't hear me ... |
| 14:19.55 | [TK]D-Fender | basta : Behind NAT? |
| 14:20.22 | jbalcomb | [TK]D-Fender how does this look? http://pastebin.com/514514 |
| 14:20.26 | basta | he says he isn't, seems is working with a cisco peer |
| 14:20.32 | *** join/#asterisk redax (n=redax@r6.hu) |
| 14:20.41 | basta | it a quescom gateway |
| 14:20.57 | redax | hi! |
| 14:21.01 | jbalcomb | fuzza that is certainly odd. upgrade seems like a reasonable step towards resolution at this point. (sorry for the MS answer) |
| 14:21.17 | redax | is bristuff-0.3.0-PRE-1g working with asterisk-1.2.2 ? |
| 14:21.33 | fuzza | jbalcomb: heh... hm, I'll have to see if the boss (contractor) will pay me to do it :-( |
| 14:21.36 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 14:21.36 | *** mode/#asterisk [+o denon] by ChanServ |
| 14:22.29 | [TK]D-Fender | jbalcomb : 2 x PRI? I wouldn't bother using differnt context's for the inbound calls (they are DID's anyways) and those gain settings are SCARY |
| 14:22.48 | [TK]D-Fender | basta : So you have a public IP on your box? |
| 14:22.58 | fuzza | jbalcomb: so _should_ any of the zap* stuff be used with isdn/capi? I seem to have them all commented out in modules.conf (and no autoload); I thought they were more for analog lines |
| 14:23.12 | fuzza | (ztdummy for timing if needed, but apart from that) |
| 14:25.29 | jbalcomb | [TK]D-Fender the 2 PRIs are for different companies |
| 14:26.16 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 14:27.27 | [TK]D-Fender | jbalcomb : But once it goes into your box, who cares? Just seperate by DID in 1 big pool..... and comment it. |
| 14:28.20 | jbalcomb | fuzza: I am not sure about that really, sorry. |
| 14:28.26 | [TK]D-Fender | Your customers dial DID's, who cares which PRI they land on? That means PRI#1 could fill up and then start rejecting calls. Balance the 2 of them and things work out better... |
| 14:30.14 | basta | TK. yes my asterisk is public and his quescom is public |
| 14:31.24 | basta | if I call from a lynksys pap connected to my box which routes to the quiscom it works (just tried) |
| 14:31.31 | [TK]D-Fender | basta : So 5060 is going through as well as 10000-20000? |
| 14:31.48 | *** join/#asterisk ckruetze (n=ckruetze@131.8.dsl3.ip.foni.net) |
| 14:31.49 | caio1982 | tzafrir_laptop: hey tzafrir, could you take a look at this bug? http://bugs.debian.org/cgi-bin/bugreport.cgi?bug=348090 |
| 14:32.07 | groogs | jbalcomb: yeah i agree with [TK]D-Fender.. though, it may be a good idea to use Set/CheckGroup etc to limit the number of calls per company, so one company can't hog all of the lines and there's always some availalbe for the other co |
| 14:32.10 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 14:32.12 | caio1982 | tzafrir_laptop: it's pretty simple to fix, and i noticed that just now when trying to build smsq too |
| 14:32.19 | jhiver | hi all |
| 14:32.23 | basta | TK, yes, the problem is with asterisk client ->> asterisk server ->> quescom, I'm connecting to the client to take a look |
| 14:32.45 | jhiver | I'm having a little issue with asterisk and a dialplan, can somebody with a charitable mind help me out? |
| 14:33.01 | ckruetze | Hi, could somebody tell me what is wrong with this line: exten => s,2, Dial(IAX2/klaus&IAX2/heidi,40,t,r) ? |
| 14:33.22 | jhiver | The space between the , and the Dial? |
| 14:33.29 | jbalcomb | [TK]D-Fender well, the 2nd PRI actually belongs to a client so I don't know that we are allowed to do that |
| 14:33.43 | kaldemar | ckruetze: the comma between t and r. |
| 14:33.45 | [TK]D-Fender | groogs : Not entirely agreeing with that. A channel is a channel and with Caller ID forging should be treated as such. |
| 14:33.58 | ckruetze | kaldemar: thanks |
| 14:34.27 | ckruetze | jhiver: No, spaces are ok |
| 14:34.28 | [TK]D-Fender | jbalcomb : Grey area. If they own it may take some arranging. Thats politics, not tech :) |
| 14:36.42 | jhiver | can anybody tell me what's wrong with this? |
| 14:36.44 | jhiver | [world] |
| 14:36.44 | jhiver | include => special |
| 14:36.44 | jhiver | _0692X. => s,1,Dial(Zap/3/0692${EXTEN:4}) |
| 14:36.45 | jhiver | _0692X. => s,2,Dial(SIP/00262692${EXTEN:4}@finalcut-out) |
| 14:36.45 | jhiver | _0692X. => s,3,Dial(IAX2/543@voipjet/011262692${EXTEN:4}) |
| 14:37.02 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 14:37.13 | jhiver | when I use this, I can't dial out to 0692XXXXXX anymore |
| 14:37.20 | jhiver | while with my old dialplan I could |
| 14:37.36 | *** join/#asterisk gvag11 (n=g@ipa95.5.tellas.gr) |
| 14:37.41 | gvag11 | hi all |
| 14:37.42 | *** join/#asterisk JMcA (n=jmcadams@pixout.appriss.com) |
| 14:38.20 | gvag11 | I am trying to find the right TIFF format for trasmission with tx_fax... Any idea ? |
| 14:38.58 | [TK]D-Fender | jhiver : very wrong. |
| 14:39.12 | jhiver | ? |
| 14:39.18 | jhiver | please let me know :) |
| 14:39.22 | [TK]D-Fender | jhiver: exten => _0692X.,1,Dial(Zap/3/0692${EXTEN:4}) |
| 14:39.26 | [TK]D-Fender | exten!!!! |
| 14:39.33 | jhiver | ah :) |
| 14:39.37 | jhiver | duuuuh :) |
| 14:39.50 | [TK]D-Fender | and fix the rest accordingly. |
| 14:40.03 | jhiver | cheers |
| 14:40.15 | [TK]D-Fender | jhiver : And why such a large prefix? |
| 14:40.22 | jhiver | this lcr stuff has gotten my mind blown up so I forgot the simplest things :) |
| 14:40.27 | jhiver | how do you mean? |
| 14:41.10 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 14:41.19 | [TK]D-Fender | 0692x as a prefix to dial out.... most peole jsut use "9" or something like it. |
| 14:41.48 | gvag11 | I am trying to find the right TIFF format for trasmission with tx_fax... Any idea ? Because with TIFF files generated by SPANDSP its ok, all the others fails.... |
| 14:42.42 | jhiver | oh |
| 14:42.49 | jhiver | it's part of a much bigger dialplan |
| 14:43.00 | *** join/#asterisk mog_work (n=mogorman@gateway.digium.com) |
| 14:43.11 | jhiver | 0692, by french standards, is mobile reunion island numbers |
| 14:43.16 | [TK]D-Fender | jhiver : I can only imagine why you'd neet something that large.... |
| 14:43.33 | jhiver | which is why it becomes 011262692 when dialed through VoIPJet |
| 14:43.34 | [TK]D-Fender | need* |
| 14:44.10 | jhiver | I am building an 'optimized' dialplan and doing LCR shit and trying now to see if it works |
| 14:44.28 | jhiver | the global optimized dialplan will be around 12k lines (!) |
| 14:45.09 | [TK]D-Fender | jhiver : How many different providers / area codes? |
| 14:46.16 | jhiver | well at the moment |
| 14:46.32 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 14:46.32 | jhiver | voipjet, phonext, nufone, plus a couple of area codes through Zap |
| 14:47.26 | jhiver | area codes... let me see |
| 14:47.26 | [TK]D-Fender | jhiver : Then it shouldn't be such a huge dia-plan unless you're being "messy" about it. |
| 14:47.29 | jhiver | lots :) |
| 14:47.36 | jhiver | oh yeah? |
| 14:47.38 | jhiver | how come? |
| 14:47.48 | jhiver | every provider lists tons and tons of prefixes |
| 14:48.09 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 14:48.38 | caio1982 | jhiver: what you mean by tons and tons? :) 12k lines man |
| 14:48.44 | jhiver | voipjet has 3000 of them |
| 14:48.52 | [TK]D-Fender | jhiver : You need a small diaplan and a small database to select the provider. |
| 14:48.55 | *** join/#asterisk lorinc (n=ang@caracas-3585.adsl.interware.hu) |
| 14:49.10 | caio1982 | couldn't you just make generic patterns to match some in just one hit? |
| 14:49.11 | jhiver | no that sucks because then you need to use an external app... blergh |
| 14:49.22 | [TK]D-Fender | jhiver : External app? Says who? |
| 14:49.25 | jhiver | Let me test if this works :) |
| 14:49.30 | jhiver | says me |
| 14:49.51 | [TK]D-Fender | jhiver : And that why you're doomed :) LEave LCR conceptualizing to us :) |
| 14:50.31 | jhiver | lol |
| 14:50.40 | jhiver | how would you do it then |
| 14:50.52 | [TK]D-Fender | You could do it all in ASTDB if you wanted. I'd suggest a small AGI with a database like SQLite personally. |
| 14:50.56 | jhiver | at first I had an AGI produce the proper dialstring |
| 14:51.18 | jhiver | but then I though "a separate process launched for each call, sucks" |
| 14:51.30 | [TK]D-Fender | jhiver : What kind of call volume? |
| 14:51.43 | jhiver | so if you don't want to use AGI (= external app.) it has to be static |
| 14:52.07 | jhiver | at the moment, I don't route that much |
| 14:52.12 | jhiver | maybe 20kminutes / day |
| 14:52.16 | [TK]D-Fender | jhiver : Give me a number :) |
| 14:52.20 | [TK]D-Fender | of CALLS. |
| 14:52.25 | jhiver | hang on |
| 14:53.17 | jhiver | ok for yesterday that was 6558 calls |
| 14:53.23 | *** join/#asterisk RoyK (n=roy@10.80-203-106.nextgentel.com) |
| 14:55.12 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 14:55.14 | X-Files | hey all! Microsoft Windows Messenger 5.1 (not MSN) work fully VIDEO and Contact list STREAM ???? |
| 14:55.55 | jhiver | These are the longest prefixes: |
| 14:55.57 | jhiver | [world] |
| 14:55.57 | jhiver | include => special |
| 14:55.58 | jhiver | exten => _00180954291X.,1,Dial(IAX2/jhiver@NuFone/180954291${EXTEN:11}) |
| 14:55.58 | jhiver | exten => _00180954291X.,2,Dial(IAX2/543@voipjet/180954291${EXTEN:11}) |
| 14:56.02 | jhiver | looking any better? |
| 14:57.09 | redax | strange, |
| 14:57.31 | jhiver | hey it looks like Asterisk is adding all those 12k extensions now, cool :) |
| 14:57.41 | redax | asterisk 1.2.1+bristuff0.3.0-pre-1g having 2 zaphfc cards, 1TE and 1NT |
| 14:58.09 | redax | calling local numbers working (ie 6digits), but calling longdistance not working |
| 14:58.30 | caio1982 | jhiver: [12:50:45] <caio1982> couldn't you just make generic patterns to match some in just one hit? |
| 14:58.36 | redax | always getting this app_dial.c: Unable to forward voice |
| 14:58.37 | jhiver | hey that worked :) |
| 14:58.43 | jhiver | well, the way I do it |
| 14:58.52 | jhiver | first I make a list of all prefixes |
| 14:58.55 | *** join/#asterisk Cyon (n=cyon@cyons.net) |
| 14:59.08 | jhiver | then for each separate prefix I make a list of providers from cheapest to more expensive |
| 14:59.20 | jhiver | then I do a bunch of dials() |
| 14:59.36 | jhiver | and I start with longest prefixes first and end with shortest prefixes |
| 14:59.47 | jhiver | so I already do pattern mach, that why there is an X. |
| 15:00.01 | jhiver | I really can't see any other proper way to do it |
| 15:00.18 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-179-75-202.cust.bezeqint.net) |
| 15:00.36 | jhiver | so for example for reunion island '0262' it produces this |
| 15:00.45 | jhiver | exten => _0262X.,1,Dial(Zap/3/0262${EXTEN:4}) |
| 15:00.46 | jhiver | exten => _0262X.,2,Dial(IAX2/543@voipjet/011262262${EXTEN:4}) |
| 15:00.46 | jhiver | exten => _0262X.,3,Dial(SIP/00262262${EXTEN:4}@finalcut-out) |
| 15:00.59 | jhiver | but for france proper '0' it produces this |
| 15:01.18 | Cyon | Has anyone experienced asterisk becoming a zombie process? I can't kill it, can't hup it, can't connect to the cli...but it's still got a hold on the ports... |
| 15:01.23 | jhiver | oh, too long to paste in IRC channel :) |
| 15:01.23 | caio1982 | life turns easier with AGIs or macros |
| 15:01.38 | jhiver | but then this isn't too bad |
| 15:01.51 | jhiver | the massive dialplan is produced automatically, then I just include it |
| 15:02.18 | X-Files | hey all! Microsoft Windows Messenger 5.1 (not MSN) work fully VIDEO and Contact list STREAM ???? |
| 15:02.37 | Katty | hi lovables. |
| 15:02.42 | jhiver | I know it _looks_ ugly but it's accurate... |
| 15:02.45 | zoa | hey ho darling |
| 15:02.47 | zoa | :p |
| 15:03.20 | jhiver | plus it's not really the script's fault if providers list zillions of prefixes... |
| 15:03.46 | jhiver | if you want a simpler dialplan, you simply need to trim down the number of prefixes but then there is no point in doing LCR stuff either |
| 15:04.44 | Katty | mister fender! |
| 15:05.15 | tzanger | good morning katty my dear |
| 15:06.50 | X-Files | MrChimpy: ? |
| 15:07.08 | MrChimpy | was that a sentence or just a stream of random words? |
| 15:07.24 | tzanger | MrChimpy: hahahaha |
| 15:07.45 | jhiver | ok lads, thanks for the help! |
| 15:08.10 | jhiver | [TK]D-Fender, thanks for your input |
| 15:08.12 | jhiver | cya all |
| 15:09.18 | gvag11 | I am trying to find the right TIFF format for trasmission with tx_fax... Any idea ? Because with TIFF files generated by SPANDSP its ok, all the others fails.... |
| 15:10.48 | ManxPower | gvag11, I can put a Perl script on pastebin that converts files to the correct format for transmitting. The script is not fully working, but the file conversion part is. |
| 15:10.52 | coppice | gvag11: they need to be 1728 pixels wide |
| 15:11.28 | gvag11 | ok ... thanks guys ... |
| 15:11.36 | gvag11 | Manxpower can i have the script ... thanks |
| 15:12.08 | gvag11 | coppice thats the only really prerequisite ? I mean no Group3 -2d compression ? |
| 15:12.43 | coppice | well they need to be in FAX format |
| 15:13.04 | Cyon | WOw that was really ugly |
| 15:14.19 | gvag11 | coppice : i use gs to turn a pdf to tiffg32d (or tiffg3) , papersize a4 and it doesn't work.... So i should set the wide to 1728, right ? |
| 15:15.14 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 15:15.40 | ManxPower | gvag11, www.fnords.org/~eric/asterisk/email2fax.pl |
| 15:16.19 | gvag11 | manxpower thanks |
| 15:16.36 | ManxPower | gvag11, don't ask me any questions about it. 8-) |
| 15:16.41 | [TK]D-Fender | Katty: mew,. |
| 15:16.42 | gvag11 | ok.... |
| 15:16.44 | [TK]D-Fender | (was AFK) |
| 15:17.15 | [TK]D-Fender | And about to be again :) |
| 15:18.16 | ManxPower | <PROTECTED> |
| 15:18.40 | Katty | the fire marshall is here! |
| 15:19.21 | gvag11 | manxpower : i am using the same but PAPERSIZE=A4... I will try with letter ... |
| 15:19.50 | ManxPower | gvag11, I'm in the USA, so letter is what is expected. |
| 15:20.13 | *** join/#asterisk Lathos42 (n=Lathos42@adsl-69-210-24-249.dsl.lgtpmi.ameritech.net) |
| 15:20.19 | ManxPower | The script has several...issues.. Such as what if you want to send a legal size document (which is what many of my users want)? |
| 15:20.44 | gvag11 | manxpower: i will take a closer look i think ... |
| 15:21.18 | coppice | ManxPower: I just said. the legal size is 1728 pixels wide. :-) |
| 15:21.38 | ManxPower | coppice, I'm still on my 1st cup of coffee. |
| 15:22.07 | iCEBrkr | :D |
| 15:22.26 | JMcA | ManxPower: doncha just hate it when you have too much blood in your caffeine stream? |
| 15:22.32 | ManxPower | *sigh* I'm starting various projects and am starting to realize just how much equipment I lost in Katrina. 8-( |
| 15:22.44 | iCEBrkr | JMcA: I hate it when I have too much blood in my alcohol stream. |
| 15:22.54 | iCEBrkr | ManxPower: Wow, that sucks man |
| 15:23.17 | *** join/#asterisk strecher (n=123687@calderdale.ac.uk) |
| 15:23.31 | *** join/#asterisk yogurt2ungue (n=yogurt2u@24-48-231-201.fibertel.com.ar) |
| 15:23.33 | ManxPower | iCEBrkr, I lost all my Digium cards, all my spare hard drives, my backup drive for my TiVo.... |
| 15:23.39 | iCEBrkr | fuck |
| 15:23.43 | *** join/#asterisk Mimmus (n=viggiani@ext.pitagora.it) |
| 15:23.44 | strecher | what does Asterisk software do? |
| 15:24.00 | ManxPower | strecher, It turns you into a bitter and hateful pbx admin. |
| 15:24.00 | iCEBrkr | strecher: It's a software PBX |
| 15:24.05 | iCEBrkr | strecher: http://www.asterisk.org |
| 15:24.08 | jaike | manx: hehehe |
| 15:24.10 | strecher | thx |
| 15:24.33 | Mimmus | I flashed my phone with IAX firmware and now I'm unable to to call transfer |
| 15:24.38 | ManxPower | I realized last night that I lost my TiVo "IR Blaster" cable, since it was in the box of cables that was lost. |
| 15:24.40 | Mimmus | with SIP, I had no problem |
| 15:25.03 | strecher | What is a pbx? |
| 15:25.13 | jaike | strecher: you need to READ |
| 15:25.24 | Mimmus | neither phones buttons nor Asterisk features work anymore |
| 15:25.29 | yogurt2ungue | hello people |
| 15:25.37 | ManxPower | Mimmus, Flash it back. |
| 15:25.38 | Mimmus | any help? |
| 15:25.39 | strecher | hello |
| 15:25.45 | jaike | Mimmus: what kinda phone |
| 15:26.02 | Mimmus | ManxPower: is it normal? |
| 15:26.12 | *** part/#asterisk strecher (n=123687@calderdale.ac.uk) |
| 15:26.21 | Mimmus | jaike: an ATCom AT320 (made in china) |
| 15:26.24 | ManxPower | Mimmus, I have no idea. |
| 15:26.45 | ManxPower | Mimmus, I am not aware of anyone using those phones in a production enviroment using IAX |
| 15:26.46 | yogurt2ungue | I didn't compile the free G729 codec with Asterisk 1.2 |
| 15:26.48 | jaike | if it was working better with sip..better use sip |
| 15:27.07 | Mimmus | ManxPower: with SIP, I had some rare brief pauses during calls |
| 15:27.16 | Mimmus | ManxPower: now I'm trying IAX |
| 15:27.36 | Mimmus | IAX works better but I lost call transfer :( |
| 15:27.47 | jaike | <PROTECTED> |
| 15:28.02 | iCEBrkr | ManxPower: So I take it you're moved back in and starting over again? |
| 15:28.10 | Mimmus | jaike: I have #2 in my features.conf |
| 15:28.33 | ManxPower | iCEBrkr, move back? Um, there isn't much of a town to move back to. I moved to the top of a mountian in Alabama. |
| 15:28.36 | Mimmus | jaike: atxfer => #2, blindxfer => #1 |
| 15:28.49 | iCEBrkr | ManxPower: Ahhhh, staying there? |
| 15:29.22 | ManxPower | iCEBrkr, yup. Been wanting to move out of Waveland MS for a while, but really didn't want the hassle of moving all my stuff. |
| 15:29.35 | ManxPower | Katrina took care of much of (but not all of) my stuff. |
| 15:29.43 | iCEBrkr | :-/ |
| 15:31.32 | ManxPower | Unfortunatly all of my spare hardware that was used to build temp servers, etc was on the bottom shelves and was flooded. |
| 15:31.52 | Katty | ManxPower: you just set off my hilight. |
| 15:32.01 | MrChimpy | surely stanky flood damaged digium cards have a certain hack charm? |
| 15:32.06 | coppice | Manx: the disk chambers are supposed to be sealed :-) |
| 15:32.18 | MrChimpy | "look what *I* dredged up!" |
| 15:32.24 | Katty | iCEBrkr: kthx. |
| 15:32.28 | ManxPower | coppice, yeah, but all the other electronics were rusted. |
| 15:32.46 | coppice | rust? you have steel PCBs? |
| 15:32.46 | ManxPower | and you never can quite get all the mold off them. |
| 15:32.46 | MrChimpy | buy same drive and swap electronics |
| 15:33.24 | ManxPower | MrChimpy, I'm not trying to recover data. |
| 15:33.48 | ManxPower | All my DATA (production) systems were fine since they were on higher shelves. |
| 15:33.52 | MrChimpy | ah |
| 15:34.17 | iCEBrkr | coppice: I'm pretty sure they were in some pretty swampy water. I kinda doubt there's anything to save |
| 15:34.42 | ManxPower | iCEBrkr, the water that flooded my place was salt water. |
| 15:34.50 | iCEBrkr | ...and that too |
| 15:35.28 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-109.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 15:36.30 | ManxPower | It's not the end of the world, but it's just annoying to keep realizing that various things where destroyed. |
| 15:36.45 | ManxPower | all of my video tapes were flooded, for example. |
| 15:37.06 | coppice | all that porn lost. so sad |
| 15:37.20 | iCEBrkr | Dude. That's probably the worst lost right there.. |
| 15:37.49 | *** join/#asterisk Prival (i=user65@HSE-Montreal-ppp3474258.sympatico.ca) |
| 15:37.54 | MrChimpy | don't cry. now it's even filthier |
| 15:38.07 | fugitivo | not the porn! not the porn! |
| 15:38.12 | h3x | the worst part is nobodys gonna give a damn about marti gras so less bare boobies |
| 15:38.51 | *** join/#asterisk |vinsik| (n=vinsik@gw-ff.verkkokauppa.com) |
| 15:38.55 | coppice | they'll still find a reason to get them out. urges like that are primeaval |
| 15:38.57 | ManxPower | coppice, yes, I lost my small porn collection, but I also lost all my OTHER tapes as well, including the ones of taped shows off the television. |
| 15:38.59 | h3x | coppice hows t.38 in spandsp going |
| 15:39.25 | Prival | Got a question about the Dial options. If I use the tT options, the person originating the call can perform the transfer using the # sign, but if he/she is on an IVR of a remote PBX which asks for the # in a menu, asterisk interprets this as a transfer request... How do you make this work? |
| 15:39.31 | n3c8 | i would just like to say for the record that disk chambers are NOT sealed, and there is not a vacumn inside a disk... that was at coppice |
| 15:39.52 | MrChimpy | Katrina Appeal: Send Porn |
| 15:40.10 | ManxPower | Prival, don't use Tt and use the transfer feature of the phone. |
| 15:40.12 | MrChimpy | after the flood... |
| 15:40.20 | MrChimpy | comes the flood of mucky man yoghurt |
| 15:40.21 | h3x | Prival: How often do you need to use an ivr on an INCOMING call |
| 15:40.21 | h3x | heh |
| 15:40.28 | coppice | h3x: although the 2100 was rather late to get t.38, but I hear it works better than most. |
| 15:40.29 | iCEBrkr | STAT! |
| 15:40.33 | Katty | hrm. |
| 15:40.38 | h3x | iCEBrkr: I bet FEMA had more porn than ice and water |
| 15:40.45 | iCEBrkr | doh! |
| 15:40.47 | coppice | n3c8: who said anything about a vacuum? |
| 15:40.58 | zoa | coppice, what ata's actually do t.38 for the moment ? |
| 15:40.59 | h3x | coppice: is t.38 with asterisk still a passthru thing? |
| 15:41.01 | zoa | found any ? |
| 15:41.07 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 15:41.22 | blitzrage | zoa: ! |
| 15:41.30 | zoa | blitzrage! |
| 15:41.32 | n3c8 | they didn't but its a common misconception, as is the same with the chambers being sealed, which is not true... its like a hardware old wives tale |
| 15:42.07 | Prival | ManxPower: I'll have to retest, but I think the transfer button does not work when you initiate the call... |
| 15:42.21 | jbalcomb | [TK]D-Fender welp, i did the rxgain/txgain adjustments this morning and it didn't make wonderful things happen. :( people uses headsets are getting comlpaints on every call about it being too quiet |
| 15:42.29 | coppice | zoa: quite a few do it, but not many do it well. quite a few say on the box they do it, but don't. I have one which won an award for its t.38, and doesn't have it at all. |
| 15:42.32 | ManxPower | Prival, what make/model of phone are you using? |
| 15:42.34 | *** join/#asterisk mhnoyes (n=mhnoyes@user-38lc0bb.dialup.mindspring.com) |
| 15:43.10 | coppice | n3c8: they are sealed, but there is pressure equalisation. take the cover off outside a clean room, and you can throw the thing in the bin |
| 15:43.23 | Prival | Aastra 9133 and 480i mainly |
| 15:43.36 | Prival | h3x: the issue is on an outgoing call |
| 15:43.43 | Mimmus | jaike, ManxPower: (about call transfer) I tried again using Asterisk features and it works. Good. |
| 15:43.46 | *** part/#asterisk mhnoyes (n=mhnoyes@user-38lc0bb.dialup.mindspring.com) |
| 15:44.00 | n3c8 | its actually a very fine filter |
| 15:44.07 | n3c8 | not sealed |
| 15:44.32 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-89.rockynet.com) |
| 15:44.32 | gvag11 | coppice and Manxpower, with 1728 width and tiffg32d using gs , works fine thanks ... |
| 15:44.34 | coppice | n3c8: it could well be on some. oil from the bearing slowly pollutes the cavity |
| 15:45.18 | jbalcomb | iCEBrkr what phones are you guys using? |
| 15:45.33 | iCEBrkr | jbalcomb: We're not |
| 15:45.44 | n3c8 | they have in the past tried using a diaphagm, but it does not work well. thus nowadays it is a pretty primative filter, as you so rightly said, to allow for pressure equalization |
| 15:45.57 | iCEBrkr | jbalcomb: I use SPA2k's and a BT100 |
| 15:46.05 | *** join/#asterisk SplasPood (i=jwb@206.252.198.100) |
| 15:46.13 | tzanger | oil from which bearing pollutes t.38? :-) |
| 15:46.28 | *** join/#asterisk riddlebox (n=blah@24-171-40-167.dhcp.stls.mo.charter.com) |
| 15:46.29 | iCEBrkr | SplasPood: sup'fag |
| 15:46.59 | riddlebox | hello |
| 15:46.59 | SplasPood | :P |
| 15:47.03 | iCEBrkr | :) |
| 15:47.22 | SplasPood | I've got 5 polycoms sitting on my desk.. I wonder how many more I can fit |
| 15:47.35 | iCEBrkr | SplasPood: Sounds like you need to donate one of those to me :P |
| 15:47.43 | Katty | SplasPood: duct tape :> |
| 15:48.12 | Mimmus | if I use Asterisk features to do attended call transfer, what do I need to press to take again first call if destination doesn't answer? |
| 15:48.23 | SplasPood | iCE: they're not that expensive really.. for what you get |
| 15:48.36 | jbalcomb | iCEBrkr ah, ok. guess i'll just order the three i have on my list and go from there. are you guys using faxing through asterisk? |
| 15:48.47 | iCEBrkr | SplasPood: I know, but I rarely even use my BT100.. I just want a Polycom.. Ya know, Kinda like a status statement |
| 15:48.56 | iCEBrkr | jbalcomb: Nope.. |
| 15:49.05 | iCEBrkr | jbalcomb: I'm not using Asterisk as a PBX really.. |
| 15:49.12 | jbalcomb | iCEBrkr what are you doing with it? |
| 15:49.21 | SplasPood | ice: they're very nice |
| 15:49.22 | iCEBrkr | jbalcomb: I'm almost scared to say |
| 15:49.37 | iCEBrkr | SplasPood: Yea, my friend has them scattered through his house.. I was gonna yoink one. |
| 15:49.49 | iCEBrkr | SplasPood: BUt $200 for a phone I rarely use/tinker toy? |
| 15:49.58 | iCEBrkr | jbalcomb: IVR Surveys |
| 15:50.18 | iCEBrkr | jbalcomb: 100% completely data-driven IVR stuff. |
| 15:50.31 | jbalcomb | iCEBrkr ah, like automated telemarketing or QA stuff? |
| 15:50.33 | riddlebox | can you have asterisk connect to an ip trunk provided by another pbx? |
| 15:50.49 | iCEBrkr | jbalcomb: I use my powers for good. I won't do telemarketing. |
