| 00:00.45 | RoyK | kippi: that is the other side blocking |
| 00:01.01 | RoyK | due to a loop |
| 00:02.10 | kippi | is there somewhere that will help me set this up? so i stop asking stupid questions |
| 00:03.29 | cthompson | Does anyone know if you can "portablize" a number in the US that you don't own? |
| 00:03.41 | cthompson | it's currently not in service at the local telco |
| 00:03.45 | glm2k | kippi: don't worry about stupid questions. just do a bit of research. |
| 00:04.49 | kippi | is it giving me that error because its on the same box? at the mo I don't have a spare box to test with and was hoping to be able to prove the idea on one box |
| 00:05.39 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-49-128.cybersurf.com) |
| 00:06.37 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 00:06.59 | generalhan | whats up everyone ... i need some advice !! |
| 00:07.20 | cthompson | generalhan, that shirt doesn't go with those pants |
| 00:07.23 | _Vile | cthompson, you'd have to sign up with the ILEC/CLEC/Whatever that owns the number block, probably ask for a "vanity" number, give them the number, make it live |
| 00:07.25 | _Vile | and then port it |
| 00:07.29 | generalhan | my boss wants a real nice wireless conference room phone that we can use with * ... anyone have any good expierences ? |
| 00:07.37 | _Vile | to say, vonage or someone |
| 00:10.09 | cthompson | Vile, the owner of the block is Cincinnati Bell, but it's on the other side of town. I wonder if they'll route that number over here |
| 00:11.27 | _Vile | You could setup something with them called "Remote Call Forwarding" |
| 00:11.53 | _Vile | or maybe a "Market Expansion Line", depending on how far away it is etc |
| 00:12.13 | cthompson | by car, about 30mi |
| 00:12.33 | _Vile | either way, ask for Remote Call Forwarding first |
| 00:12.36 | cthompson | unfortunately, my current home service is with Time Warner, we JUST switched |
| 00:12.55 | _Vile | but if it's not in service |
| 00:12.59 | _Vile | and you don't own the number |
| 00:13.06 | _Vile | you'll have to pay for vanity |
| 00:13.15 | _Vile | (probably) |
| 00:13.38 | cthompson | it's been out of service since about 1998. It was the number in the house I grew up in |
| 00:13.55 | cthompson | parents moved out in 98 |
| 00:14.19 | cthompson | we have friends that live in that area, worst case I have one of them pick it up and we figure out how to port it from there |
| 00:14.42 | _Vile | I would ask them for remote call forward, probably what you want to do. |
| 00:14.52 | _Vile | unless you want to throw it to vonage or the likes |
| 00:15.02 | _Vile | which you could also do with remote call forwarding |
| 00:15.21 | cthompson | my goal, eventually, is to have it either with broadvoice or teliax |
| 00:15.24 | cthompson | whoever I choose |
| 00:15.40 | cthompson | I don't have a provider yet, because I still can't afford the sipura :) |
| 00:16.50 | generalhan | Anyone in here used a wireless conference phone with asterisk with good results? people keep telling me to go with cisco, but i dont need to spend $1000 on a phone ! |
| 00:18.06 | *** join/#asterisk milkyflava (n=milkyfla@20-156-237-24-mvl.ewc.gci.net) |
| 00:18.16 | milkyflava | Hello |
| 00:18.47 | milkyflava | I have a tdm400p and am wanting to get a PRI line. What would be the card I would need from Digium? |
| 00:19.29 | russellb | TE110P for a single T1 |
| 00:19.38 | russellb | there are also dual span and quad span cards. |
| 00:20.24 | milkyflava | Thanks russellb, I was looking but wasn't sure if they were the ones I would need. |
| 00:20.30 | Katty | hi lads. |
| 00:20.34 | russellb | you're welcome |
| 00:24.43 | diclophis | so.. why would a line hangup after 3 seconds of being connected? |
| 00:24.49 | diclophis | and during a STREAM FILE playback? |
| 00:30.41 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 00:31.31 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 00:31.44 | r_evolution | do ya'll have different books of the bible than i do? are ya'll Gideons? |
| 00:32.04 | russellb | r_evolution: ? |
| 00:32.13 | r_evolution | nothing... being random. |
| 00:32.28 | r_evolution | Bill Hicks |
| 00:32.38 | russellb | ah. |
| 00:32.59 | r_evolution | Go for it... you feel the urge to ban. Ban away... but then I guess i'll be relegated to the mailing list instead of realtime :p |
| 00:33.03 | milkyflava | russellb, do you know of a site where I can see a picture of a PRI line? Stupid question I know but I just want to see what the line looks like. |
| 00:33.11 | russellb | r_evolution: no, I thought you were a bot at first :) |
| 00:33.14 | r_evolution | ah |
| 00:33.14 | r_evolution | no |
| 00:33.25 | r_evolution | milky... do you have PRI line run into your loc? |
| 00:33.34 | russellb | milkyflava: it's just RJ45 ... |
| 00:33.34 | milkyflava | not yet |
| 00:33.38 | r_evolution | since you're asking for pics... i'll assume no... they're not cheap |
| 00:33.40 | milkyflava | excellent |
| 00:33.57 | weinerk | Please help figure out VoiceMail: |
| 00:33.57 | weinerk | Default A@H install. |
| 00:33.58 | weinerk | Setup two extentions 200 and 201 with xten softphones logged in. |
| 00:33.58 | weinerk | Dialing from 201 to 200. |
| 00:33.58 | weinerk | Get this: "The person at extention" and then it hangs up. |
| 00:34.05 | russellb | ~aah |
| 00:34.07 | jbot | well, aah is Asterisk@Home. The Asterisk@Home support forum is here: http://sourceforge.net/forum/forum.php?forum_id=420324 |
| 00:34.14 | russellb | ~amp |
| 00:34.15 | jbot | [amp] NOT supported here! people using it should join #amportal |
| 00:34.20 | r_evolution | agghhh I think i'm going to end up hanging myself in a few weeks... |
| 00:34.37 | r_evolution | we're going to port our entire calling card platform over to * |
| 00:34.38 | r_evolution | ;) |
| 00:34.50 | r_evolution | THAT'LL be fun. |
| 00:35.27 | *** join/#asterisk simprix (n=simprix@nowhere.simprix.net) |
| 00:36.12 | *** join/#asterisk weinerk (n=irc@bzq-88-152-197-222.red.bezeqint.net) |
| 00:36.18 | simprix | I have alot of static on outgoing calls so that it won't make the call. I have hooked a analog phone up to the phones going into the asterisk box and they sound ok. What could this be ? |
| 00:36.57 | weinerk | sorry, got disconnected. |
| 00:37.54 | weinerk | Default A@H install. |
| 00:37.54 | weinerk | Setup two extentions 200 and 201 with xten softphones logged in. |
| 00:37.54 | weinerk | Dialing from 201 to 200. |
| 00:37.54 | weinerk | Get this: "The person at extention" and then it hangs up. |
| 00:37.54 | weinerk | Any advice? |
| 00:38.11 | r_evolution | debug log? |
| 00:38.34 | I-MOD | weinerk: how did you manage to flood out? |
| 00:39.22 | *** join/#asterisk Ariel_ (n=Ariel@dsl-20-177.cofs.net) |
| 00:39.55 | weinerk | I-MOD, I am not sure - I am just trying to figure out this IRC thing |
| 00:39.56 | Ariel_ | hello everyone |
| 00:40.05 | I-MOD | sup Ariel_ |
| 00:40.18 | weinerk | r_evolution, you mean CLI output? |
| 00:40.36 | r_evolution | well yeah that could help... but does not aah have a debug log? I've never used aah... soooo |
| 00:40.48 | weinerk | <PROTECTED> |
| 00:40.48 | weinerk | <PROTECTED> |
| 00:40.48 | weinerk | <PROTECTED> |
| 00:40.48 | weinerk | <PROTECTED> |
| 00:40.48 | weinerk | <PROTECTED> |
| 00:40.49 | weinerk | <PROTECTED> |
| 00:40.51 | weinerk | <PROTECTED> |
| 00:40.53 | weinerk | <PROTECTED> |
| 00:40.55 | weinerk | <PROTECTED> |
| 00:40.57 | weinerk | <PROTECTED> |
| 00:40.59 | weinerk | <PROTECTED> |
| 00:40.59 | Ariel_ | yes r_evolution it has a full log |