| 15:51.02 | jbalcomb | iCEBrkr you are good man. ;) |
| 15:51.05 | Mark_Halverson | lol |
| 15:51.09 | iCEBrkr | jbalcomb: I work for a company which handles/processes customer satisfaction surveys |
| 15:51.24 | n3c8 | sounds like direct marketing to me |
| 15:51.25 | iCEBrkr | jbalcomb: Neilson is a competitor, just to give you an idea |
| 15:51.27 | n3c8 | hehehe *jk* |
| 15:51.34 | Mark_Halverson | if i dont want the call....its telemarketing |
| 15:51.35 | Mark_Halverson | lol |
| 15:51.40 | iCEBrkr | n3c8: haha, our clients aren't trying to sell anything to anyone. |
| 15:51.52 | Mark_Halverson | if its not my mom then its telemarketing |
| 15:51.53 | iCEBrkr | Mark_Halverson: It's not telemarketing if they're not trying to sell you something. |
| 15:52.10 | iCEBrkr | Thing is, the system doesn't dial random people either. |
| 15:52.13 | fugitivo | it's just teleannonying |
| 15:52.17 | jbalcomb | iCEBrkr: i don't think i like dealing with the phone system anymore. |
| 15:52.25 | iCEBrkr | The only way we contact you is if you've had some sort of interaction with our clients. |
| 15:52.27 | Mark_Halverson | take a marketing class...by calling there marketing there customer service....helping their image...ie marketing |
| 15:52.44 | Mark_Halverson | i understnad....im just playing |
| 15:52.57 | Mark_Halverson | and expressing that if your not family i dont want to hear from you |
| 15:52.59 | Mark_Halverson | lol |
| 15:53.20 | iCEBrkr | Mark_Halverson: Hey, if you were a high-roller in Vegas and a casino called you and wanted to know how your experience went and it sucked and they comp'd you a weeks stay in the penthouse suite + free room service, I think you'd wanna take the survey. |
| 15:53.23 | Mark_Halverson | fugitivo: EXACTLY |
| 15:53.36 | fugitivo | it doesn't matter if it's marketing or not |
| 15:53.41 | ManxPower | Having an Asterisk IVR answer all calls pretty much screen out all auto dialers |
| 15:53.53 | n3c8 | the house always wins thou! |
| 15:53.55 | Mark_Halverson | Manx: yeep |
| 15:53.55 | fugitivo | all telesomething is annoying |
| 15:53.59 | iCEBrkr | ...and as for TeleAnnoying.. I've done my best for answering machine detection and it's supposed to hang up on them.. We're trying to be as passive/transparent as possible. |
| 15:54.24 | Mark_Halverson | so ice you have no live agents? |
| 15:54.30 | fugitivo | iCEBrkr: what are you using? amd? |
| 15:54.32 | iCEBrkr | fugitivo: Yea, so just don't get a phone number... Who needs phones?? |
| 15:54.39 | iCEBrkr | fugitivo: ??? |
| 15:54.45 | ManxPower | Mark_Halverson, If there's a real person on the other end when the IVR picks up, they can just select the correct option to get to me. |
| 15:54.53 | fugitivo | iCEBrkr: for answering machine detection |
| 15:55.01 | jbalcomb | I'm going to bitch slap the first telemarketer that calls my cell phone |
| 15:55.04 | fugitivo | iCEBrkr: there's a module called amd, that works pretty well |
| 15:55.06 | iCEBrkr | fugitivo: I'm using a combination of a few things... |
| 15:55.08 | fugitivo | with some tweaking |
| 15:55.24 | Mark_Halverson | oh ok...i was going to say...i think there is an FTC ruling that they MUST be able to reach a live OP within x number of seconds |
| 15:55.25 | iCEBrkr | Mark_Halverson: Nope. |
| 15:55.28 | fugitivo | what things? |
| 15:55.29 | riddlebox | in order to port a number from Verizon, do you have to have that number for a certain period of time before you can port it? |
| 15:55.42 | ManxPower | Ugh. I'm out of smokes. |
| 15:55.43 | Mark_Halverson | law school lied to me then |
| 15:55.44 | Mark_Halverson | lol |
| 15:55.56 | Mark_Halverson | 3 years of nothing |
| 15:55.56 | jbalcomb | I'm out of gumption |
| 15:55.57 | iCEBrkr | fugitivo: app_machinedetect and BackgroundDetect |
| 15:56.17 | ManxPower | which means I have to drive down the mountian |
| 15:56.21 | fugitivo | iCEBrkr: amd replaces all that |
| 15:56.23 | iCEBrkr | fugitivo: I've had some good results. I actually have a 'test bed' of answering machines I've dialed against |
| 15:56.28 | iCEBrkr | fugitivo: Ooo!! Ooo!! |
| 15:56.29 | fugitivo | iCEBrkr: wait, not for faxes |
| 15:56.38 | fugitivo | are you detecting faxes too? |
| 15:56.59 | iCEBrkr | fugitivo: I haven't tested that 100%, but I thought Asterisk landed in a fax extension |
| 15:57.21 | fugitivo | that's right, but not using sip or iax trunks |
| 15:57.37 | iCEBrkr | fugitivo: We're not doing VoIP yet |
| 15:58.04 | fugitivo | ok, remember that when doing voip :) |
| 15:58.18 | *** join/#asterisk razu (n=razu@213-35-173-39-dsl.prn.estpak.ee) |
| 15:58.21 | fugitivo | there's another module for fax detection |
| 15:58.31 | fugitivo | app_nfaxdetect or something like that |
| 15:58.53 | fugitivo | it works for sip/iax |
| 15:58.57 | ManxPower | yes. I use it in my autofax scripts |
| 15:59.05 | iCEBrkr | app_machinedetect works. It's a bitch to tweek tho. |
| 15:59.13 | tzanger | heh |
| 15:59.19 | tzanger | in 17 days we had 5200 calls |
| 15:59.29 | fugitivo | iCEBrkr: i tried using that and background detect, but i found amd and it was easier to teak |
| 15:59.32 | iCEBrkr | not to mention, I had to make multiple calls to it.. |
| 15:59.44 | iCEBrkr | fugitivo: I'll look into that for v2 :P |
| 15:59.51 | coppice | iCEUBrkr: you must be using a definition of works with which I am not familiar |
| 15:59.54 | iCEBrkr | Too close to the deadline to start muck'n with it now |
| 16:00.25 | iCEBrkr | coppice: It was literally 2wks of work trying to come up with a "reliable" way to use it. |
| 16:00.36 | iCEBrkr | I dunno if reliable is even the word to use. |
| 16:00.47 | coppice | I'm not familar with this use of reliable, either |
| 16:00.49 | iCEBrkr | I just know I lost much hair over it |
| 16:01.09 | iCEBrkr | But I finally was able to come up with something that even works against my T-Mobile voicemail |
| 16:01.24 | *** join/#asterisk steve___ (n=steve@store-fw.porchlight.ca) |
| 16:01.25 | coppice | iCEBrkr: ah, that part I am familiar with :-) |
| 16:01.30 | iCEBrkr | I could guestimate that I'm able to detect 90% of the machines out there |
| 16:01.37 | iCEBrkr | haha |
| 16:01.47 | iCEBrkr | Tho, I only have 6 machines here to test against |
| 16:02.07 | iCEBrkr | Those digital ones are a bitch. I left the preset greeting on the one... |
| 16:02.11 | X-Files | hey all! Microsoft Windows Messenger 5.1 (not MSN) work fully Message and Contact list STREAM ???? |
| 16:02.19 | iCEBrkr | Sometimes it'll get it.. sometimes it'll miss and leave a message |
| 16:02.26 | coppice | iCEBrkr: you suspect that because you haven't tried it enough. as you try it some more the picture will change :-) |
| 16:02.56 | fugitivo | X-Files: asking the same question like a spambot won't give you any answer |
| 16:03.08 | iCEBrkr | coppice: Well it's all about doing a tap dance with silence detection and noise detection.. Fine grey line and making the best descion |
| 16:03.27 | *** join/#asterisk edwin__ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 16:03.34 | fugitivo | iCEBrkr: this app_amd module, will count words also |
| 16:03.35 | iCEBrkr | regardless, it's a frick'n pain in the ass. |
| 16:03.41 | iCEBrkr | fugitivo: sweet |
| 16:03.51 | X-Files | ;| |
| 16:03.55 | iCEBrkr | fugitivo: I'll start looking into that so when this blows up I'll have a backup plan :P |
| 16:03.56 | tzanger | for fax detection? Why not just detect the echo can disable tone? |
| 16:04.03 | coppice | iCEBrkr: been there several times, trying to convince the people who read the crap that say it works 99% of the time that the 99% is bogus |
| 16:04.10 | tzanger | I realize that modems use it too but if you're just looking to hang up... |
| 16:04.14 | fugitivo | tzanger: answering machine detection |
| 16:04.18 | tzanger | fugitivo: ah |
| 16:04.18 | coppice | there is no echo can disable tone from a fax |
| 16:04.24 | tzanger | coppice: there isn't? |
| 16:04.54 | *** join/#asterisk killer-ch (n=killer-c@quasimodo.csn.tu-chemnitz.de) |
| 16:04.55 | tzanger | how is it I see dozens of "echo canceller disabled due to tone (rx) on channel 'x'" in dmesg for my fax dids? |
| 16:05.10 | coppice | that's the result of a fax tone detector |
| 16:05.35 | fugitivo | iCEBrkr: this module will count words, silence, words before and after silence, noise, etc, you can tweak all that parameters to detect an answering machine, obviously it's not 100% accurate |
| 16:05.51 | fugitivo | i don't think it's 90% accurate |
| 16:06.04 | iCEBrkr | fugitivo: Yea, nothing like that is. |
| 16:06.05 | [TK]D-Fender | jbalcomb : Your audio is too low because you have whacked out gain's in the negative. Why are you even playing with that? |
| 16:06.21 | iCEBrkr | fugitivo: But if it removes this crazy skip-logic I have, it'll make things easier on me |
| 16:06.53 | fugitivo | iCEBrkr: i think it will, it did for me |
| 16:06.55 | MrChimpy | must stop giggling at http://snipurl.com/hellofloatydogwoof - the office think i'm doing dialplans but i'm watching that over and over again |
| 16:07.10 | iCEBrkr | Cool |
| 16:07.13 | coppice | the fact that you need to tweak should tell you its almost useless |
| 16:07.15 | jbalcomb | [TK]D-Fender http://www.voip-info.org/wiki/view/Grandstream+GXP-2000+-+Solving+Echo+Problems |
| 16:07.16 | iCEBrkr | Thanks for the info |
| 16:08.13 | *** join/#asterisk Traderzz (n=Trazz@ip-66-80-141-13.chi.megapath.net) |
| 16:08.14 | *** join/#asterisk kannan (n=kannan@dsl-Chn-static-223.45.101.203.touchtelindia.net) |
| 16:08.58 | [TK]D-Fender | jbalcomb : Thats like solving an ant problem by moving to Antactica. Your PHONES are teh problem. You've admitted as much. Either roll-back those "newly essential" features or be prepared to suffer. |
| 16:09.14 | [TK]D-Fender | jbalcomb : Either that or change your T1 card. |
| 16:10.56 | jbalcomb | [TK]D-Fender *shrug* im storta stuck with the way things are now as far as the phones go. I dont know anything about the T1 card idea. |
| 16:11.19 | killer-ch | can anyone tell me if it is possible register at an asterisk with the same account from different phones? |
| 16:11.47 | [TK]D-Fender | jbalcomb : Do you get echo from phone-phone, or phone-pstn only? |
| 16:12.22 | [TK]D-Fender | jbalcomb : Though I am aware the a large portion of the echo is directed at the GXP's |
| 16:12.38 | jbalcomb | [TK]D-Fender there is some echo phone-phone but the majority of issues are phone-pstn |
| 16:12.43 | [TK]D-Fender | killer-ch : No. Shared line appearance support does not exist in * yet |
| 16:12.55 | killer-ch | thx [TK]D-Fender |
| 16:13.25 | [TK]D-Fender | jbalcomb : You could reduce your overall problem to the size of the smallest occurance (phone-phone) if you were to switch to a good EC capable card. |
| 16:14.26 | jbalcomb | [TK]D-Fender what is EC and where can I find material on this idea? |
| 16:14.45 | beebz | which ntp do most people use? |
| 16:14.55 | beebz | ntp server rather |
| 16:14.58 | [TK]D-Fender | jbalcomb : Echo Cancellation? |
| 16:15.05 | [TK]D-Fender | beebz : pool.ntp.org |
| 16:15.20 | jbalcomb | [TK]D-Fender ah, yes, of course |
| 16:15.29 | beebz | stkn: boner, currently using that and my polycomes are getting |Could not load time from 202.55.152.4(202.55.152.4). << and thats from the pool |
| 16:15.37 | beebz | err, htat was to [TK]D-Fender |
| 16:15.51 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:15.51 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:16.05 | [TK]D-Fender | beebz : I have 27 here that work just fine. You sue theres a route for them to access it? |
| 16:16.17 | [TK]D-Fender | s/sue/sure/ |
| 16:16.29 | beebz | lol |
| 16:16.39 | beebz | lovely bot |
| 16:17.43 | beebz | :q<SNTP tcpIpApp.sntp.resyncPeriod="86400" tcpIpApp.sntp.address="pool.ntp.org" tcpIpApp.sntp.gmtOffset="-21600" tcpIpApp.sntp.daylightSavings.enable="1" |
| 16:17.47 | BeHappy_ | uhm.. it's safe to directly access the asterisk database? for example for controlling the blacklist via a webapp |
| 16:17.49 | beebz | should i bump my resync down a bit? |
| 16:18.02 | beebz | BeHappy_: the cdrdb? |
| 16:18.08 | BeHappy_ | beebz, nope |
| 16:18.12 | fugitivo | BeHappy_: nothing is safe, specially from the web |
| 16:18.14 | BeHappy_ | the asterisk db |
| 16:18.42 | BeHappy_ | fugitivo, of course, but all the code i've seen calls asterisk via a shell |
| 16:19.12 | BeHappy_ | fugitivo, i think it should be safer to hook the database file from my app, but i dont know if this could be done |
| 16:19.55 | [TK]D-Fender | beebz : I think the GMT offset should be in hours.... |
| 16:20.03 | *** join/#asterisk pigpen2 (n=mark@66.118.8.82) |
| 16:20.48 | [TK]D-Fender | beebz : Do you have a DHCP server dishing out that parameter as well? |
| 16:22.11 | coppice | a GMT offset in hours would have problems in places like India |
| 16:22.25 | beebz | [option nntp-server "pool.ntp.org"; |
| 16:22.29 | [TK]D-Fender | coppice : I'll take your word for it... |
| 16:22.31 | beebz | err, minus the [ |
| 16:22.41 | [TK]D-Fender | beebz : and the offset in dhcpd.conf? |
| 16:22.44 | coppice | India is GMT + 5.5 hours |
| 16:22.50 | beebz | [TK]D-Fender: correct |
| 16:22.57 | [TK]D-Fender | coppice : You can do fractions, silly! |
| 16:23.24 | [TK]D-Fender | beebz : paste it plz |
| 16:23.26 | coppice | don't be silly. computers only understand integers |
| 16:23.32 | *** join/#asterisk Flusher (i=flusher@filer.euroserv.com) |
| 16:23.35 | Flusher | hi |
| 16:23.47 | [TK]D-Fender | beebz : Here's mine for EST |
| 16:23.49 | [TK]D-Fender | option ntp-servers pool.ntp.org; |
| 16:23.49 | [TK]D-Fender | option time-offset -18000; |
| 16:23.57 | beebz | option nntp-server "pool.ntp.org"; |
| 16:23.57 | beebz | option time-offset -6; # Central Standard Time |
| 16:24.18 | [TK]D-Fender | beebz : Umm.. you need to fix that :) |
| 16:25.01 | *** join/#asterisk sdf (n=lala@222.185.17.230) |
| 16:25.49 | gaupe | beebz: nntp-server? |
| 16:26.17 | gaupe | do you have usenet enabled ip-phones? |
| 16:26.18 | [TK]D-Fender | too many "n"'s, and the offset type is the wrong UOM |
| 16:26.39 | *** join/#asterisk HamYai (n=HamYai@125.24.9.193) |
| 16:27.14 | jbalcomb | feh. ugh. thrrp. fuck asterisk and double fuck grandstream. |
| 16:28.16 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 16:28.44 | darkskiez | jbalcomb: Calm down dear |
| 16:29.57 | *** join/#asterisk Prival (n=someone@209-161-233-37.dsl.look.ca) |
| 16:30.17 | jbalcomb | darkskiez :) sry. just losing my grip. been awhile since I had to deal with something I couldn't grasp. |
| 16:30.28 | Prival | I just saw that asterisk stopped logging in /var/log/asterisk aster a logrotate... Any hints? |
| 16:30.51 | jbalcomb | Prival there is a 'logger restart' command or something in the CLI you have run |
| 16:30.54 | HamYai | Hi, my cdr tends to provide incorrect "billing" records |
| 16:31.10 | darkskiez | prival: logrotate has a copy and truncate option, try that also |
| 16:31.13 | HamYai | anyone having the similar problem? |
| 16:31.35 | darkskiez | HamYai: incorrect, how so? |
| 16:31.44 | Prival | Ok, will give that a shot. Thanks. |
| 16:31.44 | DarkFlibble | [TK]D-Fender, there are islands that are 23:52minutes offset |
| 16:32.38 | *** join/#asterisk ErMeS|Work (n=ermsewrk@217.220.121.62) |
| 16:33.36 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 16:35.44 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 16:35.44 | *** mode/#asterisk [+o denon] by ChanServ |
| 16:36.18 | HamYai | darkskiez: I'm making outgoing calls via FXOs and the line seems to answer nearly immediately |
| 16:36.39 | *** join/#asterisk apardo (n=apardo@62.97.121.95) |
| 16:36.49 | [TK]D-Fender | Prival : Maybe do a "touch" on it to refresh the day. |
| 16:36.53 | *** join/#asterisk m0narch (n=r3b3l@melloyello.mmi.net) |
| 16:37.17 | basta | Hama: maybe there's an answer in you dialplan before the actual dial ? |
| 16:37.24 | HamYai | darkskiez: so, my cdr recorded all calls as being answered |
| 16:37.29 | ManxPower | HamYai, Yes. Analog ports are considered answered as soon as dialing is done |
| 16:37.50 | HamYai | ManxPower: is there a way to fix this? |
| 16:37.58 | ManxPower | HamYai, Don't use analog |
| 16:38.22 | *** join/#asterisk EriSan (n=erisan@81-174-25-141.f5.ngi.it) |
| 16:38.31 | ManxPower | The telco cannot provide any indication to Asterisk that the call has been answered when using analog ports |
| 16:38.35 | [TK]D-Fender | Prival : Oh and for your transfer issue earlier : if you're using a SIP hard-phone you shouldn't need DTMF based trnasfers and yes it would royally suck if you're in an IVR |
| 16:39.10 | HamYai | ManxPower: I heard that gnudialer can detect these things, is it true? |
| 16:39.14 | [TK]D-Fender | jbalcomb : PM |
| 16:39.38 | *** join/#asterisk lonelone (n=nameee@217.52.57.71) |
| 16:40.13 | ManxPower | HamYai, I doubt it. It may TRY, but I strongly doubt it can do it reliability. |
| 16:40.23 | HamYai | ManxPower: how do the tecos record the right billing time then? |
| 16:40.36 | DarkFlibble | HamYai, using hardware |
| 16:40.36 | lonelone | hi all .. one fast question . i use a asteriska at home server and sometime ppl who is behind firewall cannot hear voice of the other side .. ( one way sound ) .. what could that be ? |
| 16:40.44 | ManxPower | HamYai, they do not use analog ports |
| 16:41.19 | ManxPower | or more correctly they do not use analog FXO ports. On an FXS port this is not an issue and when you get an analog phone line from the telco it's an FXS port from their equipment's perspective |
| 16:41.21 | ManxPower | ~fxofxs |
| 16:41.23 | jbot | somebody said fxofxs was An FXO port expects to receive dialtone and receive ring voltage, you plug a phone LINE into this. An FXS port expects to provide dialtone and provide ring voltage, you plug a PHONE into this. |
| 16:42.32 | HamYai | ManxPower: I might consider changing to ISDN PRI then |
| 16:42.59 | ManxPower | HamYai, Yes. If you just want to do testing, etc then get a VoIP account from a service provider like Teliax. |
| 16:43.01 | DarkFlibble | HamYai, use digital technology over analogue where possible in telecoms |
| 16:43.04 | ManxPower | They all use PRIs |
| 16:43.23 | ManxPower | But I would suggest you get a PRI ISDN before going into production |
| 16:43.44 | HamYai | ManxPower: is it DIALSTATUS on ISDN that provides the status of the line? |
| 16:43.45 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 16:43.59 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
| 16:44.01 | [TK]D-Fender | HamYai : How many "lines" are you looking to get? |
| 16:44.17 | ManxPower | HamYai, DIALSTATUS and HANGUPCAUSE |
| 16:45.01 | HamYai | [TK]D-Fender: I was to start with 8 port FXOs but gotta change my mind |
| 16:45.36 | HamYai | ManxPower: is there a way to play audio files before the "Answer" is called? |
| 16:46.17 | ManxPower | HamYai, Yes, with limitations |
| 16:46.32 | ManxPower | HamYai, you almost never need to actually run the Answer command |
| 16:46.47 | *** join/#asterisk Mike (n=mike@201.135.48.190) |
| 16:46.54 | HamYai | ManxPower: no interaction of key press is allowed? |
| 16:48.07 | [TK]D-Fender | HamYai : See if you can get a partial PRI otherwise the cost difference could add up to a lot... |
| 16:48.08 | HamYai | ManxPower: I found it useful to run the "Answer" sometimes |
| 16:49.12 | HamYai | [TK]D-Fender: yeah, to get an ISDN PRI, it will cause me around $2,500 here |
| 16:49.42 | ManxPower | HamYai, Playback, Background, and most things that play audio automatically ANSWER the line. |
| 16:49.46 | HamYai | [TK]D-Fender: and $180 monthly fee |
| 16:49.48 | [TK]D-Fender | HamYai : And for analog FXO? |
| 16:50.19 | ManxPower | HamYai, on PRI most telcos allow one-way audio before answer. caller -> callee audio, but not callee -> caller audio |
| 16:50.49 | ManxPower | this allows for the destination to play things like "the number you have called is disconneted or you are an idiot and dialed the wrong number" message to the caller without answering the line so the caller isn't billed. |
| 16:51.04 | HamYai | ManxPower: in Taxable IVR systems, I will need to ask if a caller really needs to enter the system |
| 16:51.29 | ManxPower | HamYai, you can ask all you want, but you can't receive DTMF from them unless you answer the call. |
| 16:51.39 | HamYai | ManxPower: if not, they can hang up and need not be charged |
| 16:51.58 | ManxPower | HamYai, you would need a PRI for that application |
| 16:52.06 | *** part/#asterisk fuzza (n=andrew@ppp171-157.lns1.per1.internode.on.net) |
| 16:52.37 | ManxPower | brb |
| 16:52.37 | *** join/#asterisk voipjjs (n=voipjjs@d28-25.rt-bras.wnvl.centurytel.net) |
| 16:52.37 | HamYai | ManxPower: yeah, that's what I've heard of. should there be a fix anyway |
| 16:53.11 | DarkFlibble | HamYai, why would they dial the number if they didn't want to enter the system? |
| 16:53.44 | *** join/#asterisk justinu (n=justinu@cpe-72-129-86-208.socal.res.rr.com) |
| 16:53.51 | HamYai | [TK]D-Fender: an analog line fee is 1/30 of ISDN PRI |
| 16:54.21 | [TK]D-Fender | HamYai : Well I guess you have to ask yourself how much you are going to save with PRI then. |
| 16:54.22 | HamYai | DarkFlibble: the number may be promoted on a TV spot |
| 16:54.36 | *** join/#asterisk ManxPowe (n=ewieling@dpc6745150107.direcpc.com) |
| 16:54.54 | HamYai | DarkFlibble: those callers enter the system without knowing that they'd be charged |
| 16:54.57 | ManxPowe | there. |
| 16:55.17 | ManxPowe | HamYai, the telco does not permit 2-way audio before answer to prevent fraud. |
| 16:55.22 | DarkFlibble | you don't have legal requirements to display the call cost on any adverts over there? |
| 16:55.47 | jbalcomb | [TK]D-Fender PM |
| 16:56.00 | HamYai | DarkFlibble: the spots are to short, like 15 secs |
| 16:56.35 | *** join/#asterisk pigpen2 (n=mark@66.118.8.82) |
| 16:57.22 | MrChimpy | got a TE411P with E1 just connected. seems to go to NOP and stay there. is this normal? |
| 16:57.48 | ManxPowe | MrChimpy, NOP? |
| 16:58.03 | MrChimpy | *CLI> zap show status |
| 16:58.03 | MrChimpy | Description Alarms IRQ bpviol CRC4 |
| 16:58.03 | MrChimpy | T4XXP (PCI) Card 0 Span 1 RED/NOP 0 0 0 |
| 16:58.03 | MrChimpy | T4XXP (PCI) Card 0 Span 2 RED/NOP 0 0 0 |
| 16:58.04 | MrChimpy | T4XXP (PCI) Card 0 Span 3 NOP 0 0 0 |
| 16:58.11 | HamYai | DarkFlibble: they might just remember the numbers not that they'd be charged. We are obliged to play the announcement prompt |
| 16:58.11 | MrChimpy | T4XXP (PCI) Card 0 Span 4 NOP 0 0 0 |
| 16:58.11 | DarkFlibble | still... in the UK every tv advert that displays a non-geographic number must display the call cost... |
| 16:58.20 | DarkFlibble | kk |
| 16:58.20 | MrChimpy | the two ports showing nop have an E1 connected |
| 16:59.04 | coppice | Mr Chimpy: that display is bogus, since the T4XXP driver doesn't count the errors |
| 16:59.40 | DarkFlibble | HamYai, HongKong? |
| 16:59.57 | HamYai | DarkFlibble: in Thailand |
| 17:00.00 | DarkFlibble | k... |
| 17:00.16 | DarkFlibble | couldn't resolve any hosts after hongkong |
| 17:00.31 | HamYai | coppice: are you still working on the Unicall Lib? |
| 17:00.41 | coppice | yes |
| 17:01.27 | HamYai | coppice: with R2 MFC, is the key press allowed before answering the line? |
| 17:01.28 | [TK]D-Fender | jbalcomb : That meant "private message". You should ahve another tab for that conversation in most clients... |
| 17:01.30 | MrChimpy | coppice: um, that display is from asterisk CLI? |
| 17:01.54 | MrChimpy | presumably it'll go GREEN when happy. |
| 17:02.05 | *** join/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 17:02.14 | coppice | MrChimpy: yes, but the driver isn't reporting the real counts for the T4XXP driver |
| 17:02.40 | MrChimpy | not really looking at the counts, just watching the states |
| 17:03.02 | coppice | HamYai: that isn't really anything to do with Unicall |
| 17:03.16 | ManxPowe | MrChimpy, RED means "no line connected" |
| 17:03.39 | MrChimpy | we're getting NOP for lines that are |
| 17:03.59 | HamYai | coppice: I'm wondering because my callers can send me DTMF before being charged |
| 17:04.04 | MrChimpy | and getting a lot of : Jan 20 16:49:50 rh4-asterisk2 kernel: 2G: Got interrupt, status = 0000ff0a, GIS = 0080 |
| 17:04.04 | MrChimpy | Jan 20 16:49:50 rh4-asterisk2 kernel: Tried to load 00000020 into 0000000a, but got 0000006f instead |
| 17:04.10 | MrChimpy | which doesn't look good |
| 17:05.34 | HamYai | coppice: upon receiving the DTMF, I send a pulse of 120 msecs to teco to indicate charging |
| 17:05.37 | *** join/#asterisk iDunno (i=brettp@stef.sommitrealweird.co.uk) |
| 17:05.41 | coppice | HamYai: I think the DTMF receiver in chan_unicall starts when call is accepted, so I may pick up digits before the call is answered |
| 17:05.51 | HamYai | coppice: I really want to do that with asterisk |
| 17:06.23 | coppice | HamYai: where are you? |
| 17:06.33 | HamYai | coppice: in Thailand |
| 17:07.01 | *** join/#asterisk iccomputing (n=Wireless@cpe-71-65-112-38.woh.res.rr.com) |
| 17:07.20 | HamYai | coppice: that means you can also detect DTMF before answering the line? |
| 17:07.46 | HamYai | coppice: with asterisk + unicall |
| 17:07.51 | coppice | Ah, so you want to send a charging pulse after answer. I haven't implemented those fully. very few places use charging pulses, and they are not part of the ITU R2 spec. I started implementing them in recent changes, but haven't finished that |
| 17:09.02 | *** join/#asterisk apardo (n=apardo@62.97.121.95) |
| 17:09.20 | HamYai | coppice: I think we call it the "OFFER" state when the call is received |