| 00:41.00 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 00:41.01 | weinerk | <PROTECTED> |
| 00:41.07 | russellb | ~pb |
| 00:41.08 | jbot | pb is, like, a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 00:41.10 | diclophis | gah |
| 00:41.30 | russellb | and for help with AMP, please join #amportal |
| 00:41.37 | I-MOD | i figured thats what happened last time..... |
| 00:43.00 | *** join/#asterisk jhiver (n=jhiver@office.gossamer-threads.com) |
| 00:43.41 | weinerk | thanks I understand now about pastebin.com |
| 00:43.41 | r_evolution | thanks russell |
| 00:43.55 | weinerk | sorry |
| 00:45.29 | russellb | no problem :) |
| 00:45.53 | russellb | just dont' let it happen again! orrrr ... your computer will catch on fire! |
| 00:48.04 | weinerk | Voicemail hangs up in the middle - CLI log here: http://pastebin.com/602745 |
| 00:48.08 | *** join/#asterisk alexmontoanelli (i=1000@200.193.10.102) |
| 00:49.00 | alexmontoanelli | hello? |
| 00:49.13 | justinu | hi? |
| 00:49.16 | alexmontoanelli | anybody is alive? |
| 00:49.30 | justinu | alive? not exactly |
| 00:49.33 | rpm | no, my heart has stopped. |
| 00:49.38 | alexmontoanelli | ow |
| 00:49.41 | alexmontoanelli | because |
| 00:49.49 | *** join/#asterisk robbyt (n=robbyt@70.90.77.201) |
| 00:50.05 | robbyt | hey guys- any idea where i can find polycom sip firmware? |
| 00:50.09 | robbyt | 1.6.5 actually :D |
| 00:50.42 | alexmontoanelli | hum..polycom... |
| 00:50.50 | alexmontoanelli | i never use this phones |
| 00:50.54 | alexmontoanelli | only cisco |
| 00:51.09 | robbyt | there's lots of stuff about the ciscos that piss me off |
| 00:51.15 | robbyt | if you're not using the cisco callcenter |
| 00:51.17 | *** join/#asterisk sysdebug (n=sysdebug@200.250.222.8) |
| 00:51.31 | alexmontoanelli | ok |
| 00:52.32 | alexmontoanelli | about the last release of *...is secury migrate to new |
| 00:52.33 | alexmontoanelli | ? |
| 01:00.07 | milkyflava | russellb, thanks again. I have found it. The TE110P is the card and I talked with our local phone company just now and it is a go. |
| 01:00.53 | russellb | cool :) |
| 01:00.58 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 01:04.00 | *** join/#asterisk kc5cqm (i=[U2FsdGV@cpe-68-206-126-19.stx.res.rr.com) |
| 01:04.02 | X-Rob | ~centosbug |
| 01:04.07 | jbot | somebody said centosbug was a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 01:04.40 | kc5cqm | question: I have an asterisk box on 2 networks. Can it respond to calls from phones on both? |
| 01:05.26 | kc5cqm | the problem I run into is the localnet=address/netmask line in sip.conf. How do I specify 2 networks in there? |
| 01:05.29 | BrianUT | tell that bot the link for how to ask questions the smart way |
| 01:05.50 | kc5cqm | BrianUT, that was appropriate. |
| 01:06.31 | *** join/#asterisk Flauto (n=zhao@71.194.194.48) |
| 01:07.53 | *** join/#asterisk bweschke (n=bweschke@sjcc28x184.sjccnet.com) |
| 01:07.53 | Flauto | hi all |
| 01:08.04 | *** join/#asterisk dwmw2_gone (n=dwmw2@baythorne.infradead.org) |
| 01:10.13 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 01:11.32 | vira | how can i record calls made from asterisk 1.0.7? |
| 01:12.03 | *** join/#asterisk MacWeenie (n=chatzill@82-35-73-28.cable.ubr02.dals.blueyonder.co.uk) |
| 01:12.46 | *** join/#asterisk criptos (n=criptos@201.145.227.240) |
| 01:12.51 | criptos | Hi! |
| 01:13.31 | MacWeenie | aloha |
| 01:13.31 | criptos | A pap2 linksys adapter can be configured to autodial at hook-off? |
| 01:14.30 | MacWeenie | if i have a SIP trunk, can i setup an asterisk box to do simple call conferencing? any pointers on a tutorial or howto would be great |
| 01:14.32 | MacWeenie | thanks |