| 17:09.22 | *** join/#asterisk Math` (n=math@modemcable148.4-81-70.mc.videotron.ca) |
| 17:09.45 | HamYai | coppice: once the pulse is sent, it's in the "ANSWERED" state |
| 17:09.49 | jarrod | anyone using sipura 2100 t.38 with cisco gateway? |
| 17:10.29 | HamYai | coppice: sorry to mention it, but it's in the Global Call Spec of Dialogic |
| 17:10.36 | coppice | answered is not a pulse. its a continuous change of state |
| 17:11.44 | HamYai | coppice: so, you reckon that the line is answered initially |
| 17:11.46 | *** join/#asterisk aminorex (n=tony@71-13-40-131.dhcp.dlth.mn.charter.com) |
| 17:12.13 | HamYai | coppice: and a pulse is sent afterward to indicate charging |
| 17:12.46 | *** join/#asterisk ToTo (n=ToTo@host221-49.pool870.interbusiness.it) |
| 17:13.08 | coppice | Generally Thailand behaves like China. answer is a change of line state persisting until hangup. however, some online services need to send a billing pulse after answer |
| 17:14.33 | HamYai | coppice: okay, that maybe the reason I could receive DTMF before they are charged |
| 17:15.03 | *** join/#asterisk shido6 (n=shido@d221-68-216.commercial.cgocable.net) |
| 17:15.03 | HamYai | coppice: is the pulse hard to implement? |
| 17:15.58 | coppice | no hard. I just haven't done it, as few people need it |
| 17:16.04 | HamYai | coppice: I am running some systems on MFC R2 but the pulse is really my problem here |
| 17:16.35 | Flusher | re |
| 17:16.37 | Flusher | I have this topology : [ast serv] - [nat router w/ fixed IP] --- internet --- [nat router] - [sip phone] |
| 17:16.38 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 17:17.17 | Flusher | my sip phone registers on the asterisk server when it is in the local subnet but i get a "unauthorized" (401) error when it's remote |
| 17:17.22 | coppice | someone else in thailand prompted me to start implementing billing pulses, but I didn't hear from hime again. if you want them I'll take another look |
| 17:17.53 | Flusher | of course i forwarded some ports (UDP/5060, 3748, 8000-8012) and forced RTP on ports 8000-8012 |
| 17:17.55 | HamYai | coppice: guess it was me |
| 17:18.10 | jarrod | id forward 16384-32767 |
| 17:18.11 | justinu | how do they charging pulses work? |
| 17:18.11 | Flusher | would anyone familiar with this problem / have a solution pls ? |
| 17:18.12 | jarrod | udp |
| 17:18.12 | HamYai | coppice: I came in last time and talked to you about that |
| 17:18.18 | *** part/#asterisk m0narch (n=r3b3l@melloyello.mmi.net) |
| 17:18.25 | jarrod | oh forced rtp |
| 17:18.29 | coppice | this was several months ago |
| 17:18.30 | Flusher | yes, jarrod ;) |
| 17:18.42 | Flusher | i didnt want to forward so many ports |
| 17:18.47 | *** join/#asterisk Navman_Lap (n=icechat5@62.108.206.82) |
| 17:18.52 | HamYai | coppice: yeah, I only came in 3-4 times since then |
| 17:19.05 | Flusher | moreover it appears that RTP is used when calling, not at register time |
| 17:19.11 | *** join/#asterisk pifiu (n=someone@216.5.79.1) |
| 17:19.26 | justinu | RTP is only for media (audio, generally) |
| 17:19.31 | HamYai | coppice: it was last year during May |
| 17:19.39 | Flusher | I forgot to say that I have set nat=yes and qualify=yes in my extension |
| 17:19.40 | shmaltz | the SPA 941 is realy impressive |
| 17:19.41 | jbalcomb | Flusher do you have localnet in the sip.conf for that ext? |
| 17:19.47 | shmaltz | I got my first one today |
| 17:19.51 | *** join/#asterisk roulduke_ (i=9icaw3qh@p508D3A61.dip0.t-ipconnect.de) |
| 17:20.01 | HamYai | coppice: a lot of people here still use MFC R2 |
| 17:20.16 | coppice | it was probably you, then. you need to keep prodding if you want do maintain my interest :-) |
| 17:20.24 | Flusher | jbalcomb: affirmative, i ve set the external ip and the local subnet in my sip_nat.conf, which is included in my sip.conf (i m using asterisk @ home on this server) |
| 17:20.27 | pifiu | hey everyone |
| 17:20.34 | jbalcomb | ok |
| 17:20.49 | HamYai | coppice: yeah, I sure will |
| 17:21.02 | coppice | a lot of people in many countries use R2. when I implemented it I was mostly targetting south america and china. now I find people in all sorts of places using it |
| 17:21.02 | Flusher | i also tried to specify "host=my current ip address" for the extension, but it fails as weel |
| 17:21.33 | HamYai | coppice: we're lucky to have people like you by the way |
| 17:21.48 | justinu | HamYai: very true... 3 cheers for coppice |
| 17:22.17 | coppice | V.29rx is just fine |
| 17:22.40 | justinu | it is? |
| 17:22.47 | justinu | someone was saying it didnt' receive to well |
| 17:23.29 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 17:23.34 | coppice | it receives just fine. lots of people have broken systems though, and blame the software |
| 17:23.44 | justinu | lol, fair enough |
| 17:28.48 | Flusher | anyone familiar with asterisk behind nat ? i'd need some help since i get a" unauthorized (401)" (forbidden) error at sip auth time from remote sip phones. |
| 17:29.36 | MrChimpy | ok, to get this TE411P working I'm modprobing zaptel and wct4xxp - am I missing something there |
| 17:29.39 | MrChimpy | ? |
| 17:31.27 | ManxPowe | MrChimpy, If you get a RED alarm then the card is not seeing a line. Is this a new line or an existing line moved from a different piece of equipment? |
| 17:31.48 | MrChimpy | new line |
| 17:31.51 | ManxPowe | MrChimpy, And have you looked at the README in the zaptel source directory to confirm yuo are loading the correct driver for your card? |
| 17:32.00 | MrChimpy | the best we get when connecting is a NOP |
| 17:32.23 | ManxPowe | MrChimpy, Call your telco. Say "I have a RED Alarm. Please fix it." Get a trouble ticket number. Pray. |
| 17:32.44 | MrChimpy | yeah. that's kind of the problem. my good colleague is my telco :) |
| 17:33.18 | Corydon-w | MrChimpy: are you using a straight-through cable to connect it? |
| 17:33.52 | Corydon-w | You might try a T1 crossover cable if you're using a straight-through or vice versa |
| 17:33.55 | MrChimpy | yep. got right driver |
| 17:34.02 | *** join/#asterisk wundaboy (n=asdf@c-24-21-100-201.hsd1.or.comcast.net) |
| 17:34.18 | MrChimpy | it's an E1 and we're using straight through. other end is a telco switch |
| 17:34.22 | ManxPowe | for some reason I thought the driver was wcte4xxp |
| 17:34.28 | MrChimpy | i'll get him to try the crossover |
| 17:34.33 | *** join/#asterisk flujan (n=flujan@internet.nube.com.br) |
| 17:34.46 | Corydon-w | MrChimpy: remember to use a T1 crossover, not an Ethernet crossover. |
| 17:34.53 | MrChimpy | wct4xxp TE405P - Quad Span T1/E1 Card (5v version) |
| 17:34.53 | MrChimpy | <PROTECTED> |
| 17:35.04 | MrChimpy | cory: yeah, got the pinout for that already |
| 17:35.12 | flujan | hi all.. I work for a call center which make a lot of dialed calls by day... |
| 17:35.20 | ManxPowe | MrChimpy, must be the 1 port cvards I was thinking |
| 17:35.20 | MrChimpy | would have kind of expected to see nothing if the cable was wrong... |
| 17:35.21 | Corydon-w | I've never seen a prebuilt t1 crossover; a lot of people try to use Ethernet crossover |
| 17:35.50 | MrChimpy | if you do connect a valid link the state in zztool should go green straight off? |
| 17:36.04 | Corydon-w | yellow straight off and green within a few seconds |
| 17:36.15 | flujan | We basically do a Select with something about 100 numbers and so dial it... When the call is answer, they hold until there's a operator available. |
| 17:36.22 | Corydon-w | yellow alarm means it detected a red alarm at the other end |
| 17:36.28 | MrChimpy | sounds good to me then. cable time. |
| 17:36.43 | flujan | we call a consulting company, and they just say that asterisk is a bad solution... Since it works bad with a lot of calls. |
| 17:36.46 | flujan | is it true? |
| 17:37.09 | Corydon-w | flujan: I bet they're trying to sell you something |
| 17:37.13 | justinu | they just want you to spend more money |
| 17:37.15 | zoa | flujan: bullshit |
| 17:37.16 | rob0 | haha probably means they don't know how to set it up |
| 17:37.20 | flujan | And Could the use o audiocodecs minimize the burden of processing each call in the server? |
| 17:37.26 | zoa | i do 2 million calls a month on asterisk |
| 17:37.42 | flujan | zoa, wich is your hardware configuration? |
| 17:38.03 | justinu | well... using a SIP gateway will reduce load on the server in lieu of zaptel cards |
| 17:39.28 | zoa | dual xeons |
| 17:39.48 | Math` | zoa: testing a quad xeon now with asterisk |
| 17:40.07 | benjk | zoa: you must be on the phone all day and all night long |
| 17:40.09 | Math` | (729 <> ulaw transcoding) |
| 17:40.55 | *** join/#asterisk razu (n=razu@217-159-187-162-dsl.prn.estpak.ee) |
| 17:41.26 | justinu | math: how many simultaneous? |
| 17:42.07 | benjk | flujan: try calling Coca-Cola and ask them if they will help you set up a network of Pepsi vending machines |
| 17:42.23 | Math` | justinu: dunno yet the company is moving from h323 to sip and usually the cisco gateway does g729, but another provider requires ulaw and they want to push traffic to it |
| 17:42.31 | Math` | so now the box is just sleeping |
| 17:42.34 | justinu | ah |
| 17:42.40 | justinu | i use sipp to test stuff like that |
| 17:42.51 | Math` | sipp.sf.net? |
| 17:42.52 | flujan | benjk: I know, but my boss asked me to came here as ask for information, or poof... It's sad to work in a company like this... :( |
| 17:42.58 | justinu | math: yep |
| 17:42.58 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 17:43.16 | justinu | flujan: to hell with it, let them waste their money |
| 17:43.20 | benjk | sad or not, the point is that the answer depends on who you ask |
| 17:43.23 | Math` | justinu: nice! we were looking for a way to benchmark it, I'll let you know how many simultaneous calls it can transcode |
| 17:43.30 | justinu | math: cool |
| 17:43.30 | zoa | math, dont use quad xeons |
| 17:43.38 | Math` | zoa: why? |
| 17:43.47 | justinu | i'm not a big fan of SMP machines in general |
| 17:43.56 | zoa | because they are expensive |
| 17:44.18 | Math` | zoa: when you find a quad xeon box sitting there converting only signalling between h323 and sip, you use it |
| 17:44.22 | *** part/#asterisk darkskiez (n=darkskie@194.247.78.146) |
| 17:44.29 | flujan | zoa, how many dual zeons did you have? |
| 17:44.39 | zoa | 4 |
| 17:44.41 | zoa | 2 on each end |
| 17:45.52 | justinu | math: with sipp, i benchmarked a single xeon 3.0 out at about 80 G729 channels |
| 17:46.02 | Math` | ok |
| 17:46.15 | benjk | SIPP? |
| 17:46.20 | Math` | http://sipp.sf.net/ :) |
| 17:46.48 | benjk | ah ok |
| 17:47.00 | zoa | g729 makes it more difficult |
| 17:47.04 | justinu | about 180 g711 channels before the load average went thru the roof |
| 17:47.10 | zoa | i'd say 120 max per dual xeon |
| 17:48.23 | flujan | thanks zoa |
| 17:49.00 | flujan | zoa, where can I find documentation about asterisk working in " power dialer' environments? |
| 17:49.29 | Math` | power dialer? |
| 17:49.31 | zoa | you can only try it for yourself i think |
| 17:49.36 | zoa | there are too much variables to fill in |
| 17:50.13 | *** join/#asterisk greendisease (n=greendis@fedora/greendisease) |
| 17:51.57 | MrChimpy | if you mean cold sales calling "power dialling" you should go with a commercial solution |
| 17:52.15 | MrChimpy | not because it's better, just because you bastards deserve to lose money |
| 17:52.37 | *** join/#asterisk penghb (n=npenghb@202.108.130.138) |
| 17:53.28 | justinu | MrChimpy: agreed |
| 17:53.50 | flujan | Math`, dial to a lot of phones in just one time... ;) |
| 17:54.12 | Math` | MrChimpy :) |
| 17:54.35 | zoa | flujan, if you are new to asterisk, i would not recommend to try it yourself (unless you have time to work on it) |
| 17:54.50 | zoa | or if you can afford some downtime |
| 17:58.27 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 17:59.36 | *** join/#asterisk kram (n=mark@gateway.digium.com) |
| 18:00.35 | flujan | zoa, your calls are analogic/IP our just IP? |
| 18:00.57 | flujan | cause here, we will work with both... and we should encode / decode it. |
| 18:01.07 | ManxPowe | zoa is too smart to have analog ports |
| 18:01.26 | MrChimpy | Tried to load 00000082 into 0000000a, but got 0000006f instead - i'm getting a lot of these to syslog from TE411P |
| 18:01.29 | *** join/#asterisk jjhall (n=chatzill@94-253.69-92-cpe.cableone.net) |
| 18:01.36 | *** join/#asterisk tomas_ (n=tomas@78.121.broadband3.iol.cz) |
| 18:01.37 | *** join/#asterisk Syrus_ (n=pascal@tahiti.mpl.rullier.net) |
| 18:01.37 | flujan | zoa, and which digium cards did you recommend? Here we have two E1 links |
| 18:03.07 | *** join/#asterisk Defraz (n=t0tal@72.165.56.43) |
| 18:04.00 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 18:04.13 | lahaine | <PROTECTED> |
| 18:05.00 | *** join/#asterisk coppice (n=chatzill@204.206.17.210.dyn.pacific.net.hk) |
| 18:05.10 | zoa | flujan: iax2 and zaptel (te410p) |
| 18:05.42 | zoa | guys, i just uploaded the mac version of iDEFISK, go get it |
| 18:05.46 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 18:06.04 | markit | hi :) what is the license of the set of sounds of asterisk? |
| 18:06.17 | zoa | gpl |
| 18:06.22 | zoa | + commercial |
| 18:06.40 | zoa | + some music on hold with another license |
| 18:07.42 | flujan | zoa, please do you just do VOIP calls ou analog ones too? |
| 18:07.51 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 18:08.02 | markit | zoa: ok, so for a translated set of sounds, (the sounds in the sound.txt), GPL would be fine for asterisk users, except for digium (no commercial possible), correct? |
| 18:08.16 | MrChimpy | hmm. |
| 18:08.50 | *** join/#asterisk _Sam-- (n=sam@mail.kneedraggers.com) |
| 18:09.14 | MrChimpy | acpi even |
| 18:09.53 | markit | in any case, maybe GNU FDL is more apropriate (like for documentation) |
| 18:11.43 | zoa | Markit they will not go in there unless you disclaim then |
| 18:12.07 | markit | zoa: I'm not interested in being included, just people will download elsewere |
| 18:12.20 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 18:12.54 | markit | zoa: but the main problem is if a GPL license is apropriate for sounds, since is not code, or if FDL is better, or if there is a even better license |
| 18:14.15 | pimpwell | anyone from NY? |
| 18:15.00 | *** join/#asterisk wasim (n=wasim@pdpc/supporter/active/wasim) |
| 18:15.08 | *** join/#asterisk svenna_ (n=svenna@p548D0DBE.dip0.t-ipconnect.de) |
| 18:15.17 | svenna_ | hi again :-) |
| 18:15.48 | svenna_ | i ve got a lucky problem now |
| 18:16.02 | svenna_ | (lucky because everything works) :) |
| 18:16.14 | svenna_ | ok, was isnt working: |
| 18:16.43 | svenna_ | when i lift up a phone, i get conected to the "s" extension |
| 18:16.52 | svenna_ | thats, how it should work |
| 18:16.58 | svenna_ | but i cant dial out then |
| 18:17.09 | *** part/#asterisk tomas_ (n=tomas@78.121.broadband3.iol.cz) |
| 18:17.16 | svenna_ | i have to dial a number and then lift up |
| 18:17.19 | *** join/#asterisk FastJack (i=fastjack@p5091E315.dip.t-dialin.net) |
| 18:17.56 | svenna_ | i know that, and its ok - my fax machine doesnt bother - and cant dial out :-( |
| 18:18.16 | [TK]D-Fender | svenna_ : What kind of phone? |
| 18:18.24 | svenna_ | i know its just a little problem between my ears... |
| 18:18.44 | svenna_ | hi [TK]D-Fender ! |
| 18:18.55 | Flauto | it sounds like dtmf problem |
| 18:19.01 | svenna_ | its a isdn pbx on a bri zap channel... |
| 18:19.19 | Flauto | then, i dotn' know |
| 18:19.20 | Flauto | hehe |
| 18:19.25 | svenna_ | :-) |
| 18:19.52 | flujan | ping zoa |
| 18:20.02 | iq | Hi, I'm trying to install TE110P on my Asterisk/CentOS machine. modprobe wct1xxp I get following erro: FATAL: Module wct1xxp not found. is module not installed or something else is wrong here? |
| 18:20.18 | *** join/#asterisk kll (i=kll@insomnia.juniks.net) |
| 18:20.34 | Juggie | iq, README.udev |
| 18:20.35 | svenna_ | when i do exten => s,1,Answer - i just get the tone and cant dial out... |
| 18:20.39 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 18:20.43 | kll | how do I set callerpres to hidden via RealTime? |
| 18:21.03 | iq | Juggie: I tried insmod but it failed as well. Still need to read README.udev? |
| 18:21.03 | *** join/#asterisk qw3rty (n=qw3rty@c-67-167-79-57.hsd1.il.comcast.net) |
| 18:21.04 | Juggie | yes |
| 18:21.13 | iq | Juggie: okay, will do that. Thanks |
| 18:21.14 | Juggie | follow the instructions in README.udev |
| 18:21.29 | flujan | We have analogic and digital phones... Do you think asterisk can handle about 1000 calls a day without problems? |
| 18:21.42 | qw3rty | Is there a way to sync or forward LookupBlacklist requests to an external database? |
| 18:21.57 | [TK]D-Fender | svenna_ : BRI? Ok, can't help there.. think I've been here before... |
| 18:22.25 | [TK]D-Fender | svenl : Set "immediate=no" in zapata.conf |
| 18:22.36 | [TK]D-Fender | flujan : Sure |
| 18:22.49 | *** join/#asterisk tank10 (n=tank10_c@netblock-72-25-92-13.dslextreme.com) |
| 18:23.04 | svenna_ | :-) |
| 18:23.05 | tank10 | ? for someone |
| 18:23.14 | svenna_ | ok, thx again :-) |
| 18:23.23 | svenna_ | i go an google a little more... |
| 18:23.46 | flujan | [TK]D-Fender, zoa says that he does something about 2 million calls a day using 4 dual xeons... |
| 18:24.00 | tank10 | * box works fine internaly making and reciving calls, works fine when an outbound call comes in. But when they call outbound there is no voice unless the put the caller on hold than take them off hold. |
| 18:24.14 | flujan | [TK]D-Fender, but i think that it is just IP. Not analog and IP calls... :D |
| 18:24.16 | [TK]D-Fender | flujan : # call/day isn't the proble, its # of CONCURRENT CALLS that impacts things... |
| 18:24.25 | tank10 | not natted |
| 18:24.31 | jpablo | hey people, i need some help |
| 18:24.37 | tainted_ | anyone have problems with grandstream ATAs? |
| 18:24.52 | flujan | [TK]D-Fender, ok ... How many concurrent calls asterisk can handle? |
| 18:24.53 | [TK]D-Fender | flujan : And yes channel type can come into play. it depends on a lot of things. |
| 18:25.02 | tainted_ | i get calls that cut out after about 20 minutes |
| 18:25.04 | jpablo | the guy in my telco tells me that i need to set the type of my number to unknow, that im currently sending nationl, im using a digium e1 card with isdn, any ideas ?> |
| 18:25.31 | flujan | [TK]D-Fender, Here we use two E1 links. And about 500 CONCURRENT calls in some moments. |
| 18:25.40 | [TK]D-Fender | flujan : Again, it depends on a lot of things. Straight VoIP with no transcoding puts little load on *, just network traffic. Transcoding really cuts into CPU power, PSTN adds a bit as well.. |
| 18:25.52 | ManxPowe | unknown is correct more of the time. |
| 18:26.03 | flujan | [TK]D-Fender, yes we will use transconding... |
| 18:26.10 | jpablo | ManxPowe, how i do se it ? |
| 18:26.20 | [TK]D-Fender | 2 x E1 = 60 channels +/-. NO PROBLEM. If the rest of those 500 calls is just RTP passing between internal phones, then no big deal. |
| 18:26.26 | ManxPowe | pridialplan=unknown |
| 18:26.29 | kll | how do I set callerpres to hidden in a SIP Realtime setup? |
| 18:26.32 | ManxPowe | just like it says in the .sample config file |
| 18:26.53 | flujan | [TK]D-Fender, i need some guindance since a consulting company just said its impossible to asterisk handle this. |
| 18:27.25 | tank10 | [TK]D-Fender I was wondering how that was pulled off. 500 on two E1's. lol |
| 18:27.57 | tank10 | flujan of course they did they want to charge you to use cisco call manager or something useless like that |
| 18:28.02 | jpablo | ManxPowe, thanks, it worked :D |
| 18:28.21 | jpablo | ManxPowe, sorry, i deleted the example file and pasted some lines i found online |
| 18:28.24 | tank10 | can someone please assist me with my small issue |
| 18:28.26 | ManxPowe | jpablo, there is a reason the sample config file says "you almost never need to use this option" |
| 18:28.31 | [TK]D-Fender | tank10 : No-one said that the 500 calls were going OVER the 2 E1. |
| 18:28.42 | tank10 | [TK]D-Fender LOL tis true |
| 18:29.00 | tank10 | [TK]D-Fender be a neat trick though talk about compression! |
| 18:29.19 | [TK]D-Fender | flujan : Make sure your answers are right, because if * is just passing RTP for phone-phone calls internally then it isn't processing that much. its transcoding / PSTN that really adds to the load |
| 18:29.42 | iCEBrkr | docelm0: LOL! My boss found the Tampa PHP group |
| 18:29.57 | MrChimpy | thrills. I get a yellow alarm on loopback. |
| 18:29.58 | [TK]D-Fender | tank10 : If compression was involved, that would kill the CPU and I couldn't say "yes" :) The fine-print is EVERYTHING |
| 18:30.06 | tank10 | lol |
| 18:30.10 | Zodiacal | anyone know if theres a way to keep my cisco 7960g phone from causing lost packets when its unpluged? |
| 18:30.17 | Zodiacal | it acts as a switch |
| 18:30.19 | [TK]D-Fender | rob0 : Glad to help... what did I do fro you again? :) |
| 18:30.24 | iCEBrkr | rob0: There ya go.. Million dollar idea. Instead of PayPal, it's PayBeer :P |
| 18:30.28 | Zodiacal | but if unpluged it causes havic |
| 18:30.38 | tank10 | [TK]D-Fender When you have a moment i do have a quit complexing problem |
| 18:30.49 | rob0 | Told me to look in /proc/interrupts ... we found that there was a shared IRQ. |
| 18:30.51 | [TK]D-Fender | iCEBrkr : I prefer PayPal personally :) I'm not much of a drinkier... |
| 18:31.02 | iCEBrkr | lol |
| 18:31.06 | [TK]D-Fender | rob0 : Oh, and did you get everything out of the way and is it working better now? |
| 18:31.07 | tainted_ | waht does this mean: Received VNAK: resending outstanding frames |
| 18:31.24 | rob0 | it's fine now with the USB driver unloaded |
| 18:31.37 | fugitivo | [TK]D-Fender: nothing like watching in your monitor how your paypal account gets bigger and bigger |
| 18:32.10 | [TK]D-Fender | fugitivo : I've had 2 * payoffs to date. I give a lot out for free but a few bucks is always appreciated... |
| 18:32.28 | flujan | [TK]D-Fender, we will have something about 500 calls running in concurrency. Externals phones... I'm working on a call center. and we will buy another E1 channels... |
| 18:32.29 | *** join/#asterisk iq (n=iq@71-38-74-41.omah.qwest.net) |
| 18:32.48 | tainted_ | can someone help me with calls that are cut short? |
| 18:33.01 | flujan | [TK]D-Fender, which is the best protocol to use? To minimize the transconding... And Can audiocodecs help in my env? |
| 18:33.11 | iq | Juggie: I read readme.udev followed instructions. Changed my udev conf files. But still dont see any zap device in /dev |
| 18:33.42 | [TK]D-Fender | flujan : inside of your LAN, G.711u |
| 18:34.18 | Zodiacal | any ideas? |
| 18:34.22 | [TK]D-Fender | flujan : Actually since you're in EU it'd be G.711a = free and the "native" format of audio in your E1 links |
| 18:35.14 | flujan | [TK]D-Fender, would you recomend audiocodecs? |
| 18:35.33 | [TK]D-Fender | flujan "audiocodecs" ? |
| 18:35.36 | [TK]D-Fender | huh? |
| 18:35.46 | *** join/#asterisk aster (n=junk@59.93.68.52) |
| 18:36.20 | aster | please, anybody suggest good termination providers |
| 18:36.40 | *** join/#asterisk Prival (n=someone@209-161-233-37.dsl.look.ca) |
| 18:36.44 | tank10 | if you don't need fax broadvoice seems to do a decent job for one of my clients |
| 18:37.39 | aster | we need in the order of 15,000 mins per month; would you suggest them? |
| 18:37.53 | Prival | Anyone knows if from the TDM400P pci configuraton space I can find out if FXO or FXS modules are installed and on which ports? |
| 18:38.00 | Juggie | iq, the device wont exist until you load the module |
| 18:38.36 | *** join/#asterisk apardo (n=apardo@87.218.44.151) |
| 18:38.40 | *** join/#asterisk Zodiacal- (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:39.25 | _Sam-- | broadvoice doesnt do any trunking, so if your 15,000 minutes are incoming it will probably be a pain. |
| 18:40.06 | _Sam-- | if you needed 15,000 outgoing it would probably work if you had a bunch of accounts, which would also be a pain |
| 18:40.13 | *** join/#asterisk darkskiez (n=darkskie@bb-87-81-62-203.ukonline.co.uk) |
| 18:40.28 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 18:42.11 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 18:43.27 | Prival | anyone can share his/her lspci -xxx of a configuration using the TDM400P and give me it's module configuration? |
| 18:43.34 | rob0 | offtopic, but so what ... :) I am going to build a new server. Have decided I want socket 939 (AMD 64). Any favorite server motherboards? |
| 18:43.42 | rob0 | Prival: sure, just a min. |
| 18:43.44 | Qwell[] | rob0: tyan |
| 18:43.46 | Qwell[] | always good |
| 18:44.04 | flujan | [TK]D-Fender, a machine that decode/encode audio. :D |
| 18:44.04 | Prival | rob0: Thanks, what config FXO/FXS do you have? |
| 18:44.12 | rob0 | single fxs |
| 18:44.14 | Qwell[] | and if you're going to get a 939 for a server...you should really look at the new 939 opterons...they're hot |
| 18:44.23 | Qwell[] | (pun accidental) |
| 18:44.28 | rob0 | ok thx Qwell[] :) |
| 18:44.40 | Qwell[] | rob0: 170+, dual core...mmm |
| 18:45.43 | De_Mon | if my dialplan has s,n,WaitExten how long will it wait? |
| 18:46.02 | Qwell[] | De_Mon: until the timeout period, just as though it were a new call, I believe |
| 18:47.17 | Qwell[] | De_Mon: You can also pass in a timeout to waitexten |
| 18:47.42 | rob0 | Prival: I posted this yesterday: http://pastebin.ca/37425 |
| 18:47.43 | Prival | rob0: do you want to e-mail it directy to me? |
| 18:47.51 | Prival | rob0: great thanks. |