| 01:16.51 | *** join/#asterisk angom_w (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 01:20.17 | r_evolution | kc... you should be able to use two networks without that much of an issue... can you not? |
| 01:21.09 | ambriento | kc5cqm, just add another line stating the other net/subnet |
| 01:21.30 | ambriento | that way you'll have: |
| 01:21.34 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 01:21.43 | ambriento | localnet=net1/submask |
| 01:21.49 | ambriento | localhet=net2/localnet |
| 01:22.06 | ambriento | oops, s/localnet/submask/g |
| 01:23.08 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 01:23.09 | kc5cqm | ambriento, thanks ;-) |
| 01:23.55 | robbyt | so guys- any idea where i can find polycom sip firmware? |
| 01:25.31 | criptos | So, pap2 can be put to autodial? |
| 01:25.46 | kc5cqm | r_evolution, I ve got 172.16.1.0/255.255.255.255 for my internal hardphones and internet feeding in via a NAT IP on a 192.168.1.1/255.255.255.255 network, both IAX2 and SIP... once I moved the hardphones the dedicated 172.16.1.0/24 network my SIP connectivity to outside SIP providers seems to have failed. Also, I wanted to add a softphone on one of the 192.168.1.0/24 machines. |
| 01:26.57 | r_evolution | hrm. and I'm assuming you're not having any problems pinging from either interface directly to the outside? |
| 01:27.22 | kc5cqm | the 172 network is totally isolated...just talks to * |
| 01:27.26 | *** part/#asterisk angom_w (n=Administ@red-corp-200.38.16.10.telnor.net) |
| 01:27.26 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@c-71-228-12-47.hsd1.il.comcast.net) |
| 01:27.27 | r_evolution | meaning your interfaces are set up with proper routing... |
| 01:27.50 | kc5cqm | yeah, routing is set up. the * server actually has an IP on both networks |
| 01:27.51 | r_evolution | nono i mean... you can use either interface to ping the outside world |
| 01:27.55 | r_evolution | ok |
| 01:28.22 | kc5cqm | I'm attempting the localnet= twice now... |
| 01:28.23 | kc5cqm | hehe |
| 01:28.35 | r_evolution | go into the command line for whatever distro you're using and see if you can ping the outside world using the specific adapters |
| 01:29.23 | kc5cqm | actually I don't want the 172.16.1.0/24 to be able to route to the outside world at all |
| 01:29.41 | kc5cqm | the 192.168.1.0/24 one can though |
| 01:29.44 | r_evolution | ^ |
| 01:29.52 | *** part/#asterisk AlexCTI (n=alex@64.251.9.44) |
| 01:29.54 | r_evolution | see if you can ping the outside using the 192 adapter |
| 01:29.54 | kc5cqm | I've only got hardphones on the 172, and they only talk to this * box |
| 01:37.02 | *** join/#asterisk Husk (n=tcg@202.55.153.169) |
| 01:39.42 | justinu | i see my r_evolution is giving back to the community ;) |
| 01:39.57 | justinu | my prodigy r_evolution, that is |
| 01:40.04 | r_evolution | si senor... soon i will snatch the pebble from your hand ;) |
| 01:40.16 | justinu | lol |
| 01:40.23 | justinu | you can have it |
| 01:40.30 | r_evolution | you're no fun |
| 01:42.28 | orlock | Anybody here used sail/selintra? |
| 01:44.59 | *** join/#asterisk jhnjwng (n=wj1918@pool-70-21-174-24.nwrk.east.verizon.net) |
| 01:53.09 | ambriento | kc5cqm, you have at least 2 NICs on your *box, right? |
| 01:53.27 | kc5cqm | yes |
| 01:55.31 | *** join/#asterisk forao (n=fjaosdfj@ool-4354d6b4.dyn.optonline.net) |
| 01:55.50 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 01:56.01 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 01:56.04 | *** join/#asterisk the_magic_bean (n=the_magi@66-73-141-82.ded.ameritech.net) |
| 01:56.21 | criptos | which mpg123 everybody is using? |
| 01:56.33 | MikeJ[Laptop] | .59r |
| 01:56.42 | criptos | I just went to mpg123.de and says something about security issues... |
| 01:57.57 | gaupe | use the native mp3 support from the addons package instead |