| 18:47.53 | *** join/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 18:47.59 | De_Mon | hmm, I'll have to try it again, the timeout seemed loger than the set timeout, it would be response timeout, not digits timeout? |
| 18:48.11 | iq | Can anyone help me set up my first T1 card - its digium TE110P |
| 18:48.23 | De_Mon | digits timeout is for after the first # is received |
| 18:48.30 | *** join/#asterisk TK9 (n=Miranda@p54B2A6F3.dip0.t-ipconnect.de) |
| 18:48.38 | }btorch{ | anyon eher ehas used or likes the grandstream sip phones ? |
| 18:49.06 | Qwell[] | }btorch{: no and no |
| 18:49.11 | MrChimpy | only problem I had configuring TDM400 was realising that I needed to do fxoks=1 and fxsks=4 according to the physical positions of the cards |
| 18:49.17 | rob0 | Qwell[]: I should add that I don't know much about modern hardware. I really just want something in the AMD 64 architecture, to get out of ix86 (which is all I've ever had.) Someone recommended 939 to me. |
| 18:49.18 | Qwell[] | They're pretty much junk... |
| 18:49.24 | iq | when I do "modprobe wct1xxp" I get this messag "Module wct1xxp not found". I updated my udev .conf files. But still dont see any device in /dev |
| 18:49.36 | Qwell[] | rob0: if you want amd, you really have 2 choices. 939 and 940 |
| 18:49.42 | justinu | iq: that problem has nothing to do with /dev |
| 18:49.43 | MrChimpy | rather than it just doing them sequentially - my card turned up with modules in 1 and 4 but all the sample configs were 1 and 2 |
| 18:49.44 | justinu | or udev |
| 18:49.51 | Qwell[] | 939 is athlon64, and (now) dualcore single config opterons |
| 18:50.04 | iq | justinu: could you give me a starting point please |
| 18:50.07 | Qwell[] | 940 is mostly 2+ opterons, and I think it might include a few athlons |
| 18:50.16 | *** join/#asterisk ghento2 (i=ghento@CPE0011d8a291a6-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 18:50.24 | rob0 | ok that helps Qwell[], thx again. |
| 18:50.36 | Prival | rob0: I need the pci configuration space also: lspci -xxx -s 00:09.0 |
| 18:50.44 | Qwell[] | rob0: I would check out newegg, and go through their CPU list. They have a good "search" tool |
| 18:50.45 | rob0 | Prival: ok |
| 18:50.45 | De_Mon | is 64bit hardware worth it yet |
| 18:50.52 | Qwell[] | De_Mon: totally |
| 18:50.54 | ghento2 | Hi folks. I'm using Record() to record voice messages into wav files. Which conf file do I go to edit the voice quality? I want to increase it |
| 18:51.08 | justinu | iq: how about just modprobe zaptel? |
| 18:51.08 | Qwell[] | (and their prices aren't bad, to boot) |
| 18:51.22 | De_Mon | ^_^ I need to upgrade an old thunderbird, I'll have to look into it closer. |
| 18:51.27 | Qwell[] | newegg should pay me, I swear...I'm a damn good spokesperson |
| 18:51.28 | iq | justinu: FATAL: Module zaptel not found. |
| 18:51.39 | justinu | iq: did you compile and install zaptel ? |
| 18:51.42 | *** join/#asterisk ke4qqq (n=chatzill@68-115-212-155.static.spbg.sc.charter.com) |
| 18:51.56 | Qwell[] | I applied for a job at newegg...didn't get a response...bastards |
| 18:52.02 | iq | justinu: it is installed. I can see all .ko module files under kernel folder |
| 18:52.18 | De_Mon | I use googlegear almost as often as newegg |
| 18:52.26 | rob0 | Prival: http://pastebin.ca/37616 |
| 18:52.33 | justinu | iq: hmm, some kind of problem with the running kernel vs. kernel sources installed, maybe. |
| 18:52.35 | Qwell[] | didn't googlegear change names like...2 years ago? |
| 18:52.39 | Qwell[] | zipzapzoom? |
| 18:52.45 | justinu | zipzoomfly |
| 18:52.47 | tuxinator_linux | zipzoomfly |
| 18:52.48 | Qwell[] | whatever :p |
| 18:52.58 | tuxinator_linux | I like newegg also |
| 18:53.12 | MrChimpy | do E1s usually use HDB3 framing or is AMI reasonably common |
| 18:53.16 | Qwell[] | if I buy something from newegg, if I do ground shipping, I can guarantee it'll be in my hands the next morning |
| 18:53.19 | iq | justinu: reinstalling might help? Actually, on this specific machine I used asterisk@home. Did not compile myself |
| 18:53.25 | MrChimpy | ? |
| 18:53.56 | justinu | iq: yeah - bad call w/ a@h |
| 18:53.56 | jbalcomb | ok, so if I set my phone to Send DTMF via 'rfc2833' and my entry in the sip.conf to dtmfmode=rfc2833 then that works for just my phone? |
| 18:54.06 | De_Mon | oh yeah.. I just goto googleglear and follow the link :P |
| 18:54.09 | justinu | iq: you could try downloading the zaptel source and build it yourself |
| 18:54.21 | jbalcomb | and all the other phones can still be set to DTMF via SIP INFO and dtmfmode=info? |
| 18:54.29 | iq | justinu: oh you think so... shall I rebuild the machine with CentOS and compile everything manually? |
| 18:54.37 | Prival | rob0: Many bytes differ... I'll need to get more samples of different config to figure it out. Thanks. |
| 18:54.45 | Qwell[] | jbalcomb: Yes, they must always match |
| 18:54.46 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 18:54.47 | justinu | iq: i would do that. |
| 18:54.49 | Prival | anyone can share his/her lspci -xxx of a configuration using the TDM400P and give me it's module configuration? |
| 18:54.59 | Qwell[] | Prival: It shows up as tigerjet |
| 18:55.05 | jbalcomb | Qwell but theres not trouble having some phones set to sip info and some to rfc2833? |
| 18:55.14 | iq | justinu: thanks... I will do that on monday I guess... :) |
| 18:55.16 | Qwell[] | jbalcomb: not really |
| 18:55.22 | jbalcomb | Qwell ok, thanks |
| 18:55.41 | Qwell[] | I mean...if they get reinvited to each other, then yeah, maybe...but a phone won't understand dtmf anyhow |
| 18:55.48 | justinu | iq: i run centos 4.2, and everything build and installs fine |
| 18:55.55 | *** join/#asterisk Cresl1n (n=matt@gateway.digium.com) |
| 18:55.59 | *** part/#asterisk TK9 (n=Miranda@p54B2A6F3.dip0.t-ipconnect.de) |
| 18:56.24 | Prival | Qwell: I need the configuration space bytes 00 through ff |
| 18:56.30 | Qwell[] | eh? |
| 18:56.48 | iq | justinu: i c. So I'll install latest stable CentOS and then get subversion of Asterisk source and compile myself. Is there anything else I need to take care of while doing that? Shall I leave the T1 card inside or remove it ? |
| 18:57.03 | justinu | iq: i recommend going with the latest stable asterisk tarballs... 1.2.2 |
| 18:57.06 | Qwell[] | iq leave it in |
| 18:57.09 | justinu | leave your t1 card in |
| 18:57.09 | *** join/#asterisk wizard545 (n=wizard@tor/session/x-cd59cdce9845d479) |
| 18:57.22 | Prival | Qwell: do lspci -xxx and you will see a bunch of hex bytes. This is known as the PCI configuration space for each devuce |
| 18:57.25 | justinu | get asterisk-1.2.2.tar.gz, zaptel-1.2.2.tar.gz |
| 18:57.31 | Qwell[] | Prival: yes, but why? |
| 18:57.33 | iq | justinu: Qwell[]: okay, I'll do that. Thanks :) |
| 18:57.45 | Qwell[] | you don't need it to do the config |
| 18:57.46 | justinu | iq: good luck |
| 18:57.53 | justinu | man, these GXP2000s are retarded |
| 18:57.58 | iq | justinu: ya - need lots of it ;) ...thanks again |
| 18:58.05 | Prival | Wqell: I need to find a way to identify which modules (FXO or FXS) and on which ports by software |
| 18:58.08 | justinu | they DHCP for an address, but they won't get the tftp server from DHCP!!! |
| 18:58.12 | justinu | wtf? |
| 18:58.24 | wizard545 | which is the best to use out of this list... Vocoder: g729, g711 ulaw, g726-32. |
| 18:58.25 | Qwell[] | Prival: it doesn't show the ports, only the main board |
| 18:58.54 | Qwell[] | you could probably use one of the zt* programs to find out |
| 18:59.10 | Qwell[] | wizard545: depends |
| 18:59.14 | justinu | prival: cat /proc/zaptel/1 |
| 18:59.18 | Prival | It might in the configuration space. For exemple the 1st 4 bytes of the configuration space is the manufacturer id and device id: 59 e1 01 00 = e159:0001 |
| 18:59.20 | Qwell[] | or that |
| 18:59.30 | wizard545 | i need fast, ok quality |
| 18:59.34 | Qwell[] | Prival: it doesn't change when you change modules |
| 18:59.46 | Qwell[] | wizard545: 80k per channel okay? |
| 19:00.06 | Prival | justinu: that find after zaptel has loaded. I need to do this during our OS install in order to automatically do a pre-configuration of zaptel... |
| 19:00.07 | wizard545 | ... rather it be a lot less |
| 19:00.13 | Qwell[] | Then use g729 |
| 19:00.17 | wizard545 | this is for a large concurrent setup |
| 19:00.20 | justinu | wizard545: g711 if you have bandwidth and want toll quality, otherwise g729 |
| 19:00.28 | justinu | prival: ahh |
| 19:00.31 | Qwell[] | with a log of calls, transcoding g729 is expensive |
| 19:00.35 | Qwell[] | lot* |
| 19:00.43 | De_Mon | what about gsm? |
| 19:00.47 | justinu | Prival: i think you'll have to talk with the guys who designed the digium cards |
| 19:00.55 | Prival | Qwell: you know that for a fact? menufacturer id and device id won't change, the the other 508 bytes might... |
| 19:01.06 | justinu | prival: or look at the zaptel kernel module code and figure out how they inventory the hardware. |
| 19:01.16 | wizard545 | on a p3 server.. on a 100meg line (datacenter) how many calls can I concurrently run? |
| 19:01.21 | wizard545 | anyone got a ballpark |
| 19:01.23 | Prival | justinu: just e-mailed them... |
| 19:01.27 | Qwell[] | wizard545: a lot more if you do g711 |
| 19:01.29 | justinu | wizard545: 30-40 g729 calls |
| 19:01.40 | Qwell[] | 30-40? on a p3? |
| 19:01.44 | justinu | wizard545: if you're transcoding |
| 19:01.48 | De_Mon | a p3 what? |
| 19:01.55 | Qwell[] | 350, heh |
| 19:01.56 | De_Mon | 250mhz p3? |
| 19:01.57 | justinu | oh, i had a dual p3-600 benchmarked out at 40 |
| 19:02.00 | wizard545 | 1ghz |
| 19:02.10 | Qwell[] | alright, so maybe 30-40 then |
| 19:02.14 | Qwell[] | still more with ulaw |
| 19:02.23 | justinu | yeah, 100+ w/ ulaw |
| 19:02.25 | wizard545 | on a p4 2ghz.. could be a lot more? |
| 19:02.32 | wizard545 | ulaw? |
| 19:02.38 | justinu | wizard545: i get about 80 g729 channels on a 3.0 xeon |
| 19:02.38 | Qwell[] | ulaw/g711u |
| 19:02.51 | wizard545 | nice |
| 19:02.51 | De_Mon | what should my codec order be? I thought gsm was the better solution |
| 19:03.00 | wizard545 | so this isn't gonna break the bank.. rock on |
| 19:03.00 | Qwell[] | De_Mon: gsm sounds icky |
| 19:03.19 | De_Mon | ulaw, alaw, gsm ? |
| 19:03.21 | wizard545 | i work at a datacenter... we have boxes laying around everywhere |
| 19:03.26 | Qwell[] | no need for ulaw and alaw |
| 19:03.35 | Qwell[] | I mean, not both |
| 19:04.03 | rob0 | Qwell[]: chipset preference for a good server board? |
| 19:04.20 | Qwell[] | rob0: got me... |
| 19:04.27 | wizard545 | another quick question.. we bought a port from telesip.. it never rings busy using the same DID with like 6 calls, is that normal? i thought one port could only handle one or two calls |
| 19:04.28 | brad_mssw | gsm isn't too bad ... just most devices don't support gsm |
| 19:04.39 | justinu | gsm is sucks |
| 19:04.46 | justinu | i hate my cell phone |
| 19:04.47 | rob0 | ok thx again |
| 19:05.03 | Qwell[] | wizard545: most per minute providers will let you have an infinite number of simultaneous calls |
| 19:05.17 | wizard545 | gotcha |
| 19:05.38 | justinu | you can make it "ring busy" if you want :P |
| 19:05.48 | wizard545 | haha |
| 19:05.55 | wizard545 | i'm on the "unlimited plan" |
| 19:06.03 | wizard545 | no per minute, which is cool |
| 19:06.13 | Qwell[] | Then they're probably charging you extra for each extra simultaneous call |
| 19:06.24 | wizard545 | i'll check haha |
| 19:06.27 | Qwell[] | and... |
| 19:06.29 | Qwell[] | ~unlimited |
| 19:06.34 | jbot | hmm... unlimited is <Nugget> unlimited voip == punch the monkey to win a free ipod |
| 19:06.34 | justinu | heh |
| 19:06.45 | wizard545 | gotcha |
| 19:06.50 | justinu | yeah, be careful |
| 19:07.07 | Qwell[] | brb |
| 19:07.15 | justinu | wizard545; try a prepaid account, something like junctionnetworks, asterlink, voicepulse, etc. |
| 19:07.56 | wizard545 | it's for a calling card business |
| 19:08.18 | file | you are not going to run a calling card business of an unlimited plan |
| 19:08.21 | justinu | well... chances are you'll get burned by the SIP provider if you go over their idea of what unlimited is |
| 19:08.33 | wizard545 | yea |
| 19:08.35 | brad_mssw | justinu: btw, i got a response from junctionnetworks about the outage ... did you? |
| 19:08.41 | file | they will probably fire you |
| 19:08.44 | justinu | brad_mssw: nope! they don't like me |
| 19:09.20 | Zodiacal- | on the cisco 7960g's lcd, are the 6 buttons on the right for real phone lines or for internal extention's? |
| 19:09.31 | brad_mssw | justinu: "There was a major MPLS conflict between Level 3 and WCG (recently purchased by Level 3). It was resolved at around 1pm (EST). We are planning a major capital outlay to fully upgrade our network over the next 3-4 weeks. One of the enhancements will be to acquire secondary and tertiary carriers." |
| 19:09.35 | justinu | they're "line appearances" |
| 19:09.53 | Zodiacal- | justinu ok that confuses me. in the dox they say lines.. but im not sure which |
| 19:10.08 | Zodiacal- | justinu does that mean they can be either? |
| 19:10.09 | justinu | brad_mssw: interesting |
| 19:10.27 | justinu | Zodiacal: not sure about cisco specifically, but on polycom yes. |
| 19:10.53 | pigpen | I have a business partner that is trying to convince me that Asterisk@Home can be used in the corporate environment....Could someone please help me either way. Anything@Home I am not crazy about. |
| 19:11.10 | Zodiacal- | justinu okie i guess i need to read more.. they seem to act like external lines, but i would rather them be extentions.. |
| 19:11.21 | file | pigpen: well, the chance of you getting help on that in here is very slim, so that shows you how we feel :) |
| 19:11.35 | file | I won't even help with A@H for money. |
| 19:11.50 | pigpen | file: yeah, that is what I figured...nor would I... |
| 19:11.56 | justinu | lol |
| 19:12.13 | jbalcomb | [TK]D-Fender We have a 'Wildcard TE411P' .. is that decent for EC? |
| 19:12.37 | tuxinator_linux | jbalcomb: EC? |
| 19:12.50 | pigpen | So, better off just loading my own and doing AMP, etc... ? (for those that -must- have a GUI) |
| 19:12.53 | tainted_ | can anyone help me with calls that get cut short? |
| 19:12.58 | justinu | pigpen: probably |
| 19:13.06 | pigpen | k |
| 19:13.10 | jbalcomb | tuxinator_linux echo cancellation |
| 19:13.13 | Qwell[] | file: What if I offered you muffins? |
| 19:13.19 | pigpen | Thanks all... |
| 19:13.23 | file | Qwell[]: I'd type rm -rf / :P |
| 19:13.25 | De_Mon | most soft phones support G.711, gsm looks like thats the order I want them in |
| 19:13.50 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 19:15.16 | *** join/#asterisk Billy (i=Billy@pcp03947165pcs.brghtn01.mi.comcast.net) |
| 19:15.58 | aster | hello - I am trying to transcode between gsm and ulaw. I allowed only gsm in iax.conf and allow=ulaw in my sip conf |
| 19:16.22 | aster | how can I make sure that the transcoding is taking place. Any comment to check at CLI? |
| 19:16.34 | Billy | Having an issue with tdm2400 install - anyone able to help? |
| 19:16.35 | Qwell[] | file: rm: cannot remove `:P': No such file or directory |
| 19:16.45 | tzanger | Qwell[]: heh |
| 19:16.55 | justinu | ugh, analog problems |
| 19:16.57 | justinu | pita! |
| 19:16.59 | wizard545 | is 1 penny a minute high? |
| 19:17.05 | justinu | no |
| 19:17.10 | Qwell[] | wizard545: No, but it'll be a crap provider |
| 19:17.24 | Qwell[] | cheap, reliable, good customer service...pick up to 2 |
| 19:17.38 | file | Qwell[]: evil! |
| 19:17.39 | Qwell[] | (if you get below a certain price, you choose cheap twice) |
| 19:17.50 | wizard545 | gotcha |
| 19:18.13 | brad_mssw | Qwell[]: ok, I choose reliable and good customer service ... who is that ? |
| 19:18.22 | Qwell[] | 4) has a dns server that doesn't hate you |
| 19:18.31 | Qwell[] | brad_mssw: asterlink/nufone |
| 19:18.46 | brad_mssw | Qwell[]: asterlink's website doesn't even come up for me |
| 19:18.55 | Qwell[] | brad_mssw: See above :P |
| 19:18.56 | brad_mssw | Qwell[]: nufone only offer's michigan phone numbers |
| 19:19.04 | Qwell[] | they both offer 8xx dids |
| 19:19.32 | brad_mssw | Qwell[]: yeah, still need to move over some 321 and 352 did's though :/ though I guess those could be hosted elsewhere |
| 19:19.54 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 19:21.00 | Qwell[] | file: status of CA dids? |
| 19:21.06 | Qwell[] | my CA, not yours |
| 19:21.31 | iCEBrkr | <PROTECTED> |
| 19:21.38 | pigpen | anyone know how to tell what clec owns a particular npa-nxx ? |
| 19:21.40 | iCEBrkr | Is that anything I can fix? or even have to worry about? |
| 19:21.50 | brad_mssw | what is the address of nufone's IAX server? want to run a latency test |
| 19:22.06 | Qwell[] | dunno, not at home |
| 19:22.08 | iCEBrkr | brad_mssw: It's not nice to run 'ping -f' |
| 19:22.22 | MrChimpy | aaargh |
| 19:22.32 | MrChimpy | feckin 411P ain't talking |
| 19:23.11 | brad_mssw | iCEBrkr: was thinking more along the lines of traceroute to see how far away they are |
| 19:23.54 | *** join/#asterisk pardove (n=pardove@195.146.47.239) |
| 19:23.56 | pifiu | can someone tell me how to setup IAX in between 2 servers so they talk to each other? |
| 19:24.56 | pardove | has anybody used spandsp-0.0.2pre22? |
| 19:25.43 | wizard545 | how many companies do the actual termination? like voip to pstn |
| 19:25.44 | caio1982 | me |
| 19:25.57 | pardove | has anybody used spandsp-0.0.2pre22? |
| 19:25.58 | Billy | pardove: I tried once. |
| 19:26.09 | caio1982 | pardove: yes |
| 19:26.10 | pardove | Billy: did u have any success? |
| 19:26.25 | Billy | Couldn't compile it. |
| 19:26.46 | Billy | I don't think it liked my AMD 64 |
| 19:27.00 | pardove | i've compiled it but cant get any fax!!!! |
| 19:27.17 | Ariel_ | pifiu, http://www.voip-info.org/wiki/view/Asterisk+-+dual+servers |
| 19:27.48 | *** join/#asterisk svenna_ (n=svenna@p548D399D.dip0.t-ipconnect.de) |
| 19:27.52 | *** join/#asterisk ManxPowe (n=ewieling@dpc6745150107.direcpc.com) |
| 19:28.28 | Billy | I'm lookign for my notes on Asterisk config for fax |
| 19:28.38 | Katty | Billy: did you look under the bed? |
| 19:29.10 | pardove | Billy: downgrading to spandsp-0.0.2pre21 solves the problem. |
| 19:29.23 | Billy | k |
| 19:29.46 | beebz | is there a way to loop a playback() until someone picks up an extension? |
| 19:30.00 | Billy | Katty: I'm afraid to look under my bed. |
| 19:30.06 | brad_mssw | anyone have nufone's iax server address so I can traceroute it ? |
| 19:30.42 | pardove | btw, has anybody made spandsp-0.0.2pre22 to work? |
| 19:30.46 | coppice | spandsp and the fax apps work just fine on AMD64 machines |
| 19:30.54 | Katty | Billy: )= |
| 19:31.13 | pardove | i'm talking about it's new version: "spandsp-0.0.2pre22" |
| 19:31.31 | Samoied | hello |
| 19:31.34 | Ariel_ | brad_mssw, switch-1.nufone.net |
| 19:31.50 | brad_mssw | Ariel_: thanks |
| 19:31.50 | Samoied | Anyone have the BETATEST firmware for Grandstream HT488? |
| 19:32.04 | Qwell[] | ugh |
| 19:32.08 | Samoied | The directory in /BETATEST/ is empty! |
| 19:32.15 | Qwell[] | it isn't enough to use crap phones...now it's crap phones with crap firmware |
| 19:32.17 | brad_mssw | pardove: doesn't work for me ... even if I do it 100% on my local network |
| 19:32.34 | _Sam-- | the gxp2000s arent really that bad! |
| 19:32.56 | Qwell[] | _Sam--: Do they have a speaker phone? |
| 19:32.59 | _Sam-- | of course |
| 19:33.08 | Qwell[] | one that isn't complete junk? |
| 19:33.08 | pifiu | what are some good dependable IP phones that are not expensive? |
| 19:33.09 | pardove | brad_mssw: i've the same problem. fax tone is listened but it fails all the time :( |
| 19:33.13 | pifiu | for my grandmother |
| 19:33.13 | pifiu | lol |
| 19:33.14 | Billy | pardove: I had problems with pre20. haven't tried latest - sorry. |
| 19:33.20 | Qwell[] | pifiu: I hear the spa941 is good |
| 19:33.23 | pifiu | need to give me one to her and one to my parents |
| 19:33.27 | brad_mssw | pardove: yeah, it acts like it's doing something ... but nothing works |
| 19:33.27 | pifiu | something like stupid easy to use |
| 19:33.34 | _Sam-- | qwell...i guess its more a matter of what you compare them to. its an 85 dollar phone with 200 dollar features |
| 19:33.44 | _Sam-- | and in my mind, it blows away the 941 twice the money |
| 19:33.50 | Qwell[] | please |
| 19:33.50 | brad_mssw | Qwell[]: yeah, got a 941 here ... much nicer than I expected |
| 19:33.53 | _Sam-- | the 941 comes with 2 lines, pay extra for another 2 |
| 19:33.54 | Billy | Anyone help with tdm2400 problem? |
| 19:34.01 | Qwell[] | Billy only if you ask |
| 19:34.04 | *** join/#asterisk Wipe (n=louis_el@65.94.0.98) |
| 19:34.05 | pardove | we must find steveu now |
| 19:34.08 | justinu | _Sam--: you run GXPs? |
| 19:34.18 | _Sam-- | i do...not hundreds, but dozens. |
| 19:34.26 | [av]bani | gxp2000 is suprising for $80 |
| 19:34.31 | brad_mssw | pardove: he's in here as coppice |
| 19:34.32 | justinu | _Sam--: figured out TFTP provisioning? |
| 19:34.47 | brad_mssw | pardove: he'll probably kill me for telling you that though |
| 19:34.48 | Katty | tftp makes me all sad inside. |
| 19:34.54 | zoa | sam, the st302 is even better and cheaper |
| 19:34.54 | pardove | but he doesn't answer his mails ;( |
| 19:34.56 | Qwell[] | Katty: tftp doesn't like you |
| 19:35.00 | Katty | Qwell[]: i knew it :< |
| 19:35.10 | _Sam-- | i was going to use tftp for the last 12 i did, but even with tftp you cant fully configure everything from what i saw |
| 19:35.25 | Katty | tftp is like a teddybear without a human. |
| 19:35.25 | Billy | Installed tdm2400 with one FXO. modprobe works fine, all tests look good. asterisk starts fine, finds channels . . . |
| 19:35.32 | Qwell[] | _Sam--: yeah, that totally sounds like a $200 feature |
| 19:35.54 | pardove | can anybody make a contact with him, it's about 10days that he has put new buggy version in his site... |
| 19:35.54 | Billy | Dialing won't pick up line. |
| 19:36.03 | Samoied | anyone use Handytone with T.38? |
| 19:36.11 | Katty | dialing isn't a good pickup line anyway. |
| 19:36.16 | Billy | Dialing in won't answer. |
| 19:36.17 | Samoied | My HT488 dont detect Fax signal |
| 19:36.24 | tuxinator_linux | Katty, you're funny |
| 19:36.34 | Katty | tuxinator_linux: i'm feeling funny. |
| 19:36.49 | Katty | tuxinator_linux: might be all that caffeine i just drank. |
| 19:36.53 | justinu | _Sam--: the problem with TFTP I've found is you still have to punch the damn TFTP IP address into the phone! |
| 19:36.57 | Billy | ztmonitor sees dialing and voice on handset, but handset doesn't have any sound. |
| 19:36.59 | justinu | that's so retarded |
| 19:37.10 | Katty | let's not insult the challenged. |
| 19:37.14 | _Sam-- | justinu: if you have a hub you can configure your IP to one the phone is looking for |
| 19:37.16 | Qwell[] | handset? fxo? huh? |
| 19:37.17 | Katty | they needs teddybears too. |
| 19:37.19 | _Sam-- | and it will connect with no changes |
| 19:37.26 | Qwell[] | Billy: Surely you don't have phones plugged into your FXO? |
| 19:37.27 | justinu | _Sam--: hub? |
| 19:37.32 | _Sam-- | yeah a cheap ass hub |
| 19:37.35 | Billy | SIP handest that origniated the call through zap channel |
| 19:37.36 | _Sam-- | 10/100 hub |
| 19:37.40 | brad_mssw | ok, anyone have any input on the most reliable voip provider for business ? |
| 19:37.44 | _Sam-- | configure your box with the ip the phone is looking for |
| 19:37.47 | justinu | i don't get it... i have a PoE network |
| 19:37.47 | _Sam-- | plug it into the cheap ass hub |
| 19:37.52 | _Sam-- | plug the phone into the hub |
| 19:37.53 | Qwell[] | BillyL sounds like a SIP issue then |
| 19:37.58 | _Sam-- | turn it on, the phone will connect to your computer |
| 19:38.06 | _Sam-- | i used ethereal to dump the IP it was looking for |
| 19:38.08 | justinu | i just want the phones to DCHP and get a config like like the polycoms do! |
| 19:38.17 | *** join/#asterisk kjl (n=junk@59.93.68.52) |
| 19:38.26 | justinu | i shouldn't have to know anything other than the MAC of the phone |
| 19:38.45 | Qwell[] | mac addresses are for wussies |
| 19:38.46 | _Sam-- | how do the phones know where to get the config file from? |
| 19:38.50 | Billy | TDM400's work fine on same machine and setup. only difference is tdm2400 instead. |
| 19:38.51 | Qwell[] | XMLDefault.cnf.xml :p |
| 19:38.54 | Katty | twisted[asteria]: ping! |
| 19:38.59 | justinu | _Sam--: from the DHCP server! |
| 19:39.07 | pardove | can anybody make a contact with "steveu", it's about 10days that he has put new buggy version of spandsp (0.0.2pre22) in his site... |
| 19:39.13 | Qwell[] | pardove: coppice |
| 19:39.27 | Qwell[] | he just said it worked fine. it's you |
| 19:39.28 | *** join/#asterisk gaz00 (n=darren@68.144.64.211) |
| 19:39.33 | [av]bani | pardove: he's hiding from you. he said you're scary. |
| 19:39.36 | coppice | if pre22 proves buggy, use pre21 |
| 19:39.39 | *** join/#asterisk juanjoc (n=juanjoc@200.73.189.82) |
| 19:39.50 | [av]bani | http://bani.anime.net/phonez/ \o/ |
| 19:39.54 | pardove | pre21 works just fine |
| 19:40.02 | *** join/#asterisk basta (n=basta@194.150.162.129) |
| 19:40.20 | pardove | i've sent a mail to *-users regarding the problem with debug details |
| 19:40.54 | pardove | coppice: the fax tone is heard fine but no fax is recieved |
| 19:40.57 | Billy | Is there anything wierd/unique about the 25 pair connector? I'm using a standard telco 25 pair, connecting to the first four lines. |
| 19:41.10 | pardove | downgrading to -pre21 solves the problem |
| 19:41.23 | coppice | so use pre21 |
| 19:41.45 | juanjoc | Has anyone ever used the 'trustrpid' setting in sip.conf and verified that it works? |
| 19:42.02 | juanjoc | I mean, when setting trustrpid=yes |
| 19:42.07 | pardove | coppice: :-) i want to know if the bug is confirmed or not? |
| 19:42.25 | _Sam-- | justinu: im sorry to sound stupid (only because i am) but if you have 1 config file for dozens of phones, how do the phones get the SIP user info / details? |
| 19:42.40 | zoa | hey ho sam |
| 19:42.41 | Qwell[] | _Sam--: By not using shitty phones. :) |
| 19:42.55 | zoa | idefisk for mac will support zeroconf |
| 19:42.55 | Qwell[] | and you don't have just one config |
| 19:42.56 | zoa | :) |
| 19:43.04 | *** join/#asterisk NDT (n=me@cpe-24-194-166-119.nycap.res.rr.com) |
| 19:43.06 | coppice | dunno. apart from you and people having trouble because they put 0.0.3 on their machines, I haven't had any real feedback about pre22 |
| 19:43.07 | Qwell[] | zoa: Where is idefisk for Linux? :p |
| 19:43.11 | jbalcomb | iCEBrkr any reason to go with the SPA-3000 rather than SPA-2002? we are looking at using them for fax machines. |
| 19:43.13 | _Sam-- | you have a cfg file for each phone? according to grandstream, you would have a cfg-mac-address for each phone |
| 19:43.14 | justinu | _Sam--: you have one config file per phone. |
| 19:43.19 | justinu | that's right. |
| 19:43.34 | *** join/#asterisk paulos_ (n=paulos@200-168-112-132.dsl.telesp.net.br) |
| 19:43.36 | brad_mssw | jbalcomb: unless you need an FXO, i wouldn't think there's a reason |
| 19:43.52 | _Sam-- | so how you make, say 200 config files, 1 for each user? just a shell type script? |
| 19:44.02 | NDT | For 5ess... for switchtype do you just put 5ess? Or is it a different syntax like NI2 being national? |
| 19:44.09 | Qwell[] | meh |
| 19:44.09 | zoa | qwell |
| 19:44.10 | zoa | online |
| 19:44.10 | jbalcomb | brad_mssw: what is an FXO and what is it for? |
| 19:44.15 | Qwell[] | You people and your silly SIP... |
| 19:44.18 | zoa | http://www.asteriskguru.com/tools/idefisk_linux.php |
| 19:44.19 | Qwell[] | zoa: That new? |
| 19:44.19 | justinu | _Sam-: something like that |
| 19:44.26 | pardove | coppice: a full debug of app_rxfax is put on the *-users. subject is "spandsp-0.0.2pre22 not working" |
| 19:44.31 | _Sam-- | and then you have to find the mac of each phone? |
| 19:44.42 | justinu | yeah, it's printed on the bottom of the phone |
| 19:44.42 | _Sam-- | might as well just configure them manually |
| 19:44.49 | justinu | you're crazy |
| 19:44.54 | Qwell[] | I swear I just looked for a Linux version a few days ago |
| 19:44.59 | _Sam-- | it takes me about 2 minutes per gxp2000 |
| 19:45.02 | zoa | it just came out 2 days ago |
| 19:45.03 | _Sam-- | to do it from the web interface |
| 19:45.03 | zoa | :) |
| 19:45.08 | Qwell[] | great |
| 19:45.14 | _Sam-- | it would take longer to do it automated |
| 19:45.15 | justinu | _Sam--: i'd rather have those 2 minutes of my life back |
| 19:45.22 | zoa | its pretty new so might have some bugs |
| 19:45.24 | Qwell[] | zoa: expect a bunch of feedback from me :p |
| 19:45.28 | _Sam-- | it would take 1 minute to look up the mac and type cfg-mac |
| 19:45.31 | zoa | i had one bugreport on 150 downloads so far |
| 19:45.39 | Qwell[] | cool |
| 19:45.40 | *** join/#asterisk Iam8up|lappy (n=dontemai@cpe-71-65-112-38.woh.res.rr.com) |
| 19:45.42 | Iam8up|lappy | zu++ |
| 19:45.45 | zoa | with a second audio device that was not working |
| 19:45.46 | Iam8up|lappy | i don't get it |
| 19:45.47 | zoa | dunno why |
| 19:45.48 | Qwell[] | it does sip and aix, right? |
| 19:45.50 | zoa | hard to debug |
| 19:45.51 | justinu | _Sam--: sometimes you can't always get to the web interface of the phone |
| 19:45.53 | zoa | only iax for now |
| 19:45.55 | zoa | working on sip |
| 19:45.56 | Qwell[] | iax rather |
| 19:45.59 | justinu | _Sam--: like when it's at a customer who's behind nat |
| 19:46.00 | Qwell[] | does the windows client do sip? |
| 19:46.04 | zoa | not yet |
| 19:46.07 | Qwell[] | oh |
| 19:46.16 | Qwell[] | Do the two share any code? |
| 19:46.24 | zoa | the iaxclient library |
| 19:46.25 | Billy | Katty: are you a bot? |
| 19:46.26 | zoa | thats all |
| 19:46.32 | zoa | the rest is completely different code |
| 19:46.41 | zoa | but mac and linux share quite a lot of code |
| 19:47.01 | Qwell[] | alsa? |
| 19:47.10 | Katty | Billy: no, that's jbalcomb |
| 19:47.14 | Katty | Billy: i mean jbot |
| 19:47.16 | zoa | no idea |
| 19:47.16 | *** part/#asterisk greendisease (n=greendis@fedora/greendisease) |
| 19:47.17 | _Sam-- | i do hear what youre saying...but why wouldnt you configure the phones before you put them behind the nat? :) |
| 19:47.21 | zoa | i didnt write it |
| 19:47.24 | Qwell[] | oh |
| 19:47.28 | _Sam-- | like i configure my phones, then i install them |
| 19:47.32 | _Sam-- | not vice versa |
| 19:47.36 | Billy | Katty: If you were, you'd be a good one. |
| 19:47.36 | Katty | Billy: and /yes/ before you ask, i am female. |
| 19:47.40 | justinu | _Sam--: things change... |
| 19:47.56 | Katty | Billy: i'm going to take that as a compliment. |
| 19:48.05 | Billy | sure;-) |
| 19:48.07 | _Sam-- | fair enough, im just trying to understand, not trying to be a jackass (comes naturally) |
| 19:48.11 | Katty | Billy: because it would insanely annoy me if you were suggesting i was dumb. |
| 19:48.40 | justinu | _Sam--: walking a customer thru a web interface is something I don't enjoy doing |
| 19:48.48 | jbalcomb | Katty: Woman! I ain't no bot! |
| 19:48.56 | Katty | jbalcomb: i sowwy. |
| 19:49.04 | paulos_ | Hi, all. I'm using Unicall/MFC-R2. I got the following: Unicall/XX protocol error. Cause 32773 |
| 19:49.07 | Katty | jbalcomb: you screwed up my tab completeyness. |
| 19:49.17 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 19:49.18 | rob0 | <== a bot which uses /dev/random |
| 19:49.26 | _Sam-- | there's a way to configure those grandstreams to do exactly what you want using a central provisioning server...but its exactly as you you said...you have to configure the phone first to use it. |
| 19:49.36 | jbalcomb | Katty: yeah, cause IRC knows better than to let some damn bot have priority over me |
| 19:49.36 | De_Mon | trying to use h263 and I'm getting errors and no video: Jan 20 14:48:58 WARNING[12476]: chan_sip.c:3470 process_sdp: Unknown SDP media type in offer: video 13928 RTP/AVP 105 106 107 108 109 104 103 34 |
| 19:49.40 | Qwell[] | why wouldn't you configure a phone before using it? |
| 19:49.53 | Katty | jbalcomb: :> |
| 19:49.57 | _Sam-- | you have to configure the phone to use the central provisioning server |
| 19:49.59 | jbalcomb | Katty :) |
| 19:50.05 | Qwell[] | _Sam--: That's dumb |
| 19:50.10 | justinu | _Sam--: compeltely stupid |
| 19:50.13 | Billy | Katty: ooo look what I started! |
| 19:50.14 | Qwell[] | like I said, they're crap phones |
| 19:50.21 | iccomputing | Iam8up:<name>++ is for upping someones Karma if they are helping you out |
| 19:50.22 | Katty | Billy: and look what i finished. |
| 19:50.30 | _Sam-- | thats not true, but you can have your opinion. |
| 19:50.37 | _Sam-- | they are cheap phones that work well for cheap phones. |
| 19:50.43 | *** join/#asterisk pardove (n=pardove@195.146.47.239) |
| 19:50.48 | justinu | they need some work |
| 19:50.50 | justinu | but they hold promise |
| 19:51.04 | Qwell[] | I doubt we'll ever see a GS come out that people like... |
| 19:51.04 | zoa | _Sam--: the st302's are even better |
| 19:51.12 | zoa | and the thomsons are incredible |
| 19:51.45 | *** join/#asterisk simone (n=tin@host75-97.pool80181.interbusiness.it) |
| 19:52.05 | zoa | http://www.voipsolutions.be/product_info.php/cPath/54_24/products_id/210 |
| 19:52.07 | justinu | qwell: if they'd just open source the firmware........ |
| 19:52.09 | _Sam-- | i havent seen the st302 here |
| 19:52.13 | _Sam-- | <jbalcomb> Katty: Woman! I ain't no bot! |
| 19:52.13 | _Sam-- | <Katty> jbalcomb: i sowwy. |
| 19:52.15 | Qwell[] | meh, even that won't help |
| 19:52.20 | zoa | have a look at the thomson |
| 19:52.24 | Qwell[] | the hardware itself is junk |
| 19:52.37 | *** join/#asterisk zotz (n=zotz@24.231.47.175) |
| 19:52.51 | generalhan | Hey Everyone, i just upgraded to 1.2.1 and im getting some errors ive never seen before when i tail the message log, can some one please take a look at these and see if i should be worried about any of them ??? http://generalhan.pastebin.ca/37630 |
| 19:53.20 | justinu | qwell: i'm a bit more optimistic, i guess |
| 19:53.20 | [av]bani | have a source for st302 sales? |
| 19:53.32 | gaz00 | anyone got a recommendation for a voip provider in canada? |
| 19:53.42 | [av]bani | Qwell[]: people were saying that about sipura... then the 941 came out :) |
| 19:54.10 | [av]bani | and everyone was OMG OMG |
| 19:54.11 | Qwell[] | worst I heard about the sipura was that the buttons were sticky, or something |
| 19:54.42 | pardove | . |
| 19:54.53 | justinu | yeah, the buttons suck |
| 19:54.57 | justinu | and the speakerphone sucks |
| 19:55.01 | justinu | otherwise, it's a fine phone |
| 19:55.31 | generalhan | anyone have any suggestions about these errors ? http://generalhan.pastebin.ca/37630 ? i need some serious help |
| 19:56.20 | tainted_ | how do u guys solve NAT cutting off calls? |
| 19:56.33 | tainted_ | do u open/forward ports for RTP stream? |
| 19:56.46 | gaz00 | can't use iax? |
| 19:56.48 | jbalcomb | generalhan: the msg about dundi, cdr_custom, and extensions.ael are irrelevant |
| 19:56.51 | _Sam-- | nat usually doesnt cut calls off, in my opinion...it may prevent you from making them, or prevent you from hearing the other party or vice versa |
| 19:56.56 | tainted_ | gaz00 unfortunately no |
| 19:57.03 | [TK]D-Fender | generalhan : You have bugs in your dial-plan |
| 19:57.07 | tainted_ | do STUN servers help NAT issues? |
| 19:57.09 | Qwell[] | the router could lose the port mapping |
| 19:57.09 | [av]bani | tainted_: proper nat support in your client |
| 19:57.12 | Qwell[] | not uncommon with dlinks |
| 19:57.20 | [av]bani | stun servers only help if your client properly supports it |
| 19:57.21 | tainted_ | [av]bani meaning |
| 19:57.37 | generalhan | [TK]: any idea where they might be by looking at the errors ? |
| 19:57.41 | [av]bani | apparently, the expensive polycoms have shit nat support |
| 19:57.58 | [av]bani | while the cheapy gxp2000 have great nat |
| 19:58.04 | generalhan | cuase this config worked for the 1.0.9 implimentation, and i just upgraded and now nothing works |
| 19:58.09 | tainted_ | i have polycom 301 and gs 488s |
| 19:58.12 | tainted_ | both have nat issues |
| 19:58.13 | _Sam-- | i dont have any problems with any SIP or IAX client behind 1 single firewall using nat=yes |
| 19:58.20 | jbalcomb | [TK]D-Fender We have a Wildcard TE411P and a Wildcard TE110P (unused) |
| 19:58.29 | [av]bani | you really want to use stun with a grandstream, it is able to figure most of the nat stuff out that way just fine |
| 19:58.41 | tainted_ | _Sam-- what router, SIP client |
| 19:59.00 | [av]bani | and yeah, you need nat=yes in sip.conf for the extension |
| 19:59.12 | [TK]D-Fender | jbalcomb : I suggest napalm for them :D |
| 19:59.15 | [av]bani | too bad asterisk isnt smart enough to auto-detect |
| 19:59.39 | [TK]D-Fender | generalhan : it tells you the line #! go look at it. |
| 19:59.39 | jbalcomb | [TK]D-Fender nice |
| 19:59.43 | *** join/#asterisk benjk (n=benjamin@f8a01-0357.din.or.jp) |
| 19:59.56 | [TK]D-Fender | jbalcomb : So whatcha doing about your "issues"? |
| 20:00.04 | [av]bani | tainted_: in fact, i don't think the grandstream will properly do nat without a stun server |
| 20:00.10 | jbalcomb | [TK]D-Fender staring at my screen |
| 20:00.22 | _Sam-- | ive used xlite, idefisk (IAX), sipps, gxp2000s all behind residential type gateways no problem.... |
| 20:00.23 | [av]bani | at least, i had problems till i set a stun server up for it to use |
| 20:00.24 | jbalcomb | [TK]D-Fender taking extra smoke breaks |
| 20:00.45 | _Sam-- | where they are behind 1 single firewall, register to remote hosts fine, and make and receive calls fine, behind a firewall, with nothing special. |
| 20:01.06 | *** join/#asterisk FastJack (i=fastjack@p5091E315.dip.t-dialin.net) |
| 20:01.13 | generalhan | [TK]D-Fender: ok i fixed the issues in the dial plan that was just retardism, but im still getting this one that i dont understand .. any ideas on this one ? Failed to load configuration file. M |
| 20:01.13 | generalhan | odule not activated. |
| 20:01.27 | generalhan | sorry guys that was an accident i was suposed to paste the pastebin url |
| 20:01.35 | MrChimpy | i'm confused. if you're using TE411P should zaptel be using libpri? |
| 20:02.26 | tainted_ | [av]bani should i set up my own STUN server or use a public one |
| 20:02.38 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 20:02.51 | [av]bani | tainted_: i never had luck using a public server, i set up my own |
| 20:03.02 | tainted_ | which stun server did u use? |
| 20:03.10 | _Sam-- | someone said that xten runs a public stun server that works well |
| 20:03.30 | justinu | stun.xten.net |
| 20:03.59 | [av]bani | tainted_: http://sourceforge.net/projects/stun/ |
| 20:04.03 | pigpen | Here is a nice little NPA-NXX Lookup: http://bellsmind.net/Engine/BellsMindSearchPage.html |
| 20:04.35 | [av]bani | that stun server worked for me with gxp-2000 and spa-3000 |
| 20:06.53 | *** join/#asterisk sigmounte (n=sigmount@www.sighq.net) |
| 20:07.06 | *** part/#asterisk Samoied (n=Samoied@popeye.opens.com.br) |
| 20:08.53 | NewSole | hello |
| 20:11.12 | De_Mon | oh, hello.. I needed to uncomment videosupport=yes |
| 20:11.41 | De_Mon | how does stun work? I need a diagram. I mean, how does asterisk know about the stun server? |
| 20:12.11 | *** join/#asterisk ^^Gu[L]Can (n=MetRopoL@85.108.151.16) |
| 20:12.11 | justinu | read the STUN rfc |
| 20:12.54 | pifiu | in IAX, the "user" is the server and the "peer" is the client? |
| 20:13.12 | iccomputing | De_Mon: I dont think that the Asterisk server ever knows about the STUN per say... |
| 20:13.41 | iccomputing | from my experience, the STUN is just a proxy that forwards packets in and out of your NAT'd network to avoid the firewall... |
| 20:14.21 | justinu | stun isn't a proxy at all |
| 20:14.53 | iccomputing | I may be wrong, but from what i have experienced, there is no Asterisk config for the stun...as for a diagram, not sure... |
| 20:15.03 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 20:15.08 | iccomputing | sorry, maybe proxy was the wrong term but i dont know what else to call it.. |
| 20:15.13 | DarkFlibble | stun allows a client to understand the firewalls/translations between it and the net |
| 20:15.29 | justinu | yep |
| 20:15.33 | DarkFlibble | asterisk doesn't need to understand the client should make the changes |
| 20:15.35 | justinu | it doesn't proxy/forward any packets |
| 20:15.55 | iccomputing | my mistake...that was my impression...thank you for clarifying |
| 20:17.12 | iccomputing | We use the FWD stun server for some of our Wifi phones...it works great |
| 20:17.24 | iccomputing | since it worked, i never bothered to find out why =P |
| 20:18.37 | De_Mon | so stun just tells the client "that address is behind NAT here's how to reach them" |
| 20:18.48 | }btorch{ | anyone here can recomend an USB phone that can be used with * .. cheap one |
| 20:18.55 | justinu | stun tells the client what kind of NAt it's behind, and what it's external IPs are |
| 20:19.04 | justinu | that's it |
| 20:19.12 | iccomputing | AU100 |
| 20:19.18 | De_Mon | interesting, ok. |
| 20:19.38 | iccomputing | }btorch{: AU100...its a neat little IAX phone.. |
| 20:20.00 | iccomputing | we have had some bugs with the buttons...namely, the hang up button just puts people on hold =) |
| 20:20.21 | Iam8up|lappy | actually firefly seems to thin kthat the disconnect button is the same as the connect button |
| 20:20.30 | DarkFlibble | STUN can even cope with really strange NAT setups... (its worked with a dual net setup for me but YMMV) |
| 20:20.31 | }btorch{ | iccomputing: what about SIP ? |
| 20:20.33 | Iam8up|lappy | i'm sure theres a better softphone app out there that handles those phones correctly |
| 20:20.40 | X-Files | In eyeBeam worked Users status online, Video stream and Message ? |
| 20:21.03 | iccomputing | }btorch{: why use SIP if you have Asterisk??? |
| 20:21.23 | *** join/#asterisk h3x0r (n=h3xor@64.192.116.16) |
| 20:21.51 | }btorch{ | is IAX better ? Is just that I have been familiar with SIP and using x-lite so .. never changed to IAX |
| 20:22.08 | *** join/#asterisk dasuberdavid (n=dasuberd@gateway.digium.com) |
| 20:22.27 | jbalcomb | [TK]D-Fender: whats the trouble with those cards? I am not seeing anything out there about issues connected with them. |
| 20:22.29 | }btorch{ | actually i don't have a production box yet .. still shopping for parts like phones and stuff so I guess I have time to decide |
| 20:22.35 | De_Mon | for menu recordings, I am better off recording in PCM(ulaw) and then creating gsm's to avoid transcoding, right? |
| 20:22.41 | iccomputing | }btorch{: if you are using Asterisk, IAX is much much better...it has no NAT problems and the packets are less piggy ..about 30k instead of the SIP 80k packets.. |
| 20:22.53 | }btorch{ | really |
| 20:22.54 | Qwell[] | De_Mon: If you're using gsm, sure |
| 20:23.00 | [TK]D-Fender | De_Mon : depends if your server is under a lot of load or now |
| 20:23.11 | Qwell[] | iccomputing: The last part of that statement was very incorrect |
| 20:23.17 | }btorch{ | and using IAX2 on * I could have say 100+ users ? |
| 20:23.35 | Qwell[] | so was the first part |
| 20:23.44 | De_Mon | not really under load, but I'd rather not have to 'fix it' later |
| 20:23.45 | Iam8up|lappy | Qwell[] - correct us, please =) |
| 20:23.47 | zoa | btorch, yes |
| 20:23.57 | iccomputing | }btorch{: Use IAX if you can..you will have less problems!! however, most handset phones are SIP..no big deal, just more config...the beauty of IAX on usb phones is the mobility |
| 20:23.59 | Qwell[] | iax and sip packet sizes depend on the codec |
| 20:24.08 | iccomputing | Qwell...explain? |
| 20:24.17 | Qwell[] | what's to explain? |
| 20:24.33 | iccomputing | my experience is 24 - 30k IAX packets and 80k packets on standard ulaw? am i wrong? |
| 20:24.40 | Qwell[] | extremely wrong |
| 20:24.41 | [TK]D-Fender | jbalcomb : You mean what wrong with your 2 T1 cards? |
| 20:24.43 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:24.52 | Qwell[] | considering ulaw is 64k without overhead |
| 20:24.54 | jbalcomb | [TK]D-Fenderyes'm |
| 20:25.14 | [TK]D-Fender | jbalcomb : PM |
| 20:25.21 | iccomputing | i can get them to like lower with g729...but some equipment doesnt support it |
| 20:25.35 | Qwell[] | add overhead in there, and you've at 80k, whether you use sip or iax |
| 20:25.49 | iccomputing | i am talking real world with overhead... |
| 20:25.55 | }btorch{ | iccomputing: what you mean by just more config... on the usb phones ? |
| 20:25.55 | Qwell[] | yes, 80k, regardless |
| 20:26.03 | Qwell[] | you want less, use a different codec |
| 20:26.04 | jaike | its 84kbps |
| 20:26.07 | }btorch{ | does x-lite work with IAX ? |
| 20:26.13 | dudes | no |
| 20:26.18 | Qwell[] | jaike: yes, yes, get technical :P |
| 20:26.19 | jaike | g729 is 27k |
| 20:26.31 | jaike | :) |
| 20:26.32 | iccomputing | our traffic managers show 80k RTP with SIP and 30-40 with IAX... |
| 20:26.36 | Qwell[] | iccomputing: and iax is not "much much better" than sip |
| 20:26.41 | Qwell[] | iccomputing well, they're wrong |
| 20:26.54 | jaike | iccomputing: your iax channels are using a diff codec |
| 20:26.56 | }btorch{ | I'm gonna create some aix extensions and test them out next week |
| 20:27.18 | Qwell[] | aix? I wish asterisk did aix |
| 20:27.22 | iccomputing | }btorch{: more NAT problems...more port forwarding for signalling and things...SIP is not the best for mobility unless you use STUN |
| 20:27.30 | }btorch{ | hehe sorry |
| 20:27.32 | justinu | why? aix is a beast |
| 20:27.33 | Qwell[] | sip + nat = easy |
| 20:27.38 | Qwell[] | justinu: exactly! |
| 20:27.57 | Givur | Since we (small company, more outbound service, people sitting in germany and belgium) have switch to IAX2 we have much less problems. |
| 20:28.00 | jaike | last version of aix i used was 4.2 |
| 20:28.05 | Givur | And the quality is sometimes better as SIP. |
| 20:28.13 | Qwell[] | no, the quality is not better |
| 20:28.15 | iccomputing | Qwell, how do you figure? SIP is a piggy protocol and has many more config issues than IAX...thats why IAX was created! |
| 20:28.18 | Givur | (quality of using) |
| 20:28.35 | justinu | SIP is also capable of a lot more than IAX |
| 20:28.45 | Qwell[] | Inter-Asterisk eXchange...not Inter-Softphone eXchange |
| 20:28.57 | De_Mon | any AIX softphones with video support? |
| 20:29.00 | Qwell[] | though, that would be cool |
| 20:29.08 | Qwell[] | Softphone EXchange |
| 20:29.12 | iccomputing | haha...nice...good point yet the advantages are huge for our mobile clients |
| 20:29.13 | De_Mon | don't see any on wiki |
| 20:29.29 | Iam8up|lappy | softphone exchange sounds good =) |
| 20:29.31 | De_Mon | mmm SEX |
| 20:29.37 | Iam8up|lappy | ya, we got it.. |
| 20:29.41 | }btorch{ | what was the AIX phone you guys suggested again ? |
| 20:29.42 | De_Mon | :P wasn't sure |
| 20:29.46 | De_Mon | firefly? |
| 20:29.55 | Qwell[] | iax not aix |
| 20:30.01 | zoa | try idefisk |
| 20:30.07 | iccomputing | Atcom AU100 - firefly is good but now they are making it proprietary |
| 20:30.32 | iccomputing | there are a few out there, DIAX is another good one... |
| 20:30.36 | *** join/#asterisk n4y (n=tmalkut@fw.orasoft.net.pl) |
| 20:30.37 | zoa | http://www.asteriskguru.com/tools/idefisk_beta.php |
| 20:30.40 | iccomputing | not real fancy, but works.. |
| 20:31.00 | *** join/#asterisk dily (n=dily@ip-85-108.sn2.eutelia.it) |
| 20:31.13 | *** join/#asterisk ToTo (n=ToTo@host221-49.pool870.interbusiness.it) |
| 20:31.14 | ghento2 | Hi folks. I'm using Record() to record voice messages into wav files. Which conf file do I go to edit the voice quality? I want to increase it |
| 20:31.29 | iccomputing | Qwell, do you at least agree with me that IAX phones are good for mobility? AKA hotspot users? |
| 20:31.42 | zoa | iccomputing: did you try idefisk yet ? |
| 20:32.03 | iccomputing | idefisk? i missed that one..is it a softphoen? |
| 20:32.05 | zoa | yes |
| 20:32.07 | dily | can anyone help me to compile cdr_sqlite.so? plz? |
| 20:32.10 | zoa | available for mac, windows and linux |
| 20:32.14 | iccomputing | ahh ahh..i see it now.. |
| 20:32.15 | jaike | ghento2: you can only choose formats, gsm, wav, wav49 |
| 20:32.19 | Qwell[] | ghento2: You can tell record what format to use |
| 20:32.28 | iccomputing | Does it work with the USB phones? |
| 20:32.32 | Qwell[] | jaike: You sure about that? |
| 20:32.42 | zoa | depends what type i guess, dont have any myself |
| 20:32.47 | zoa | send me one and i will make it work |
| 20:32.53 | jaike | was thinking of Monitor |
| 20:33.04 | *** join/#asterisk gushi (i=danm@prime.gushi.org) |
| 20:33.06 | iccomputing | zoa: haha nice... |
| 20:33.15 | iccomputing | we bought like 5 of them for like 20 buck each |
| 20:33.16 | Qwell[] | iccomputing: I'd say he's serious |
| 20:33.33 | gushi | Hey all -- if I have a different ip for inbound and outbound, can I just have them both added to DNS to avoid having to add both entries? |
| 20:33.41 | iccomputing | i dont doubt it! |
| 20:33.59 | *** join/#asterisk kazalt (n=ftanguay@69.157.209.178) |
| 20:34.16 | ghento2 | wav49 would be better quality then wav i'm assuming? |
| 20:34.21 | iccomputing | wow, my tech is already loading idefisk!! |
| 20:34.22 | iccomputing | haha |
| 20:34.24 | dily | anyone use sqlite? |
| 20:34.36 | Qwell[] | ghento2: You should be able to use .ul |
| 20:34.37 | iccomputing | i didnt even get to ask him to do it =P |
| 20:34.39 | jaike | ghento2: smaller size |
| 20:34.41 | iccomputing | where is te fun in that!!? |
| 20:35.09 | gushi | i.e. provider X is having me send my outbound calls to x.x.x.12 but sending me calls FROM x.x.x.16 -- if they have x.x.x.16 resolve to callserver.x.com (PTR Record) but callserver.x.com resolves to x.x.x.12 (A Record) -- will that work? |
| 20:35.34 | zoa | yeah im serious :) |
| 20:35.38 | gushi | so I can just add a sip.conf entry for callserver.x.com instead of having to treat them separately? |
| 20:35.40 | zoa | or tell me where to find it in the eu |
| 20:35.43 | docelm0 | GUSHI! |
| 20:35.46 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 20:35.47 | zoa | im going to add all usb phones i can fidn |
| 20:36.08 | iccomputing | are you the developer zoa? |
| 20:36.22 | *** join/#asterisk _adrian (n=adrian@user-188.lns2-c11.dsl.pol.co.uk) |
| 20:36.31 | iccomputing | add the ATCOM AU100 =) |
| 20:36.31 | zoa | nopez but im the guy paying the bills :) |
| 20:36.34 | iccomputing | hahaha |
| 20:36.40 | iccomputing | i feel ya |
| 20:36.48 | iccomputing | im in that situation too |
| 20:37.19 | _adrian | Are you guys/gals up for any realy beginner questions? |
| 20:37.23 | zoa | actually there are 4 fulltime programmers working on idefisk |
| 20:37.31 | JMcA | _adrian: alright! questions I can answer! |
| 20:37.35 | iccomputing | you gonna drop the open source when its ready? |
| 20:37.37 | jaike | adrian: whats asterisk? |
| 20:37.43 | zoa | its not open source now either |
| 20:37.56 | Qwell[] | he means open source it |
| 20:38.02 | iccomputing | =P |
| 20:38.13 | _adrian | No I gather that asterisk is for VOIP, I got that from a mag! |
| 20:38.32 | ghento2 | would ulaw be the sound format that records in the highest quality? or gsm? |
| 20:38.38 | iccomputing | zoa: FYI it doesnt like the ATCOM AU100 on the first bout |
| 20:38.39 | JMcA | it can do VoIP...but its more than just that |
| 20:38.40 | Qwell[] | ulaw would be better |
| 20:39.02 | zoa | atcom au100, does that have an sdk ? |
| 20:39.31 | [TK]D-Fender | _adrian : So whats your question? |
| 20:39.40 | }btorch{ | the AU100 says it for skype though |
| 20:39.42 | _adrian | OK still learning all. I picked up the Linux Format mag when I saw the cover stories as it tied in with a work project |
| 20:39.58 | _adrian | The mag said it was simple to install!!! |
| 20:40.05 | Qwell[] | it is |
| 20:40.08 | iccomputing | }btorch{: trust me it works with Asterisj |
| 20:40.09 | Qwell[] | make && make install |
| 20:40.10 | _adrian | Could be used on an old PC ! |
| 20:40.10 | iccomputing | k* |
| 20:40.19 | Qwell[] | _adrian: also true |
| 20:40.27 | Qwell[] | So, what's the question? |
| 20:40.32 | }btorch{ | iccomputing: are you using on windows ? |
| 20:40.36 | _adrian | OK to the heart of my questions. |
| 20:40.38 | JMcA | _adrian: depending on what you're wanting to do, an old pc will likely be capable |
| 20:40.39 | iccomputing | i had a partner in the bahamas that was on vacation and called back here to the staes on it |
| 20:40.51 | iccomputing | yea...the AU100 is loaded on windows... |
| 20:40.56 | *** join/#asterisk Defraz (i=t0tal@72.165.56.43) |
| 20:40.58 | }btorch{ | iccomputing: I guess you need a softphone to work with it |
| 20:41.03 | iccomputing | it works with many client progs.. |
| 20:41.07 | _adrian | I need to demonstrate VOIP on an internal only network say max 40 clients |
| 20:41.07 | iccomputing | .yea |
| 20:41.15 | iccomputing | i use firefly currently |
| 20:42.03 | jaike | ? |
| 20:42.06 | iccomputing | }btorch{: it works with a lot of sip phones too |
| 20:42.18 | _adrian | Can it work easily with Suse as the mag tended to suggest Debian |
| 20:42.27 | Qwell[] | _adrian: Linux is Linux |
| 20:42.39 | cyburdine | _adrian: yup... I have several * machine running on suse |
| 20:42.46 | jaike | i use fedora..no problemos |
| 20:43.00 | JMcA | _adrian: I'm running it on suse...be aware that the packaged version that comes with it is a slight bit dated, but basically pretty functional |
| 20:43.03 | _adrian | Well to me it is not!, not being rude here but I have over the years tried various distros |
| 20:43.22 | Qwell[] | _adrian: Then you should know that if a program works on one distro, it'll work fine on any other |
| 20:43.25 | [TK]D-Fender | _adrian : It can work on pretty much and *nix, but you need to have the NORMAL software devel libraries installed which not all distro's do. Debian does, SAlackware does, most RH ones do. Don't know dfor SUSE specifically |
| 20:43.49 | _adrian | from old Re Hat to serveral Suse and even Mandrake some time ago and got well confused with different locations of pacakes |
| 20:44.00 | iccomputing | isnt CentOS the standard recomended? |
| 20:44.08 | Qwell[] | iccomputing: There is no "standard" |
| 20:44.25 | Qwell[] | ask 5 people, you'll get 8 different answers |
| 20:44.29 | _adrian | Sorry I have to watch the keyboard when I type and miss bits, must wait for responses |
| 20:44.41 | cyburdine | Qwell: exactly |
| 20:44.48 | JMcA | _adrian: suse 10 comes with asterisk packages, so you can just use yast to install them...config files are in /etc/asterisk, as is typical |
| 20:44.51 | iccomputing | jeeze Qwell! cut me a break! i meant from the support aspect, if you were to call Digium, dont they recommend CentOS??? |
| 20:44.58 | mog_work | no |
| 20:45.00 | mog_work | we dont |
| 20:45.00 | Qwell[] | I don't think they care |
| 20:45.03 | Qwell[] | ^ |
| 20:45.24 | Qwell[] | mog_work: !!! |
| 20:45.28 | jaike | debian does too...1.0.7 i think |
| 20:45.28 | }btorch{ | the firefly company doesn' have that anymore they got something called cubix |
| 20:45.42 | _adrian | OK I got Suse10 again from Mag but not found asterisk |
| 20:45.45 | JMcA | 1.0.9 is SUSE10 |
| 20:45.49 | Qwell[] | just compile it |
| 20:45.53 | _Sam-- | debian has 1.2.1 now |
| 20:45.54 | iccomputing | wow, the Wiki's and the forums have always read that if you need to call for support you would be best to have CentOS |
| 20:45.55 | Qwell[] | asterisk packages are silly |
| 20:45.59 | [TK]D-Fender | Many people use RHEL/CentOS so therefor its statistically easier to find people to help you with LINUX problems trying to get ASTERISK to compile/run. |
| 20:46.17 | Qwell[] | meh, gentoo is where it's really at |
| 20:46.21 | [TK]D-Fender | think of it that way |
| 20:46.27 | mog_work | support will reccomend debian, and fedora mostly |
| 20:46.30 | Qwell[] | I think many of the devs run gentoo |
| 20:46.30 | JMcA | Qwell: may be a good idea for a simple individual server or proof of concept demo....but packages are not silly when you're hearding a few hundred boxes |
| 20:46.37 | iccomputing | good to know |
| 20:46.37 | mog_work | but like qwell said it doesnt matter |
| 20:46.43 | mog_work | we love them all |
| 20:46.46 | Qwell[] | JMcA: Make your own package. Most of the distro packages suck |
| 20:46.47 | iccomputing | does XLite do IAX? |
| 20:46.51 | Qwell[] | no |
| 20:46.52 | cyburdine | _adrian: if you prefer suse, install it and I'll be happy to help where I can |
| 20:46.58 | _Sam-- | the debian package sucks...it doesnt has add-ons |
| 20:47.06 | mog_work | all the packages suck |
| 20:47.07 | _Sam-- | s/has/have |
| 20:47.09 | [TK]D-Fender | _adrian : If you install Slackware, I'll do the same :) |
| 20:47.12 | Qwell[] | I have yet to see one that doesn't |
| 20:47.15 | mog_work | just use the source |
| 20:47.16 | JMcA | Qwell: we do that when necessary, but obviously want to avoid rolling our own when possible... |
| 20:47.30 | _Sam-- | i use the debian package at a few client locations, it works fine |
| 20:47.47 | dudes | I think people would save time if they'd just compile * (it takes 5 minutes) |
| 20:47.47 | X-Files | Have Users used ASTERISK and EYEBEAM ???? I have 2 question ... |
| 20:47.49 | *** join/#asterisk malaysia (n=malaysia@c-24-131-187-30.hsd1.ma.comcast.net) |
| 20:48.04 | mog_work | OKAY!!!!!! |
| 20:48.05 | Qwell[] | X-Files: NO NEVER USED them before |
| 20:48.11 | _adrian | OK thanks for support to you all. i will check the 5 pack CD I have to see if on if not redirect to web based install |
| 20:48.12 | iccomputing | i have ran it on Debian..but i like CentOS the best.. |
| 20:48.14 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 20:48.19 | Math` | X-Files: Im using * and eyebeam |
| 20:48.26 | Qwell[] | mog_work: You're an excited Qwell? |
| 20:48.26 | rob0 | (regardless of that others do :) ) |
| 20:48.33 | JMcA | _adrian: I'm using SUSE10 with the asterisk 1.0.9 packages for it...they work, though aren't, obviously, the absolute latest version of asterisk...I'll be glad to help where I can with asterisk - SUSE interactions |
| 20:48.34 | dudes | 2 and a half on decent P4 DC |
| 20:48.49 | Qwell[] | yay! |
| 20:48.52 | qwell[excited] | yes |
| 20:48.55 | qwell[excited] | yes i am |
| 20:48.58 | Qwell[] | about what? |
| 20:49.04 | qwell[excited] | everything |
| 20:49.16 | [TK]D-Fender | _adrian : Do NOT use "packaged" Asterisk, always compile from source. |
| 20:49.18 | Qwell[] | did Mark give you redbull again? |
| 20:49.19 | Qwell[] | tsk tsk tsk |
| 20:49.24 | qwell[excited] | heh |
| 20:49.30 | qwell[excited] | no just drpepper at the moment |
| 20:49.38 | qwell[excited] | but my new computer is on the way |
| 20:49.38 | JMcA | stkn: how many boxes do you heard overall, in your job? |
| 20:49.41 | Qwell[] | ugh, don't even get me started with dr pepper |
| 20:49.48 | iccomputing | _adrian: i can atest to that..."packaged" asterisk is more headaches than learning it from the start |
| 20:49.48 | JMcA | gah...that was to [TK]D-Fender |
| 20:50.01 | Qwell[] | I can either go to a vending machine and get a sprite for $100, or to the caf and get a dr pepper for $135 |
| 20:50.09 | Qwell[] | $1.00 and $1.35...yeah |
| 20:50.14 | rob0 | whew |
| 20:50.15 | Qwell[] | period wasn't working |
| 20:50.18 | qwell[excited] | woahhh |
| 20:50.23 | Qwell[] | $1.35 for a BOTTLE...20oz |
| 20:50.24 | qwell[excited] | thats is expensive |
| 20:50.30 | rob0 | /dev/wife had the same problem |
| 20:50.33 | Qwell[] | or $.75 for a can |
| 20:50.33 | _adrian | OK next question to deal with approx 40 clients, provide mailboxes, and have to cope with say maximum of 15 sim calls what type of PC server would be recommended |
| 20:50.44 | Qwell[] | so, I've been stuck with freaking sprite for 3 months |
| 20:50.45 | Qwell[] | :( |
| 20:50.47 | [TK]D-Fender | JMcA : Nobody respectable here would want to trust a "dubious" package for it... |
| 20:50.53 | qwell[excited] | you should move here qwell |
| 20:50.55 | iccomputing | muahaha.../dev/wife ha! |
| 20:50.55 | qwell[excited] | 50cents |
| 20:51.02 | Qwell[] | I don't like cans |
| 20:51.05 | Qwell[] | I want a bottle :( |
| 20:51.07 | _adrian | I ask this because! |
| 20:51.07 | _Sam-- | qwell if you would buy gxp2000s instead of spa941 you could afford some DP |
| 20:51.09 | qwell[excited] | 1.00 |
| 20:51.13 | qwell[excited] | for a bottle |
| 20:51.17 | iccomputing | omg Qwell! i will paypal you 35 cents shit! |
| 20:51.20 | Qwell[] | yeah...this caf rips us off |
| 20:51.21 | dudes | Qwell[] - You have a Sam's club in your area? They tend to have good pop deals. |
| 20:51.28 | jbalcomb | _adrian perhaps go to digium.com and get the dell server model they recommend |
| 20:51.29 | rob0 | /dev/wife whose period is not working ... scary! |
| 20:51.31 | Qwell[] | iccomputing: 35c/day? Done |
| 20:51.37 | justinu | pop!! |
| 20:51.37 | iccomputing | haha |
| 20:51.39 | JMcA | *shrug*...I'm still in proof-of-concept stage, but if/when we go production, there's no way we'll do a raw compile from source...at the very least, we'll create our own packages...we don't do *anything* without packages |
| 20:51.39 | dudes | Myself, I'm happy with a can of coffee a week. |
| 20:52.03 | _adrian | I can either use my own PC and elderly P800, or buy myself and old server and play with that.... |
| 20:52.09 | JMcA | and if I don't see any brokenness that affects me in the pre-made packages, I see no reason to roll my own |
| 20:52.27 | Qwell[] | JMcA: Have fun during upgrads |
| 20:52.30 | Qwell[] | upgrades* |
| 20:52.31 | _adrian | I would like to go any buy a new server, but much of this is off my own back for proof of concept! |
| 20:52.32 | jbalcomb | _adrian i have a quad zeon 3.0 Ghz, 2GB RAM, debian server for 120 phones, 16 sim calls max and i dont see above .1 utilization |
| 20:52.38 | Qwell[] | probably half the packages hose your configs |
| 20:52.38 | [TK]D-Fender | _adrian : So barring Linux problems, it'll work jsut fine as a VoIP sample. |
| 20:52.38 | cyburdine | _adrian: sure you can... but you may not be able to handle 40 clients |
| 20:52.40 | Iam8up|lappy | what? i thought this was #asterisk not #vendingmachineschargemetoofuckingmuch |
| 20:52.49 | Qwell[] | Iam8up|lappy: You must be new |
| 20:52.50 | JMcA | Qwell: dude...seriously...how many boxes do you herd professionally? |
| 20:52.57 | Iam8up|lappy | Qwell[] - i must =) |
| 20:52.58 | Qwell[] | JMcA: enough |
| 20:53.02 | jbalcomb | Iam8up|lappy: you apparently missed the memo |
| 20:53.16 | Iam8up|lappy | yaaahhh..i'm gonna need another copy of that memo...that'd be greeeeeeaaaaaaaaaatt... |
| 20:53.20 | cyburdine | _adrian: to get comfortable with it I'd suggest installing it on anything you got... then scale up from there... |
| 20:53.37 | JMcA | I, and a couple of other guys manage several hundred...at that scale, you don't do *ANYTHING* without packages...it just gets insane to try it |
| 20:53.40 | _adrian | Will do |
| 20:53.56 | Qwell[] | JMcA: Make your own packages, which are compiled for your machines, with your configs |
| 20:53.59 | cyburdine | _adrian: the more you install/configure it the more comfortable you are going to feel about rolling this into production |
| 20:54.01 | Qwell[] | not difficult |
| 20:54.06 | Qwell[] | compile them once...done |
| 20:54.13 | jbalcomb | _adrian polly a 1Ghz w/ 1GB RAM would work fine for you |
| 20:54.19 | Qwell[] | That's what I don't get about people who say a gentoo farm is a dumb idea |
| 20:54.20 | JMcA | again...if the premade packges work...I see no reason to do that |
| 20:54.24 | _Sam-- | what is the advantage to compiling your own binary? |
| 20:54.33 | _Sam-- | if a precompiled binary works fine |
| 20:54.38 | JMcA | oh...gentoo...nm |
| 20:54.39 | Qwell[] | _Sam--: faster, you can strip what you want, etc |
| 20:54.39 | jbalcomb | _Sam-- things actually work right? |
| 20:55.09 | jaike | i think adrian is more confused now :P |
| 20:55.09 | _Sam-- | i have both, from source, and packages boxes...and have no difference between either |
| 20:55.13 | _adrian | 2x PII 400 XEON (DUAL installed, quad capable) 1MB L2 Cache 1024 MB RAM (original HP Part) 7x 18.2GB disks (D7174A) 2x 9.1GB disks HP Netraid II Raid controler (bios controllable, no disks required to configure, with onboard cache memory and battery) 3x PSU (parallel operation, redundant) 2x 100MBit Network cards Onboard VGA / kb / mouse / ports |
| 20:55.23 | JMcA | faster....right....you keep telling yourself that |
| 20:55.25 | jbalcomb | we use debian and the package manager doesnt mind when we install from source |
| 20:55.30 | Qwell[] | JMcA: k, will do |
| 20:55.54 | jbalcomb | _adrian that sounds just fine |
| 20:56.03 | dudes | I make tar files of asterisk and have a little bash script that does everything .... makes libpri/zaptel/asterisk/cp's configs |
| 20:56.14 | SkalTura | _adrian: damn, that has been one hell of a server at it's day! |
| 20:56.24 | Qwell[] | dudes: exactly |
| 20:56.30 | _adrian | I can afford that at £140 and not cry if it breaks |
| 20:56.31 | jbalcomb | dudes ditto also for apache/mysql/sendmail/etc. |
| 20:56.45 | cyburdine | _adrian: yup... got a similar beast here... 4 p3 xeon... it works great... go for it! |
| 20:56.47 | Qwell[] | _adrian: .ú140 ? |
| 20:56.51 | [av]bani | http://funroll-loops.org/ |
| 20:57.05 | Qwell[] | showed up as funky chars here... |
| 20:57.05 | dudes | It doesn't take long to compile on todays servers |
| 20:57.12 | Qwell[] | [av]bani: I live by -funroll-loops :P |
| 20:57.12 | _adrian | 140 UK pounds |
| 20:57.31 | jbalcomb | _adrian are there other pounds besides UK? |
| 20:57.36 | Qwell[] | that and -j16 and -O99 |
| 20:57.40 | _adrian | Yes |
| 20:57.44 | [av]bani | -O31337 |
| 20:57.51 | jbalcomb | _adrian whos? |
| 20:57.54 | JMcA | [av]bani: I'm with you |
| 20:58.26 | zoa | i hate this damn AMP |
| 20:58.29 | _adrian | There are Scottish pounds say value and Irish Pounds or punts I think called locally |
| 20:58.42 | DarkFlibble | there are no punts anymore... |
| 20:58.45 | _Sam-- | what / why are you doing with AMP |
| 20:58.49 | DarkFlibble | euros in ireland |
| 20:58.53 | _Sam-- | there is nothing good in there except the CDR stuff |
| 20:58.54 | _adrian | Ok I stand corrected |
| 20:58.57 | *** join/#asterisk pifiu-laptop (n=someone@adsl-068-213-231-041.sip.mia.bellsouth.net) |
| 20:59.04 | DarkFlibble | scottish pound is tied to british pound |
| 20:59.25 | rob0 | jbalcomb: http://www.xe.com/ucc/ yes, Egypt |
| 20:59.47 | [av]bani | _Sam--: FOP is cute |
| 20:59.48 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 21:00.05 | [av]bani | tho i think you could do the same thing these days with AJAX instead of flash |
| 21:00.08 | jbalcomb | ok, sounds pounds in egypt, scotland, ireland, and england |
| 21:00.09 | zoa | any amp guru here ? |
| 21:00.11 | _adrian | just browing yast! |
| 21:00.15 | [av]bani | zoa: #amportal |
| 21:00.19 | DarkFlibble | jbalcomb, not ireland! |
| 21:00.29 | _Sam-- | hmm...im not so fond of FOP |
| 21:00.36 | rob0 | Lebanon too |
| 21:00.37 | jbalcomb | DarkFlibble okidoki, not ireland. i hate them nasty drunkards anyway. |
| 21:00.40 | DarkFlibble | northen ireland yes... but not southern... |
| 21:00.48 | _Sam-- | if you run FOP for a while, all the stuff gets overwritten and looks like crap |
| 21:00.49 | DarkFlibble | southern is eurozone |
| 21:00.53 | [av]bani | _Sam--: what webinterface do you use with equivalent functionality to FOP then? |
| 21:01.01 | _Sam-- | 'show channels' |
| 21:01.03 | jbalcomb | DarkFlibble thats cool, i dont northern or southern |
| 21:01.07 | [av]bani | :P |
| 21:01.11 | zoa | nobody awake there |
| 21:01.21 | jbalcomb | s/dont/dont like |
| 21:01.23 | _Sam-- | what are you using FOP for, just to see whos on the phones? |
| 21:01.30 | [av]bani | the boss likes it |
| 21:01.31 | [av]bani | shrug |
| 21:01.33 | zoa | haha |
| 21:01.46 | [av]bani | i thinkhe likes the jangly animation on the phone |
| 21:01.57 | _Sam-- | lol |
| 21:02.02 | _Sam-- | that is fun when all the phones ring at once |
| 21:02.07 | [av]bani | :) |
| 21:02.21 | [av]bani | it would be cool if phones did that in real life |
| 21:02.33 | [av]bani | 'ya my phone rings so loudly the receiver bounces all over' |
| 21:02.40 | _Sam-- | lol |
| 21:02.55 | _Sam-- | my FOP sometimes isnt so accurate |
| 21:03.02 | [av]bani | it loses counters on reload |
| 21:03.04 | _Sam-- | like ive seen it show people on calls who hung up a while ago |
| 21:03.19 | [av]bani | and if you add extensions it doesnt update of course |
| 21:03.28 | _Sam-- | right, you have to restart opserver? |
| 21:03.38 | [av]bani | afaik just a page reload is ok |
| 21:03.56 | [av]bani | but as i said i think the ame thing could be done with AJAX these days instead of flash |
| 21:04.18 | [av]bani | the UI isnt so nice either |
| 21:04.34 | _Sam-- | nah ive accidentally dropped a bridged call because i double clicked or something one time |
| 21:04.54 | _Sam-- | but ajax would probably work fine, using the call manager interface through php or something |
| 21:04.56 | *** join/#asterisk dijit0 (n=dijit0@adsl-68-127-138-64.dsl.pltn13.pacbell.net) |
| 21:04.56 | [av]bani | yeh, thats what i mean. the UI isnt so nice |
| 21:05.11 | [av]bani | the way you handle pulling up data, you can accidentally terminate calls |
| 21:05.20 | [TK]D-Fender | Just run IPSwitchboard on a PC. |
| 21:05.30 | asteriskmonkey | what is ajax? |
| 21:05.33 | [av]bani | [TK]D-Fender: and for PDA? |
| 21:05.40 | [TK]D-Fender | asteriskmonkey : A household cleaner? :) |
| 21:05.45 | [av]bani | "just VNC in from PDA" |
| 21:05.49 | asteriskmonkey | a city near toronto ? |
| 21:05.50 | asteriskmonkey | lol |
| 21:05.51 | [TK]D-Fender | [av]bani : Don't get whiney with me YOU! |
| 21:05.52 | zoa | sam, im doing it in ajax |
| 21:05.53 | zoa | :) |
| 21:05.56 | Iam8up|lappy | asteriskmonkey - what's up man? |
| 21:06.04 | _Sam-- | it should be really easy |
| 21:06.12 | asteriskmonkey | Iam8up: hows beth |
| 21:06.22 | _Sam-- | at least, the really basic part...prettying it up...thats a whole nother story |
| 21:06.32 | Iam8up|lappy | asteriskmonkey - she's fine, and yourself? |
| 21:07.24 | _adrian | Back,,,, OK checked my Yast and not found any asterisk looks like not on my open suse disks |
| 21:07.26 | iccomputing | asteriskmonkey - would i have to pay duties to get yur products to the US or do you have some distro help over here? |
| 21:07.39 | asteriskmonkey | we ship to the us all the time |
| 21:07.44 | asteriskmonkey | no duties |
| 21:07.53 | asteriskmonkey | that im aware of.. im tech side though :) |
| 21:08.01 | [av]bani | [TK]D-Fender: i've got my sipura auto-provision script making sip.conf extensions on the fly now :) |
| 21:08.12 | Iam8up|lappy | asteriskmonkey - can you help me with those sh scripts when you get a chance? |
| 21:08.14 | asteriskmonkey | ajax=java api? |
| 21:08.14 | iccomputing | mmmm....i will have to ask your sales guy then.. |
| 21:08.14 | Traderzz | when was 1.2.2 released ? |
| 21:08.21 | Iam8up|lappy | Traderzz - 2 days ago |
| 21:08.29 | Traderzz | what changes were made? |
| 21:08.33 | asteriskmonkey | Iam8up: the ones i gave you work fine whats the issue? |
| 21:08.35 | Iam8up|lappy | Traderzz - the changelog is on the asterisk homepage |
| 21:08.36 | mog_work | there is a changelog |
| 21:08.39 | Traderzz | k |
| 21:08.46 | Qwell[] | mog_work: New computer? |
| 21:08.50 | Iam8up|lappy | asteriskmonkey - actually..the URLs are broken... |
| 21:08.58 | mog_work | im getting my sisters old lappy, 900mhz |
| 21:09.07 | mog_work | aka twice speed of current laptop |
| 21:09.11 | [TK]D-Fender | [av]bani : Cool.... starting your own telco now? |
| 21:09.14 | mog_work | and its 14 inch 1024x768 |
| 21:09.16 | asteriskmonkey | n othere not, try not cutting and pasting in windows hehehe |
| 21:09.17 | JMcA | asteriskmonkey: ajax is a pseudo-technology...its the use of JavaScript making XMLHTTPRequests to web servers, retrieving XML files, which are then used to update the browser DOM to change the page layout on the fly |
| 21:09.18 | mog_work | which is most important part |
| 21:09.19 | *** join/#asterisk IMG-SD (n=IMG-SD@64.5.206.131) |
| 21:09.36 | Qwell[] | mog_work: nice |
| 21:09.49 | asteriskmonkey | JMcA: nice :) |
| 21:09.50 | mog_work | 800x600 was driving me batty |
| 21:09.52 | [av]bani | [TK]D-Fender: yea, with piles of SPA-3000's |
| 21:10.08 | Qwell[] | mog_work: Linux? |
| 21:10.16 | mog_work | it will be |
| 21:10.17 | IMG-SD | I'm seeing these messages in my Asterisk CLI every seconds or so... any idea what's happening?: |
| 21:10.17 | IMG-SD | <PROTECTED> |
| 21:10.17 | IMG-SD | <PROTECTED> |
| 21:10.19 | mog_work | she runs winders |
| 21:10.22 | Qwell[] | mog_work: My wifes Dell says it only does 1024x768...it does 1280x1024 though :D |
| 21:10.22 | JMcA | asteriskmonkey: yeah, it can result in some really slick web pages/apps...but using it is not without its downsides |
| 21:10.24 | asteriskmonkey | Iam8up|lappy: cutting and pasting in windows will give you undesirable hidden characers |
| 21:10.32 | jaike | img-sd: asterisk -r |
| 21:10.37 | mog_work | yeah i think it will do 1280 in linux |
| 21:10.41 | Iam8up|lappy | asteriskmonkey - i used linux =P |
| 21:10.44 | mog_work | windows wont let her do more |
| 21:10.45 | asteriskmonkey | JMcA: probably way ass heavy on cpu |
| 21:10.47 | Qwell[] | yeah |
| 21:10.52 | Qwell[] | just depends on the video ram |
| 21:10.53 | JMcA | asteriskmonkey: FWIW, Google Maps stuff is all done with AJAX sorta stuff, so that type of interface is what's being talked about |
| 21:10.58 | mog_work | its adjustable |
| 21:10.59 | JMcA | asteriskmonkey: not so bad as you'd think |
| 21:11.03 | mog_work | so i think i can get it working |
| 21:11.03 | Qwell[] | 1280 * 1024 * 32 == how much ram you need |
| 21:11.09 | mog_work | and it has battery |
| 21:11.12 | IMG-SD | jaike: You mean someone is connected to the CLI using asterisk -r? I'm currently connected using asterisk -rvvvc |
| 21:11.13 | mog_work | my laptop has no battery |
| 21:11.15 | asteriskmonkey | there sent it in pw |
| 21:11.15 | iccomputing | anyone ever put Asterisk on a WRAP board? |
| 21:11.19 | Qwell[] | I just had to set hers to 8mb (which is only settable in a new bios) |
| 21:11.19 | jaike | yup |
| 21:11.25 | Iam8up|lappy | the problem is that http://svn.digium.com/svn/asterisk-addons/branches/1.2 doesn't contain the different files |
| 21:11.28 | [av]bani | iccomputing: i think gumstix is more interesting than WRAP |
| 21:11.28 | iccomputing | i am going to do it ...i was just cuirous of the performance.. |
| 21:11.32 | jpablo | grrr, my linux box is freezing when i mess with zaptel :S |
| 21:11.35 | [TK]D-Fender | I love my 1440x900 personally :) |
| 21:11.47 | iccomputing | <PROTECTED> |
| 21:11.47 | IMG-SD | jaike: Ok. I'll check around to see who's connected. Thanks! |
| 21:11.48 | Qwell[] | and it's weird...she wasn't able to play videos with vlc before...now that I changed it, they play fine |
| 21:12.11 | mog_work | i just want something thats not a ukranian peice of crap |
| 21:12.12 | [av]bani | iccomputing: http://www.gumstix.com/products.html |
| 21:12.21 | mog_work | i think im gonna go officespace on this computer when i get new one |
| 21:12.29 | Qwell[] | mog_work: record it |
| 21:12.36 | mog_work | heh |
| 21:12.37 | [TK]D-Fender | Qwell : I love VLC.... starts VERY fast doesn't whine at me, slim interface and programmable hot keys. and that ht EASY stuff... |
| 21:12.50 | *** join/#asterisk mazzanet (n=irc@unaffiliated/mazzanet) |
| 21:13.00 | mazzanet | something broke by itself :( |
| 21:13.02 | mog_work | vlc is cleaner but when you absolutely need it to play there is mplayer |
| 21:13.07 | mazzanet | Jan 21 08:12:18 NOTICE[3707]: chan_iax2.c:7198 socket_read: Rejected connect attempt from 202.125.42.141, request 's@from-sip' does not exist |
| 21:13.08 | JMcA | mog_work: if you come up my way, there's a place that has a machine gun shoot at an outdoor shooting range...that should satisfy your office space urgings quite well |
| 21:13.25 | mog_work | ooh |
| 21:13.26 | [TK]D-Fender | mog_work : Then again I'm using VLC on WINDOWS. I use XINE on my home server... |
| 21:13.29 | mog_work | where are you JMcA |
| 21:13.34 | JMcA | mog_work: Louisville, KY |
| 21:13.36 | mog_work | and do you have such artillery |
| 21:13.41 | mog_work | ooh i have friends out ther |
| 21:13.42 | mog_work | e |
| 21:13.46 | JMcA | its just a bit south of here, Knob Creek |
| 21:13.46 | mog_work | itd be worth going |
| 21:13.51 | Qwell[] | heh |
| 21:14.01 | Traderzz | anything in 1.2.2 significant that would make you update from 1.2.1 ? |
| 21:14.06 | JMcA | I think you can rent them at the range, though |
| 21:14.07 | mazzanet | i don't see where the 's' in the 's@from-sip' is coming from |
| 21:14.07 | Qwell[] | Traderzz: bug fixes |
| 21:14.13 | *** part/#asterisk ghento2 (i=ghento@CPE0011d8a291a6-CM00111ae4684c.cpe.net.cable.rogers.com) |
| 21:14.29 | Traderzz | Qwell, anything major or show stoppers that were fixed? |
| 21:14.35 | JMcA | I did hear someone say that the waiting list is like 2 years long to get a spot on the firing line for that |
| 21:14.40 | Qwell[] | quite a few major bugs |
| 21:14.44 | iccomputing | <PROTECTED> |
| 21:14.52 | Traderzz | ok so is the ugprade just copy over teh old files or ? |
| 21:15.07 | jaike | traderz: i think the voicemail bug was fixed..read the changelog..tons of fixes |
| 21:15.19 | Qwell[] | jaike "the" voicemail bug? heh |
| 21:15.24 | Traderzz | eheheh jaike, what voice mail bug? |
| 21:15.25 | [av]bani | [TK]D-Fender: can you give me a measurement for the ip601 screen? |
| 21:15.28 | JMcA | Traderzz: just install the upgraded package...oh wait, qwell doesn't believe in packages |
| 21:15.30 | jaike | yup the voicemail bug |
| 21:15.34 | jaike | gave us headaches |
| 21:15.45 | Qwell[] | JMcA: You'd be hard pressed to find a 1.2.2 package already |
| 21:15.47 | [TK]D-Fender | [av]bani : 4" x 2" |
| 21:15.50 | jaike | 1.2.0 didnt have it..1.2.1 had it |
| 21:15.56 | [av]bani | [TK]D-Fender: exactly? |
| 21:16.01 | [TK]D-Fender | [av]bani : yup |
| 21:16.09 | [TK]D-Fender | ruler & phone right in front of me |
| 21:16.09 | jbalcomb | what is this about? "Resyncing the jb" |
| 21:17.38 | [av]bani | [TK]D-Fender: have a 7940g around? |
| 21:17.51 | jaike | jbalcomb: i think its a jitterbuffer msg |
| 21:18.05 | zoa | jbalcomb : that happens when the timestamps are too different |
| 21:18.06 | [TK]D-Fender | [av]bani : Sorry, nope |
| 21:18.10 | zoa | and then it resynchs |
| 21:18.22 | [TK]D-Fender | [av]bani : only 60x's here and UIP-200's. SPA-941 at home |
| 21:18.39 | [av]bani | uip-200 ? |
| 21:18.45 | [TK]D-Fender | [av]bani : Uniden |
| 21:19.17 | jbalcomb | jaike zoa thanks, ill go research. im getting this a lot it seems. |
| 21:19.19 | [TK]D-Fender | A lot of good hope for them... evaporated... Disappointing phones I bought for "high risk" locations. |
| 21:19.30 | *** join/#asterisk bertian (i=darby_t@dlj145.neoplus.adsl.tpnet.pl) |
| 21:19.52 | _Sam-- | [av]bani: did you ever come up with a good recommendation for a 4 port external FXO? |
| 21:20.05 | [av]bani | _Sam--: 4 x spa-3000 :P |
| 21:21.02 | *** join/#asterisk bkw_ (n=bkw_@ppp-70-128-122-10.dsl.tulsok.swbell.net) |
| 21:22.44 | [av]bani | [TK]D-Fender: have you got a screenshot of your ip601 showing line status with your custom app? eg queue status, extension status etc |
| 21:22.55 | [TK]D-Fender | [av]bani : I could take one... |
| 21:23.03 | _Sam-- | zoa has a queue thing for windows that shows you your queue status in the tray |
| 21:23.04 | jbalcomb | [av]bani is the SPA-2002 good enough for my fax machines? |
| 21:23.07 | _Sam-- | i havent seen it yet |
| 21:23.15 | [av]bani | could you? boss is interested in ip601 but wants to know if it can display all the info he wants legibly |
| 21:23.40 | _adrian | To help me with the server PC side of things is the a channel to discuss Linux on servers? |
| 21:23.50 | *** join/#asterisk miguellinux (n=miguel@fw.vsp.com.pe) |
| 21:23.51 | Qwell[] | _adrian: Not really |
| 21:24.03 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 21:25.11 | _adrian | OK can Open Suse or Suse10 deal witl duel processors will asterisk be able to use them, how does it all work? |
| 21:25.23 | Qwell[] | sure it can |
| 21:25.34 | Qwell[] | just use an smp enabled kernel |
| 21:25.41 | [TK]D-Fender | [av]bani : Just took 3 shots that'll be ready when I get home |
| 21:26.10 | [av]bani | \o/ |
| 21:26.26 | _adrian | smp? It may be a daft question but I need to ask? |
| 21:26.32 | Qwell[] | ~smp |
| 21:26.33 | jbot | well, smp is (Symmetric Multi Processing) This refers to a technology where a computer uses multiple processors to process different instructions at the same time, in separate processing units. It is a form of parallel computing.. A feature of an SMP system is that it uses shared memory between all the processors, rather than each processor having its own unique ... |
| 21:27.06 | *** join/#asterisk fulgas (n=fulgas@a81-84-117-84.cpe.netcabo.pt) |
| 21:27.28 | _adrian | Ar |
| 21:27.33 | _adrian | sorry |
| 21:28.30 | *** join/#asterisk santiago (n=santiago@208.195.215.222) |
| 21:28.31 | [TK]D-Fender | jbalcomb : do NOT let * near your fax machines if you know whats good for you.... |
| 21:28.31 | jbalcomb | [TK]D-Fender i have 40 fax numbers and have no idea how to do that without asterisk |
| 21:28.39 | _adrian | Is there any way on knowing is an SMP distro is on my disks or whould I just try to install and see what hapens? |
| 21:28.40 | [TK]D-Fender | jbalcomb : only for DID faxing. Primary faxes (customer service, etc), and anything physical should be ANALOG. |
| 21:29.04 | jbalcomb | [TK]D-Fender its all DiD |
| 21:29.04 | *** join/#asterisk pigpen2 (n=mark@fw.seamans.cc) |
| 21:29.21 | [TK]D-Fender | jbalcomb : I mea incoming SpanDSP usage. low volume/importance. |
| 21:29.44 | [TK]D-Fender | I tried running mine on a Channel bank alone and that screwed me. VoIP is an even WORSE idea. |
| 21:29.49 | jbalcomb | [TK]D-Fender i dont know SpanDSP |
| 21:29.50 | _adrian | And last question is there a simple VOIP bandwidth calculator for SIP? |
| 21:30.13 | [TK]D-Fender | jbalcomb : What are you using for faxes? |
| 21:30.23 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 21:30.23 | [TK]D-Fender | _adrian : plenty, check the WIKI |
| 21:30.45 | jbalcomb | [TK]D-Fender ive got too much to deal with for the moment so if the SPA-2002 will offer improvement over the GS handyshits i'll take it |
| 21:30.49 | _adrian | WIKI? |
| 21:30.53 | [TK]D-Fender | ~docs |
| 21:30.55 | jbot | [docs] probably Documentation can be found at http://www.digium.com/index.php?menu=documentation or http://www.digium.com/handbook-draft.pdf or http://www.voip-info.org/wiki-Asterisk or http://www.asteriskdocs.org or http://www.oreilly.com/catalog/asterisk or http://www.asteriskguru.com, or http://www.astmasters.net/howtos.html |
| 21:30.59 | jbalcomb | [TK]D-Fender my ATAs? |
| 21:31.03 | [TK]D-Fender | jbalcomb : You're running faxes on HT's? |
| 21:31.21 | jbalcomb | [TK]D-Fender 'they' are but yes |
| 21:31.32 | _adrian | OK clicked on the link and will bookmark |
| 21:31.32 | [TK]D-Fender | jbalcomb : May God have mercy on them.... |
| 21:31.56 | jbalcomb | [TK]D-Fender i hope he starts with me cause i am now the primary contact for all phone/fax "issues" |
| 21:32.35 | jbalcomb | [TK]D-Fender i sneak around the building in hopes of avoiding the staff. |
| 21:32.59 | jbalcomb | [TK]D-Fender i use the warehouse bathroom!! |
| 21:33.11 | [TK]D-Fender | jbalcomb : If onlly you could out-source a mass clean-up to a poor Canuckian ;) |
| 21:34.12 | jbalcomb | [TK]D-Fender i gotta get through the phone consultant they hired first. Paul Winkler of PB&J Consulting (<-real company name). |
| 21:34.21 | [TK]D-Fender | lol |
| 21:34.27 | [TK]D-Fender | \I was looking at that funny.... |
| 21:34.29 | *** join/#asterisk jhiver (n=jhiver@AStDenis-105-1-4-4.w193-253.abo.wanadoo.fr) |
| 21:34.41 | jhiver | hi everybody |
| 21:34.45 | [av]bani | anyone here have a snom 360? |
| 21:34.56 | jbalcomb | [av]bani i will soon... |
| 21:35.11 | [TK]D-Fender | jhiver : Qu'est-ce que tu veut calisse?!?! ;) |
| 21:35.31 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 21:35.33 | jhiver | quoi / what? |
| 21:35.38 | [TK]D-Fender | [av]bani : tzanger does. He's of mixed opinions about it, |
| 21:35.48 | [TK]D-Fender | jhiver : Just kidding with you :) |
| 21:35.48 | _adrian | Thanks for assistance. Leaving untill I get software and server! |
| 21:36.02 | jhiver | sure but it would be best if I understood the joke :) |
| 21:36.09 | jhiver | feeling like a fool right now :) |
| 21:36.10 | *** join/#asterisk pifiu-laptop (n=someone@adsl-068-213-231-041.sip.mia.bellsouth.net) |
| 21:36.23 | Zodiacal | anyone know how i can get my cisco 7960 to show external pots lines as line appearances? is this posible? |
| 21:36.25 | *** join/#asterisk edwin_ (n=edwin@252-131-222-203.rev.techex.net.au) |
| 21:36.33 | Zodiacal | or do i have to create extentions for this that do somthing fancy? |
| 21:36.42 | *** part/#asterisk _adrian (n=adrian@user-188.lns2-c11.dsl.pol.co.uk) |
| 21:36.50 | Zodiacal | i just wanta see the status of the lines and be able to select one |
| 21:36.52 | [TK]D-Fender | Zodiacal : Doable |
| 21:36.58 | jhiver | btw [TK]D-Fender, I have done some tests and it seems that the crazy 12,000 lines dialplan works after all |
| 21:37.04 | Zodiacal | what do i use as the line name to reference my pots trunk? |
| 21:37.12 | [av]bani | [TK]D-Fender: need to know the screen size |
| 21:37.19 | [av]bani | Zodiacal: you have a 7960 there? |
| 21:37.19 | [TK]D-Fender | jhiver : I never said ti wouldn't work, I said it was the wrong way to do it :) |
| 21:37.29 | Zodiacal | [av]bani yep 7960g |
| 21:37.35 | [TK]D-Fender | bbiab |
| 21:37.35 | jhiver | it's not! it's fine :) |
| 21:37.37 | [av]bani | Zodiacal: can you tell me the lcd size? |
| 21:37.40 | jhiver | static is good man |
| 21:37.55 | Zodiacal | [av]bani umm... like 4x4 i guess |
| 21:38.00 | jhiver | I don't see any reason why it should be considered "wrong" |
| 21:38.05 | [av]bani | Zodiacal: can you check for sure :) |
| 21:38.25 | Qwell[] | [av]bani: measure me a CD, and I'll tell you the size |
| 21:38.56 | jhiver | In fact I'm uploading the latest bug fixes on CPAN now :) |
| 21:39.08 | jhiver | it's been so long since I upped something to CPAN this feels really nice |
| 21:39.22 | Zodiacal | [av]bani 3x4 exactly |
| 21:39.25 | [av]bani | Qwell[]: 4 6/8" |
| 21:39.30 | [av]bani | Zodiacal: thanks! |
| 21:39.40 | Qwell[] | 6/8"? |
| 21:39.46 | *** join/#asterisk oej (n=oej@adsl-66-143-42-162.dsl.ksc2mo.swbell.net) |
| 21:39.47 | Qwell[] | what a weird ruler you have |
| 21:40.31 | Qwell[] | oej: Hey. :) |
| 21:40.44 | oej | Hey Qwell |
| 21:40.49 | oej | On my way home, found a hotspot |
| 21:40.51 | jhiver | bbiab? what the hell does that mean |
| 21:40.52 | oej | :-) |
| 21:40.56 | Qwell[] | oej: nice |
| 21:41.14 | Zodiacal | [tk]d-fender we have 6 pots lines and 6 phones. line 6 needs to be answered with a differnt greeting than lines 1-5 by the secretaries.. do you think showing all 6 pots lines and their status on the phone is the best way to accomplish this? |
| 21:41.19 | [av]bani | Zodiacal: any idea what lcd rez the 7960 is? |
| 21:41.27 | Qwell[] | [av]bani: small |
| 21:41.29 | Qwell[] | heh |
| 21:41.35 | Zodiacal | or is there another way to maybe just flash a little icon that tells the station which greeting to use when they answer |
| 21:41.36 | [av]bani | lol |
| 21:41.56 | Zodiacal | [av]bani its pretty blocky so not very high at all |
| 21:42.07 | [av]bani | Zodiacal: 100x145 sound about right? |
| 21:42.57 | *** part/#asterisk santiago (n=santiago@208.195.215.222) |
| 21:42.58 | *** part/#asterisk }btorch{ (n=kvirc@208.63.19.172) |
| 21:43.26 | Zodiacal | 90 x 56 for the logo.bmp and that takes up like 3/4 of the screen |
| 21:43.49 | [av]bani | yea 100x145 same as 7940g then |
| 21:43.53 | Zodiacal | err 1/3 rather |
| 21:44.12 | Qwell[] | 60 is the same as the 40 |
| 21:44.19 | Qwell[] | just the extra appearances |
| 21:44.33 | Zodiacal | any ideas on how to configure my appearnaces? |
| 21:44.34 | Zodiacal | anyone |
| 21:45.21 | [av]bani | Zodiacal: prolly some xml status screen will be your only option |
| 21:45.47 | Zodiacal | nothing that will show it, with having them to press any buttons? |
| 21:46.25 | *** join/#asterisk pifiu-laptop (n=someone@adsl-068-213-231-042.sip.mia.bellsouth.net) |
| 21:47.02 | pifiu | in this line what does this mean? |
| 21:47.04 | pifiu | exten => _7XXX,1,Dial(IAX2/myserver:passwordA@IAXserverA/${EXTEN:1},30,r) |
| 21:47.05 | Zodiacal | so line apearances are just extentions? |
| 21:47.09 | pifiu | the ___7xxx |
| 21:47.21 | pifiu | that i have to dial 7 to reach the other server? |
| 21:47.41 | Zodiacal | 7 plus any other 3 numbers i think |
| 21:48.04 | Zodiacal | not 100% sure |
| 21:48.25 | pifiu | ok |
| 21:48.34 | Zodiacal | you should get one of the pdf asterisk handbooks, they show the syntax for those wildcards |
| 21:49.55 | jhiver | aaah lads I'm happy - on monday I will receive my first ISDN E1 line for landlines, yey! |
| 21:50.11 | dily | anyone use sqlite? |
| 21:50.17 | jhiver | that should open up a whole new world of possibilities (and headaches :)) |
| 21:50.32 | jhiver | dily, no I stick with MySQL usually |
| 21:50.39 | iccomputing | How many lines can you run on ISDN? |
| 21:50.50 | jhiver | it depends |
| 21:50.55 | jhiver | BRI is 2 |
| 21:50.59 | jhiver | PRI is up to 30 |
| 21:51.02 | jbalcomb | ok, Polycom IP 501 or IP 601? |
| 21:51.05 | jhiver | I'm having 30 :) |
| 21:51.36 | iccomputing | and you are using Asterisk to be the PBX? |
| 21:51.57 | jhiver | Well I will be using Asterisk for the routing and the calling card app |
| 21:52.05 | pifiu | and then waht about this part? |
| 21:52.08 | pifiu | ${EXTEN:1},30,r) |
| 21:52.14 | jhiver | which is a 'frenchized' verision of astcc |
| 21:52.18 | *** join/#asterisk Simon-_ (i=byte@proxima.arlott.org.uk) |
| 21:52.26 | pifiu | what does it mean? |
| 21:52.41 | jhiver | r means "give a ringing tone immediately" |
| 21:52.45 | Qwell[] | pifiu: Strip off the first digit |
| 21:52.46 | *** join/#asterisk _deg_ (n=deg@201.22.47.74.adsl.gvt.net.br) |
| 21:52.48 | Qwell[] | and don't use r |
| 21:53.00 | jhiver | yeah I don't really see the value of 'r' |
| 21:53.05 | Qwell[] | You're asking VERY basic questions... |
| 21:53.07 | Qwell[] | ~wikis |
| 21:53.09 | jbot | wikis is probably http://www.voip-info.org |
| 21:53.12 | jhiver | what the hell is it for anyway, apart from fooling people |
| 21:53.57 | pifiu | so just /${EXTEN},30,) |
| 21:54.02 | pifiu | the 30 means the number of seconds it will ring? |
| 21:54.15 | Qwell[] | show application dial |
| 21:54.48 | pifiu | im sorry but meaning? |
| 21:54.55 | Qwell[] | type that |
| 21:55.38 | *** join/#asterisk Simon- (i=byte@proxima.arlott.org.uk) |
| 21:55.41 | kll | what is the control mechanism for deciding whether a channel is busy or not? |
| 21:55.57 | Qwell[] | kll: quite a few things |
| 21:55.59 | jhiver | i would guess it depends on the signalling |
| 21:56.06 | Qwell[] | phone could say "go away, I don't want to talk" |
| 21:56.09 | kll | when dialing an extension that is already making a call I get a regular rining tone instead of a busy tone... |
| 21:56.10 | jhiver | it's basically the job of the signaling layer |
| 21:56.12 | Qwell[] | or asterisk could have a call limit for it |
| 21:56.36 | Qwell[] | why would you want busy anyhow? |
| 21:56.52 | Qwell[] | busy signals are so 1992 |
| 21:56.53 | kll | well the phone is busy, so why not a busy signal |
| 21:56.59 | jhiver | are you sure you don't have an ATA with more than one FXS port which accepts simultaneous calls? |
| 21:57.02 | Qwell[] | because there is something useful that you could do |
| 21:57.07 | jhiver | my fritz!fonbox does that |
| 21:57.09 | Qwell[] | like sending it to vm |
| 21:57.23 | kll | no vm |
| 21:57.38 | kll | I actually want it to be up to the user |
| 21:57.45 | Qwell[] | so let it ring |
| 21:57.55 | Qwell[] | if they don't answer...caller be damned |
| 21:57.58 | kll | but if the user wants to, asterisk should give a busy tone |
| 21:58.09 | jhiver | which user? |
| 21:58.16 | kll | some of the users are very 1992ish |
| 21:58.22 | joe | if there is a long delay before you start hearing a rign what could be the problem (handoff from a fujitxu pbx via a t1) |
| 21:58.23 | Qwell[] | well, hire new users |
| 21:58.30 | kll | jhiver: beleive it or not, I'm not actually using all the extensions myself ;) |
| 21:58.49 | kll | Qwell[]: hehe |
| 21:58.49 | jhiver | kll: you're so 1992-ish /joking |
| 21:58.54 | kll | hehe |
| 21:58.57 | *** join/#asterisk denon (i=denon@synapse.subneural.net) |
| 21:58.57 | *** mode/#asterisk [+o denon] by ChanServ |
| 21:59.04 | kll | I don't mind it just ringing, but others will |
| 21:59.09 | Qwell[] | never give users choice |
| 21:59.13 | *** part/#asterisk oej (n=oej@adsl-66-143-42-162.dsl.ksc2mo.swbell.net) |
| 21:59.17 | Qwell[] | they get what you give them |
| 21:59.23 | jhiver | kll i don't understand your problem |
| 21:59.24 | _Sam-- | that works, when you are the boss |
| 21:59.44 | kll | jhiver: I want my asterisk to send busy tones when you call an extension that is busy |
| 21:59.47 | jhiver | the phone, is it an IP phone or is it plugged to a Zap interface or channel bank? |
| 21:59.55 | kll | it's an IP phone |
| 21:59.57 | kll | well, ATA box |
| 22:00.01 | jhiver | it *should* do that out of the box |
| 22:00.15 | *** part/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 22:00.16 | jhiver | if the ATA is busy then it should send back a BUSY signal |
| 22:00.34 | kll | so it's all up to the client? |
| 22:00.48 | _Sam-- | the client is what sends its status to asterisk |
| 22:00.58 | kll | alright |
| 22:01.01 | jhiver | Well, if the client decides to say it's ringing, how can you tell it's not lying? |
| 22:01.07 | *** join/#asterisk litnimax (n=chatzill@212.0.210.237) |
| 22:01.14 | jhiver | of course it's up to the client |
| 22:01.17 | kll | jhiver: I'm not really sure |
| 22:01.22 | jhiver | in the SIP world at least |
| 22:01.23 | *** join/#asterisk [Airwolf] (n=airwolf@82-171-75-4.dsl.ip.tiscali.nl) |
| 22:01.27 | litnimax | hello folks! Anyone tried using stored procedures from app_mysql ? |
| 22:01.37 | kll | I thought that perhaps asterisk kept track of calls to different extensions |
| 22:01.45 | litnimax | simple selects do work, but stored proc seems not :-/ |
| 22:02.17 | jhiver | you should play with SER and have a look at the SIP headers using ngrep it helps understanding more things about signaling :) |
| 22:02.30 | kll | ah |
| 22:02.32 | kll | :) |
| 22:02.40 | jhiver | kll |
| 22:02.59 | jhiver | why shouldn't an extension be able to handle multiple calls? |
| 22:03.14 | kll | jhiver: it should |
| 22:03.24 | jhiver | well then |
| 22:03.41 | jhiver | why should asterisk keep track on how many channels are directed to such or such extension? |
| 22:03.50 | kll | hehe, I'm not sure |
| 22:03.53 | kll | it was just an idea |
| 22:03.56 | jhiver | unless you know the capacity of the extension in terms of "lines", it's pointless |
| 22:04.14 | jhiver | which is why you have 'RINGING', 'BUSY', and such signals |
| 22:04.17 | kll | well perhaps the numbers of lines was negiotated at registration |
| 22:04.24 | jhiver | :) |
| 22:04.29 | kll | I was just making wild guesses |
| 22:04.38 | kll | since I'm unsure of how everything work |
| 22:04.38 | kll | s |
| 22:04.49 | jhiver | I would say, check your ATA and if you see no options ask the manufacturer |
| 22:05.08 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 22:05.10 | kll | jhiver: yeah, I'll do that |
| 22:05.12 | kll | thanks |
| 22:05.19 | jhiver | it's prolly somewhere in your ATA config I would think |
| 22:05.43 | jhiver | if you want to check if it's asterisk, unhook your landline phone, and make asterisk call it |
| 22:05.50 | jhiver | see if you get your busy tone :) |
| 22:05.59 | kll | nope, it's confirmed |
| 22:06.03 | kll | just tried another box |
| 22:06.10 | generalhan | whats up all ! ? |
| 22:06.23 | jhiver | Plenty of things! |
| 22:06.32 | jhiver | http://search.cpan.org/~jhiver/Asterisk-LCR/lib/Asterisk/LCR.pm |
| 22:06.35 | jhiver | this, for a start |
| 22:06.50 | kll | jhiver: it's something with the box... I'll dig into it some day |
| 22:06.56 | jhiver | And as soon as I have by bloody PRI I'll be jumping everywhere! |
| 22:07.04 | jhiver | it was about bloody time :) |
| 22:07.04 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 22:07.08 | generalhan | Does anyone know if there is a way to turn off call waiting for specific phones, not for the whole system ???? |
| 22:07.10 | sivana | *sigh* |
| 22:07.13 | Traderzz | does 1.2.2 fix the issue where a voip call comes in over voip and the message is already played a few seconds before the caller starts to hear anything? |
| 22:07.25 | sivana | ZT_SPANCONFIG failed on span 1: No such device or address (6) <-- does this mean not found? |
| 22:07.31 | Traderzz | this occurs even with the wait command in place |
| 22:07.32 | Qwell[] | Traderzz: That isn't really an "issue". Use a Wait() |
| 22:07.39 | Traderzz | the wait doesnt fix it |
| 22:07.44 | jaike | traderz: i do playback(silence) |
| 22:07.48 | Qwell[] | What is the bug number? |
| 22:07.51 | jaike | wait doenst solve it |
| 22:08.11 | Traderzz | jaike, do you understand the issue i am getting? |
| 22:08.30 | jhiver | Call waiting indication: Implemented in Asterisk, but must be support on the phone |
| 22:08.31 | jaike | yup...1 or 2 secs of audio not heard |
| 22:08.38 | Traderzz | yes |
| 22:08.41 | jaike | i tried wait..it didnt work |
| 22:08.46 | jhiver | tip: disactivate call waiting on your phone ;) |
| 22:08.47 | Traderzz | yes.. any ideas on how to fix it ? |
| 22:09.01 | jaike | Playback(silence)..........play the silence.gsm file |
| 22:09.07 | jhiver | I think people think asterisk does a lot more than it actually does :) |
| 22:09.09 | jaike | before playing the next file |
| 22:09.38 | Traderzz | jaike, the issue is that then the time it answers properly its dead air |
| 22:09.58 | Traderzz | then the time it doesnt its played a few seconds |
| 22:10.47 | jhiver | Traderzz, this is strange |
| 22:10.57 | jhiver | do you have the same thing when connecting two extensions? |
| 22:11.02 | Traderzz | no |
| 22:11.03 | jaike | hmm |
| 22:11.26 | jhiver | do you have a timing device in the box? |
| 22:11.39 | jhiver | (just making wiiiild guesses) |
| 22:11.39 | Traderzz | not sure |
| 22:11.40 | jhiver | cause I don't have this issue at all |
| 22:11.47 | jhiver | but I have a TDM400P in the box |
| 22:12.04 | Traderzz | i dont have any cards in it |
| 22:12.07 | Traderzz | straight voip |
| 22:12.13 | Traderzz | no tdm or psdn |
| 22:12.15 | Traderzz | no tdm or pstn |
| 22:12.30 | jhiver | and what version of the kernel are you using? 2.4 had severe issues with timing |
| 22:12.30 | Traderzz | phone to phone over sip is perfect |
| 22:12.32 | jaike | you have a welcome message but its the 3rd or 4th word that is heard by the caller? |
| 22:12.39 | Traderzz | yes |
| 22:12.46 | Traderzz | yes its already started |
| 22:13.02 | ManxPowe | do an Answer then a Wait(2) before the Playback or Backgrounf |
| 22:13.03 | Traderzz | i need to get this fixed |
| 22:13.05 | ManxPowe | s/f/d |
| 22:13.15 | jaike | traderzz: i recorded a 2 second silent.wav file...played it before the welcome message |
| 22:13.21 | jpablo | hey people im seeing a lot of reboots after loading the wct4xxp driver, any ideas what might be going wrong ? |
| 22:13.22 | jaike | that fixed it |
| 22:13.37 | Traderzz | ok can you send it to me? |
| 22:13.39 | jpablo | (i load wct4xxp and the machine reboots automatically) |
| 22:14.00 | jhiver | ManxPowe, s/f/d? sex for dollars? |
| 22:14.07 | jaike | pm me your email |
| 22:14.10 | jhiver | w00t? |
| 22:14.35 | jhiver | sorry / for / drinking? |
| 22:14.41 | jhiver | just kidding :) |
| 22:14.43 | Traderzz | sex for drugs |
| 22:14.49 | jhiver | aaah indeed :) |
| 22:14.56 | jhiver | silly me :) |
| 22:15.03 | Traderzz | scifi for dummie |
| 22:15.09 | Qwell[] | I wanna go home |
| 22:15.18 | Qwell[] | mog_work: It's your fault. :( |
| 22:15.19 | jhiver | Qwell, you're not there? |
| 22:15.25 | jhiver | working I guess? |
| 22:15.28 | Qwell[] | jhiver Qwell is |
| 22:15.28 | *** join/#asterisk __deg__ (n=deg@201.22.22.27.adsl.gvt.net.br) |
| 22:15.36 | pifiu | qwell i have another question just for you |
| 22:16.02 | mog_work | yeah it is |
| 22:16.05 | jhiver | oh cause Qwell and Qwell[] aren't the same or something :) |
| 22:16.09 | mog_work | wanna fight about it ^_^ |
| 22:16.16 | Qwell[] | mog_work: You'd probably win. :D |
| 22:16.20 | jhiver | one is idler and the other one pretends he's working :) |
| 22:16.26 | Qwell[] | jhiver: correct |
| 22:16.33 | jhiver | I see |
| 22:16.36 | *** join/#asterisk greendisease (n=jack@fedora/greendisease) |
| 22:16.58 | Qwell[] | mog_work: Why don't you have a digium/asterisk mask? |
| 22:17.00 | jhiver | So Qwell[] what fascinating job do you do? |
| 22:17.10 | Qwell[] | jhiver: programmer for a bank |
| 22:17.17 | Qwell[] | s/mask/cloak/ |
| 22:17.24 | Qwell[] | umm, yeah |
| 22:17.34 | jhiver | hey that sounds good - that's where the money is :) |
| 22:17.39 | *** join/#asterisk dudes (n=dudes@12-215-33-205.client.mchsi.com) |
| 22:17.41 | Qwell[] | I wish |
| 22:17.45 | jhiver | now all you need to do is find a way to take it :) |
| 22:17.59 | *** part/#asterisk iccomputing (n=Wireless@cpe-71-65-112-38.woh.res.rr.com) |
| 22:18.04 | Qwell[] | I work right next to the people who count all the money from the ATMs... |
| 22:18.10 | Qwell[] | comes in by the truckload |
| 22:18.26 | *** join/#asterisk smallb (n=smallb@prox47-249.trinidad.net) |
| 22:18.30 | smallb | hi all |
| 22:18.34 | jhiver | looks cool... I'm sure there's a way to jack it :) |
| 22:18.35 | smallb | any ooh323 users ? |
| 22:18.50 | jhiver | nope, it's SIP all the way for me |
| 22:18.52 | Qwell[] | meh...8 figures, orit isn't work it |
| 22:19.06 | jhiver | Qwell[] agreed :) |
| 22:19.23 | smallb | it seems that most people don't like h323 for one reason or another |
| 22:19.39 | jhiver | It Doesn't Work out of the box with Asterisk :) |
| 22:19.41 | generalhan | Ok then, How would i turn off Call Waiting on a specific phone if there was no way to turn it off from the phone itself ??? |
| 22:19.51 | jpablo | a lo of people hate h323 because of the nightmare open h323 is |
| 22:20.04 | jhiver | generalhan, don't know |
| 22:20.15 | Traderzz | 2.6 kernel here |
| 22:20.18 | jhiver | what phone is it? |
| 22:20.25 | generalhan | so there isnt a way to do like a "callwaiting=no" in sip.conf or something like that ? |
| 22:20.29 | generalhan | its an Aastra 9112i |
| 22:20.30 | smallb | yeah, i tried building it several times to no avail |
| 22:21.17 | generalhan | well let me tell you my issue, maybe there is a different way to set up my dial plan so that this doesnt happen. |
| 22:21.34 | X-Files | eyebeam support VIDEO stream and Message in asterisk ??????? |
| 22:22.05 | jhiver | I don't think call waiting has anything to do with Asterisk at all, but I could be wrong |
| 22:22.08 | generalhan | I have 20 sales reps on these phones. callers dialing in go to a queue that rings all 20 phones. when some one is on the phone it still rings to them and they hear the non-stop beeping in their ear that makes it very difficult to hear |
| 22:22.26 | jhiver | it seems to be that call waiting is just a phone which says 'RINGING' and beeps you when it does... |
| 22:22.47 | jhiver | what phone is it? |
| 22:22.51 | jhiver | ah ok |
| 22:22.54 | generalhan | Aastra 9112i SIP Phone |
| 22:23.02 | *** join/#asterisk tomben (n=tomben@fw01.ext.atl.jboss.com) |
| 22:23.30 | generalhan | so thats my issue, if there was a way for the queue to see if their phone was being used and not ring to them, than that would work too |
| 22:23.48 | jhiver | http://www.netxusa.com/products/Sayson/docs/9112%20Admin_%20Guide%201.2.1.1002.pdf <--- have you checked the admin guide? |
| 22:24.16 | generalhan | well i just got off the phone with Aastra and they told me that it cant be done at all for these phones |
| 22:24.55 | Qwell[] | just set a call-limit for the peer |
| 22:25.11 | tomben | Hello all, anyone ever used a Digium TDM400P with a dell 2850? |
| 22:25.12 | generalhan | Qwell[]: what do you mean ? |
| 22:25.16 | Qwell[] | call-limit |
| 22:25.21 | generalhan | Qwell[]: how do i do that ? |
| 22:25.23 | Qwell[] | it's in the sip sample config |
| 22:25.47 | generalhan | Qwell[]: thanks ill take a look at it. so that will mean that they can only take one call from the queue at a time ? |
| 22:25.55 | jhiver | does call limit works with Queues? |
| 22:25.57 | jhiver | mhhh |
| 22:26.03 | Qwell[] | no, it means they can only take one call at a time, period |
| 22:26.09 | generalhan | hmm |
| 22:26.19 | Qwell[] | it's all or nothing...you have to decide |
| 22:26.19 | jhiver | apparently no |
| 22:26.21 | jhiver | http://bugs.digium.com/view.php?id=6111 |
| 22:26.58 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-191-212.38-151.net24.it) |
| 22:27.02 | generalhan | crap |
| 22:27.08 | generalhan | back to square one |
| 22:27.21 | jaike | better read the admin guide...had the same problems with polycoms |
| 22:27.29 | jaike | admin guide saved my life |
| 22:27.35 | Qwell[] | no, it works, that bug isn't named quite right |
| 22:28.22 | kll | anyone know what to set restrictcid to in SIP RealTime if I want it to prohibit CID? |
| 22:28.49 | generalhan | well at the end they say they fixed it by changing a line in the code |
| 22:28.54 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 22:29.11 | jhiver | generalhan, |
| 22:29.14 | *** join/#asterisk MatsK (n=mk@c213-100-73-227.swipnet.se) |
| 22:29.17 | generalhan | jhiver, |
| 22:29.21 | jhiver | why don't you use call forwarding feature of the astra? |
| 22:29.26 | jhiver | I'm just checking the manual now |
| 22:29.29 | generalhan | how so ? |
| 22:29.39 | jhiver | you have the option to forward calls when busy |
| 22:29.51 | jhiver | just forward them to an extension which returns Busy() |
| 22:29.52 | generalhan | where do i forward them to ? |
| 22:29.57 | generalhan | hmm |
| 22:30.14 | generalhan | but .... doesnt that then take that person out of the queue ? |
| 22:30.19 | Qwell[] | might |
| 22:30.27 | generalhan | then they have to start at the end of the line again' |
| 22:30.30 | jhiver | well you said all phones were ringing simultaneously |
| 22:30.47 | jhiver | so it will only do it if all of them say 'busy', right? |
| 22:31.00 | Qwell[] | the phone might do an implicit answer |
| 22:31.02 | generalhan | yes, but in the case where all 20 are on the phone the 10 other people waiting they will just shuffle in line until people get off the phone |
| 22:31.03 | Qwell[] | never know |
| 22:31.27 | jhiver | don't know |
| 22:31.36 | generalhan | ill have to turn the announcements off, people will be PISSED if they hear "you are next in line" then a minute later " you are number 12 in line" |
| 22:31.45 | generalhan | lol |
| 22:31.45 | Qwell[] | try it |
| 22:31.46 | jhiver | you would think that queues would be able to handle all phones being busy wouldn't you? |
| 22:32.00 | Qwell[] | jhiver: It isn't the phone that's busy though |
| 22:32.05 | Qwell[] | the phone may very well answer, then transfer |
| 22:32.12 | generalhan | thats what i think |
| 22:32.13 | jhiver | well |
| 22:32.22 | Qwell[] | try it |
| 22:32.22 | jhiver | usually that's not referred as 'forwarding' |
| 22:32.28 | De_Mon | grr argh |
| 22:32.38 | De_Mon | why would I want to use playback instead of background? |
| 22:32.39 | jhiver | so unless the forwarding implementation is utterly shite, it should work |
| 22:32.45 | X-Files | eyebeam support VIDEO stream and Message in asterisk ??????? |
| 22:32.46 | generalhan | just like a forwarding feature on any regular phone line it still shows that 15000 calls came in on that number and you get a bill for 15000 1 minute calls |
| 22:32.55 | De_Mon | when would I NOT want to listen for keys? |
| 22:33.09 | De_Mon | X-Files video and voice? yes |
| 22:33.15 | Qwell[] | De_Mon: Status messages |
| 22:33.26 | Qwell[] | "The system WILL be going down in 5 minutes." |
| 22:33.35 | De_Mon | hehe I see |
| 22:33.36 | jhiver | yeah, but a 'regular phone' is not a SIP phone |
| 22:33.40 | X-Files | De_Mon: voice i know work. but Message and Status Online ? |
| 22:33.42 | generalhan | well true |
| 22:34.08 | jhiver | a 'regular phone' has no way of saying 'errr actually can you forward this call <there>, while a SIP phone is perfectly capable of doing that |
| 22:34.13 | De_Mon | X-Files I'm pretty sure status online works, duno about this 'message' you speak of |
| 22:34.58 | BladeRunner05 | Get error compiling mpg123-0.59r with asterisk 1.2.2 |
| 22:35.14 | Qwell[] | BladeRunner05: 64 bit? |
| 22:35.24 | jhiver | generalhan, I think you might want to try it at any rate |
| 22:35.33 | BladeRunner05 | Qwell[]: no |
| 22:36.04 | generalhan | jhiver: the call limit or the forwarding ? |
| 22:36.08 | Qwell[] | both |
| 22:36.11 | *** join/#asterisk phos-phoros (n=James@unaffiliated/phos-phoros) |
| 22:36.12 | generalhan | lol |
| 22:36.17 | generalhan | im looking for the call limit stuff now |
| 22:36.22 | jhiver | whichever works :) |
| 22:36.45 | jhiver | it's cool, you had a problem and now at least two potential solutions :) |
| 22:36.45 | BladeRunner05 | The error is http://pastebin.com/515306 |
| 22:37.05 | generalhan | and thank you both so much, i cant take much more of them complaining to me about beeps in their ear ! lol |
| 22:37.28 | Qwell[] | BladeRunner05: What did you type? |
| 22:37.28 | jhiver | well it's certainly understandable |
| 22:37.35 | De_Mon | X-Files make sure allowvideo=yes is uncommented |
| 22:37.41 | De_Mon | X-Files in sip.conf |
| 22:37.42 | jhiver | doing this kind of help desk job is shit enough as it is :) |
| 22:37.43 | X-Files | hm |
| 22:37.44 | X-Files | wait |
| 22:37.48 | BladeRunner05 | Qwell: what u mean ? |
| 22:37.48 | jhiver | no need to make it more painful :) |
| 22:37.55 | Qwell[] | BladeRunner05: To install it |
| 22:38.13 | BladeRunner05 | qwell: I run make linux |
| 22:38.26 | Qwell[] | just type make |
| 22:38.29 | jhiver | BTW, has anybody already managed to get a cirpak sending calls through Asterisk using SIP? |
| 22:38.52 | BladeRunner05 | qwell: if I run make it show me a list of make's parameters |
| 22:38.53 | jhiver | I tried with one of my customers... |
| 22:39.01 | jhiver | cirpak -> asterisk would not work |
| 22:39.07 | X-Files | De_Mon: not work :( maybe i can paste to pastebin my sip config ? |
| 22:39.08 | jhiver | but cirpak -> ser -> asterisk works fine |
| 22:39.09 | zoa | sjhiver: im doing consultancy on it |
| 22:39.12 | jhiver | strange... |
| 22:39.19 | zoa | jhiver, i got it to work |
| 22:39.26 | jhiver | oh cool |
| 22:39.29 | zoa | but the ser should not help |
| 22:39.34 | jhiver | it does |
| 22:39.41 | jhiver | works perfectly by putting ser in the middle |
| 22:39.42 | De_Mon | X-Files ya paste a link in channel |
| 22:39.43 | zoa | it worked for us, but had no ringtone |
| 22:39.49 | zoa | ringbacktone |
| 22:39.57 | X-Files | De_Mon: ok, wait |
| 22:40.00 | jhiver | I see |
| 22:40.04 | De_Mon | X-Files can you see yourself in the drawing window before making a call? |
| 22:40.14 | zoa | so, if you dont have a ringtone, contact us on support@asteriskguru.com |
| 22:40.21 | jhiver | so zoa have you got a solution for this? are you selling it or something? |
| 22:40.22 | De_Mon | arg, my session keeps timing out cuz I'm talking on irc :P |
| 22:40.55 | zoa | you need to patch asterisk for it |
| 22:40.56 | zoa | bigtime |
| 22:41.00 | jhiver | OK :) |
| 22:41.03 | zoa | put timers in there |
| 22:41.38 | jhiver | can you send me a little message at jhiver@ykoz.net? |
| 22:42.03 | jhiver | I was helping the client but if they want to pay (and I know your price) then I will pass that cost onto them |
| 22:42.22 | zoa | no need to pay me (unless i need to do it for them) |
| 22:42.47 | jhiver | well if you do work of course you need to be paid :) |
| 22:43.04 | jhiver | I'll try to see what value this has for them |
| 22:43.27 | jhiver | but they said it worked when I used SER |
| 22:43.27 | zoa | maybe he has ringback on the pstn side |
| 22:43.33 | zoa | or maybe its not using pstn at all |
| 22:43.59 | jhiver | Well it was IP phone -> cirpak -> SER -> asterisk -> PSTN -> my mobile test |
| 22:44.13 | jhiver | fantastic quality as well :) |
| 22:44.37 | X-Files | De_Mon: yes. |
| 22:44.49 | zoa | aha, then they wont have a problem |
| 22:45.02 | zoa | i noticed a problem on pstn -> cirpak -> asterisk |
| 22:45.06 | X-Files | De_Mon: http://pastebin.ca/37698 <<- sip.conf and extensions.conf |
| 22:45.09 | jhiver | ah ok |
| 22:45.13 | zoa | patch will be put on mantis for reference |
| 22:45.31 | jhiver | so what is the problem exactly in this configuration? |
| 22:45.58 | zoa | asterisk doesnt send audio when no rtp is received |
| 22:46.14 | zoa | and cirpak doesnt send any rtp when the call is still ringing |
| 22:46.23 | jhiver | hang on |
| 22:46.27 | zoa | nor likes the ringing (it wants it played) |
| 22:46.39 | jhiver | I'm not sure to understand |
| 22:46.53 | zoa | so the pstn end heard no ringback tone while calling the other end |
| 22:47.02 | zoa | but sip -> pstn was not a problem |
| 22:47.05 | zoa | only pstn -> sip |
| 22:47.24 | jhiver | still |
| 22:47.31 | jhiver | isn't ringing just signalling? |
| 22:47.37 | jhiver | what's the need for RTP? |
| 22:47.37 | zoa | it doesnt have to be |
| 22:47.51 | zoa | when it goes to pstn, something has to generate the tone |
| 22:47.55 | zoa | and the cirpak didnt do it |
| 22:48.02 | jhiver | I thought it was coming from the PSTN |
| 22:48.04 | De_Mon | X-Files I don't see anything wrong in those configs. -- can you see yourself in the drawing window before making a call? |
| 22:48.15 | De_Mon | oh you said yes |
| 22:48.20 | jhiver | you said PSTN -> Cirpak -> Asterisk |
| 22:48.24 | zoa | yes |
| 22:48.24 | X-Files | De_Mon: =) yes yes |
| 22:48.28 | De_Mon | X-Files getting any errors at the CLI? |
| 22:48.35 | zoa | so in this case asterisk had to send a ringing tone to the cirpak |
| 22:48.36 | X-Files | De_Mon: no |
| 22:48.45 | zoa | otherwise the pstn caller would not hear a ringback tone |
| 22:48.53 | De_Mon | X-Files can you get just voice to work? |
| 22:48.58 | jhiver | mhhh |
| 22:48.58 | X-Files | voice work |
| 22:49.02 | zoa | probably could be fixed on the cirpak, but my client had no access to that |
| 22:49.07 | zoa | i need to go |
| 22:49.08 | zoa | sleep |
| 22:49.09 | zoa | :) |
| 22:49.21 | jhiver | ok speak to you later |
| 22:49.24 | De_Mon | X-Files duno, it should be working based on my experience :) |
| 22:49.29 | jhiver | i'm still unsure |
| 22:49.33 | X-Files | De_Mon: maybe put debug to pastebin ? |
| 22:49.39 | jhiver | to understand this ringback tone malarki properly :) |
| 22:51.11 | *** join/#asterisk Sniper00X (n=sniper00@ool-44c061a7.dyn.optonline.net) |
| 22:51.12 | De_Mon | will s,1,background(playbackfile) s,2,Read(VAR) do the same thing as s,1,Read(VAR|playbackfile), without waiting for the playbackfile to end? |
| 22:53.34 | X-Files | De_Mon: check it http://pastebin.ca/37701 |
| 22:53.53 | X-Files | De_Mon: this log from debug asterisk |
| 22:54.01 | De_Mon | X-Files I duno how to translate asterisk debug logs |
| 22:56.27 | Sniper00X | anyone knows if asterisk would work with a netgear WGR826V and cablevision optimum voice .. netgear acting as a pstn? |
| 23:02.56 | *** join/#asterisk thosa (n=thosa@p5487BB17.dip0.t-ipconnect.de) |
| 23:03.32 | *** join/#asterisk pifiu (n=myassisb@208.205.181.170) |
| 23:04.09 | pifiu | hey qwell i got disconnected sorry |
| 23:06.01 | ManxPowe | Sniper00X, What protocol does that device use? |
| 23:06.06 | *** join/#asterisk Mark_Halverson (n=mhlvrs@67-139-119-152.dsl1.pco.ca.frontiernet.net) |
| 23:07.35 | Sniper00X | SIP |
| 23:08.34 | Qwell[] | okay, this Brian Bell guy on the -dev list...is an idiot |
| 23:09.05 | wunderkin | must not read any of the emails |
| 23:09.08 | MikeJ[Laptop] | well that's not very nice |
| 23:09.16 | MikeJ[Laptop] | :D |
| 23:09.24 | Qwell[] | MikeJ[Laptop]: Read his message. You'll agree :P |
| 23:09.30 | drumkilla | LOL |
| 23:09.33 | drumkilla | that's awesome |
| 23:09.44 | Qwell[] | I was polite as possible... |
| 23:09.48 | drumkilla | I love how the third line tells you how to unsubscribe |
| 23:09.54 | Qwell[] | drumkilla: See my response :p |
| 23:10.17 | *** join/#asterisk pifiu-laptop (n=someone@adsl-068-213-231-041.sip.mia.bellsouth.net) |
| 23:10.19 | Qwell[] | I was holding WAYYY back...I wanted to tear into him |
| 23:10.32 | Qwell[] | so, I'll talk shit here instead :p |
| 23:10.41 | X-Files | De_Mon: u there ? |
| 23:10.51 | drumkilla | he won't get the response until his next digest |
| 23:10.54 | drumkilla | which he probably won't read |
| 23:10.56 | drumkilla | ..... |
| 23:10.56 | Qwell[] | yeah |
| 23:11.00 | Qwell[] | ironic |
| 23:11.09 | *** join/#asterisk cyburdine (n=cyburdin@208.2.145.2) |
| 23:11.40 | Qwell[] | drumkilla: You like formatting fixes, right? :) |
| 23:12.07 | Qwell[] | should take a look at 6300 |
| 23:12.10 | drumkilla | lol |
| 23:12.13 | drumkilla | oh, i love them |
| 23:12.22 | pifiu | http://pastebin.ca/37685 what does that mean? and how can i get around it? |
| 23:12.43 | Qwell[] | pifiu: Add a peer named name |
| 23:12.44 | Mark_Halverson | anyone know of any good voip exchanges? |
| 23:12.57 | pifiu | in extensions? |
| 23:13.02 | Qwell[] | in iax.conf |
| 23:13.13 | Qwell[] | iax2.conf? whatever |
| 23:13.14 | pifiu | i already have one |
| 23:13.25 | pifiu | let me double check everything |
| 23:14.02 | drumkilla | Qwell[]: is it purely formatting? |
| 23:14.10 | Qwell[] | and the small doxygen changes |
| 23:14.20 | Qwell[] | alaw had doxygen comments ulaw didn't |
| 23:14.23 | drumkilla | k |
| 23:14.24 | X-Files | grr |
| 23:14.36 | drumkilla | Qwell[]: you could have put these in a branch :) |
| 23:14.43 | Qwell[] | I can |
| 23:14.48 | Qwell[] | it'll take me like two seconds |
| 23:14.54 | drumkilla | nah |
| 23:14.55 | drumkilla | no need |
| 23:14.56 | Qwell[] | ok |
| 23:14.59 | drumkilla | too late now :) |
| 23:15.01 | Qwell[] | heh |
| 23:15.29 | Qwell[] | only reason I put them up in patch form, was so they could be reviewed easier, and seperately |
| 23:15.31 | X-Files | Ppls, Please help. I use asterisk CVS version and eyeBeam ! I can't see Video stream , voice worked ! my configure files : http://pastebin.ca/37698 <<- sip.conf and extensions.conf , debug file : http://pastebin.ca/37701 |
| 23:16.16 | *** join/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 23:16.21 | anarcat | hello |
| 23:16.41 | anarcat | ^^Gu[L]Can, stop spamming. |
| 23:16.47 | anarcat | anyways |
| 23:16.55 | drumkilla | Qwell[]: i'm just going to do a quick 'svn diff -uw' :) |
| 23:16.55 | Qwell[] | What's he spamming? |
| 23:16.58 | anarcat | anyone has experience of asterisk in a vserver? |
| 23:17.02 | anarcat | Qwell[], me. in colors. |
| 23:17.06 | Qwell[] | drumkilla: yeah, that would have worked, heh |
| 23:17.17 | anarcat | i'm having problems starting asterisk in a vserver |
| 23:17.25 | anarcat | Jan 20 23:14:37 WARNING[28021]: Failed to bind to 64.15.133.226:2727: Address already in use |
| 23:17.27 | anarcat | Jan 20 23:14:37 WARNING[28021]: Unable to open IAX timing interface: Permission denied |
| 23:17.31 | anarcat | Jan 20 23:14:40 ERROR[28021]: Unable to bind to 64.15.133.226 port 4569: Address already in use |
| 23:17.37 | Qwell[] | ooo, no -w for svn diff |
| 23:17.38 | drumkilla | use a different port ....... |
| 23:17.44 | Qwell[] | port? |
| 23:17.45 | drumkilla | Qwell[]: you serious? |
| 23:17.47 | Qwell[] | erm |
| 23:17.49 | Qwell[] | yeah |
| 23:18.04 | drumkilla | i bet there is a way! |
| 23:18.08 | Qwell[] | probably with 0x |
| 23:18.08 | drumkilla | some how, some way! |
| 23:18.09 | Qwell[] | -x |
| 23:18.18 | anarcat | drumkilla, i don't think that's the problem |
| 23:18.21 | anarcat | Jan 20 23:18:01 WARNING[28772]: Unable to open IAX timing interface: Permission denied |
| 23:18.21 | Qwell[] | ha, nope |
| 23:18.24 | anarcat | that's not good |
| 23:18.27 | Qwell[] | svn: '-w' is not supported |
| 23:18.31 | *** part/#asterisk twisted[asteria] (n=twisted@asterisk/friend-and-developer/pdpc.professional.twisted) |
| 23:19.08 | Qwell[] | drumkilla: would probably have to diff against a clean trunk |
| 23:19.16 | Qwell[] | manual diff |
| 23:19.19 | drumkilla | this is annoying |
| 23:19.46 | Qwell[] | ahh |
| 23:19.51 | Qwell[] | --diff-cmd diff -x -w |
| 23:19.54 | Qwell[] | -x -uw |
| 23:20.29 | drumkilla | yay |
| 23:20.32 | drumkilla | thanks :) |
| 23:20.35 | Qwell[] | That's still a long patch, heh |
| 23:20.37 | generalhan | Qwell[]: all i see in the sample for call limit it incominglimit, but that is described as how many OUTGOING calls at a time ? wheres the incoming one ? lol |
| 23:20.53 | Qwell[] | outgoing TO the phone |
| 23:20.58 | generalhan | i see |
| 23:21.03 | generalhan | ok lets try that out ! thanks ! |
| 23:21.11 | [av]bani | anyone know how to unlock packet8's uniden UIP1868P? |
| 23:22.55 | *** join/#asterisk dsfr (n=dsfr@gateway.digium.com) |
| 23:23.33 | drumkilla | Qwell[]: done |
| 23:23.43 | Qwell[] | you rock |
| 23:25.11 | pifiu | ok so wtf the user is created |
| 23:25.13 | pifiu | and still doenst work |
| 23:25.20 | Qwell[] | pifiu: created correctly? |
| 23:25.25 | Qwell[] | pastebin it |
| 23:25.41 | Qwell[] | put the error(s) in there again too |
| 23:28.25 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 23:28.26 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 23:28.31 | Ariel_ | hello everyone |
| 23:30.00 | [TK]D-Fender | [av]bani : I wouldn't touch it... then again I dislike the UIP-200.... |
| 23:30.28 | Ariel_ | ahh talk about phones.... one word....Polycom's |
| 23:30.43 | [TK]D-Fender | Ariel_ : Find your own choir :D |
| 23:30.56 | Ariel_ | [TK]D-Fender, hehehe |
| 23:31.59 | X-Files | Ppls, why not work Status Online users in EYEBEAM ??? |
| 23:33.38 | *** join/#asterisk pr0m (n=pr0methe@24-75-196-70.chvlva.adelphia.net) |
| 23:34.59 | *** join/#asterisk [hC] (i=turnerd@66.199.130.40) |
| 23:35.16 | [hC] | Is there any way to tune down the number of rings it takes this tdm400p to answer a call? |
| 23:35.24 | [hC] | it usually takes 2-3 rings before asterisk decides to pick it up |
| 23:35.28 | X-Files | Jan 21 01:40:25 NOTICE[14960]: chan_sip.c:11133 handle_request: Unknown SIP command 'PUBLISH' from 'x.x.x.x' |
| 23:35.41 | X-Files | hmm why this notice ? |
| 23:35.47 | pifiu | qwell still nothing. =( |
| 23:36.34 | Qwell[] | [hC]: Do you have cid on the line? |
| 23:36.53 | [hC] | Qwell: if i turn callerid=no will it reduce the time? |
| 23:36.54 | *** join/#asterisk Dr-Linux (n=nah@202.59.75.58) |
| 23:37.00 | [hC] | Qwell:I presume it waits so it can pick up cid.. |
| 23:37.05 | Qwell[] | yep |
| 23:37.08 | [hC] | 10-4 |
| 23:37.13 | [hC] | makes sense. |
| 23:37.21 | [hC] | thanks |
| 23:37.31 | [hC] | im picking up two more tdm400's since i think my first one is toast |
| 23:37.36 | *** part/#asterisk anarcat (n=anarcat@pdpc/supporter/active/anarcat) |
| 23:37.36 | Dr-Linux | anybody familiar with Asterisk AGI with Java? |
| 23:37.37 | hugo-v6 | X-Files: snom phone? |
| 23:37.38 | [hC] | well. transmit on it is toast. |
| 23:37.48 | denon | lets convert em all to ISDN! |
| 23:37.54 | X-Files | hugo-v6: no, soft phone - eyebeam |
| 23:38.27 | hugo-v6 | X-Files: sorry then. dunno this soft. |
| 23:38.47 | X-Files | ;) |
| 23:39.44 | *** join/#asterisk thosa (n=thosa@p54879931.dip0.t-ipconnect.de) |
| 23:39.55 | *** part/#asterisk thosa (n=thosa@p54879931.dip0.t-ipconnect.de) |
| 23:40.20 | hugo-v6 | X-Files: but fyi u can ignore this warning |
| 23:40.55 | X-Files | hugo-v6: but, i wanna see status users online/busy/offline |
| 23:40.58 | *** join/#asterisk Current (i=_niLgun_@62.162.14.50) |
| 23:41.39 | *** join/#asterisk BladeRunner05 (n=feelme@adsl-222-217.37-151.net24.it) |
| 23:42.32 | De_Mon | if I want a good cheap phone for asterisk, polycom the way to go? |
| 23:42.40 | *** join/#asterisk RoyK (n=roy@host-81-191-147-248.bluecom.no) |
| 23:43.06 | X-Files | De_Mon: why me not work status contacts online, away or offline ? |
| 23:43.25 | generalhan | Qwell[]: Thank you soo much. the incominglimit worked perfectly!! i just tested it out and everyone loves me again !! LOL |
| 23:43.26 | jaike | de_mon: wouldnt exactly call them cheap |
| 23:43.33 | jaike | most bang for the buck probably |
| 23:44.44 | BladeRunner05 | someone has experienced with ast gui client+ vicidial on asterisk 1.2.2 ? |
| 23:44.51 | De_Mon | oh, now that I'm looking I have yet to see any prices |
| 23:45.15 | Mark_Halverson | BladeRunner: try #gnudialer channel |
| 23:45.33 | BladeRunner05 | mark: I try |
| 23:45.47 | hugo-v6 | beside that imho look the polycom phones like crap. (well teh 50x is 'ok') |
| 23:46.36 | Ariel_ | hugo-v6, yes looks are strange but they work great and sound great. |
| 23:46.39 | X-Files | Please, ppls, help, why i can't see status contact in eyebeam ? |
| 23:47.22 | jaike | x-files: asterisk sip probably doesnt support PUBLISH |
| 23:47.57 | X-Files | jaike: hmm, but asterisk presence support... |
| 23:48.25 | X-Files | jaike: from voip-info.org |
| 23:48.26 | X-Files | Phones known to work with the current implementation of SIP Presence |
| 23:48.26 | X-Files | Snom (various models) |
| 23:48.26 | X-Files | Polycom IP30x/IP50x/IP600 |
| 23:48.26 | X-Files | Xten EyeBeam |
| 23:48.26 | X-Files | Grandstream GXP2000 (Firmware >= 1.0.1.13) |
| 23:49.01 | Qwell[] | X-Files: You need to setup presense on asterisk, and eyebeam needs to subscribe to it |
| 23:49.43 | X-Files | Qwell: setup ? check it http://pastebin.ca/37698 <<- sip.conf and extensions.conf |
| 23:49.52 | hugo-v6 | Ariel_: someday if i have to much money ill buy one and see for myself ;) |
| 23:50.02 | Ariel_ | X-Files, you need to use hint setup for that to work |
| 23:50.19 | Ariel_ | hugo-v6, there less money then the snom |
| 23:50.27 | X-Files | Ariel_: check it : http://pastebin.ca/37698 <<- sip.conf and extensions.conf |
| 23:52.10 | De_Mon | Qwell[] i think what hes trying to say, is that he has hints setup |
| 23:53.37 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-43-124.cybersurf.com) |
| 23:54.01 | De_Mon | if I do s,1,Festival('Goodbye'); s,2,Hangup All I hear on the phone is "goo" |
| 23:54.01 | hugo-v6 | Ariel_: not for me in .de beside that i want a 50x |
| 23:54.25 | De_Mon | is that.. intended? |
| 23:54.33 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 23:54.50 | hugo-v6 | goo? sounds funny :) |
| 23:55.17 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 23:55.58 | De_Mon | hugo-v6 the pbx is disconnecting before festival has a chance to say the whole string |
| 23:56.32 | hugo-v6 | De_Mon: i thought something like that. but i cant help you. sorry. |
| 23:56.48 | joe | if there is a long delay before you start hearing a ring what could be the problem (handoff from a fujitxu pbx via a t1)? |
| 23:57.13 | X-Files | De_Mon jaike Qwell: check it please : http://pastebin.ca/37713 <<-- sip debug |