| 01:58.30 | criptos | I have this: Asterisk CVS-v1-2-11 |
| 01:58.34 | criptos | and cant update :( |
| 01:58.45 | criptos | must get moh to work... |
| 01:59.05 | gaupe | you don't need to, you can compile the addons package against it |
| 01:59.39 | gaupe | or you could convert your mp3-files to a format already supported natively and play them |
| 01:59.54 | criptos | Got it.. |
| 01:59.57 | criptos | build it... |
| 02:00.08 | criptos | got format_mp3.so now? |
| 02:00.44 | criptos | humm. show modules like format show format_mp3... |
| 02:01.50 | *** join/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 02:01.54 | Husk | will asterisk listen to DTMF commands and goto different dialplans while its at Wait() ? |
| 02:02.50 | gaupe | criptos: you need to load it too |
| 02:03.55 | *** join/#asterisk Altair256 (n=icechat5@mail.clccorp.com) |
| 02:03.57 | r_evolution | Husk use WaitExten |
| 02:04.31 | Altair256 | Hello everyone |
| 02:04.53 | *** part/#asterisk kc5cqm (i=[U2FsdGV@cpe-68-206-126-19.stx.res.rr.com) |
| 02:06.04 | forao | im starting with postgresql and i might be interested in a gui, any recommendations? |
| 02:06.36 | rpm | psql.. :P |
| 02:07.36 | forao | hehe |
| 02:09.29 | glm2k | forao: phpMyPgAdmin perhaps? |
| 02:09.43 | glm2k | forao: er, sorry phpPgAdmin |
| 02:10.51 | forao | does anyone know navicat? |
| 02:12.56 | *** join/#asterisk d-tech (n=dtc@h-72-245-233-107.sfldmidn.covad.net) |
| 02:15.02 | *** join/#asterisk heath__ (n=heath__@12-215-32-56.client.mchsi.com) |
| 02:17.56 | *** join/#asterisk the_magic_bean (n=the_magi@CrawfordELP-64-72-134-185.onecall.net) |
| 02:19.26 | *** join/#asterisk fugitivo (n=ajf@201.255.178.20) |
| 02:19.48 | *** join/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 02:31.36 | criptos | humm.. |
| 02:31.54 | criptos | when playing moh, CPU goes to a load of .80 |
| 02:32.08 | criptos | to something like 3,1,Musingonhold() |
| 02:32.32 | criptos | but when musingonhold is used at a dial, the load goes to .25 and the sound is too crappy... |
| 02:32.34 | criptos | any ideas? |
| 02:33.28 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 02:43.29 | *** part/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 02:48.03 | *** join/#asterisk jmacz (n=jmacz@201.244.241.189) |
| 02:51.19 | robbyt | so does anyone here know where i can find polycom firmware? :) |
| 02:52.18 | SwK | robbyt: polycoms website |
| 02:52.43 | robbyt | they only let you download the firmware if you're a registered reseller |
| 02:52.46 | robbyt | which i am not |
| 02:52.59 | SwK | not true |
| 02:53.05 | robbyt | any other ideas? |
| 02:53.14 | robbyt | well, they let you download the old version |
| 02:53.20 | robbyt | i want to get my hands on 1.6.5 |
| 02:53.34 | robbyt | i accidently upgraded a phone to 1.6.2 without actually rtfm'ing |
| 02:53.37 | SwK | paysomeone off |
| 02:54.05 | SwK | hah |
| 02:54.24 | SwK | when did 1.6.5 get released? |
| 02:54.49 | robbyt | not sure- it's listed on their site though |
| 02:55.06 | robbyt | i'm having echo suppressor issues with the speakerphone in 1.6.2 |
| 02:55.10 | robbyt | that weren't there in 1.5.x |
| 02:55.11 | *** join/#asterisk ZX81 (n=ZX81@222-153-114-13.jetstream.xtra.co.nz) |
| 02:55.36 | ZX81 | anyone know how to fix show agents |
| 02:55.39 | ZX81 | so I don't get |
| 02:55.39 | ZX81 | 1002 (Seat 2) available at '202' (musiconhold is 'default') |
| 02:55.47 | ZX81 | when the agent is in wrapuptime? |
| 02:56.38 | fugitivo | ~seen coppice |
| 02:56.50 | jbot | coppice <n=chatzill@211.155.168.153> was last seen on IRC in channel #asterisk, 16h 55m 48s ago, saying: '"Nun"'. |
| 02:58.00 | ZX81 | hmmmm |
| 02:58.28 | *** join/#asterisk moy (n=moy@201.135.98.235) |
| 02:58.40 | moy | good night everyone |
| 02:58.45 | *** join/#asterisk nurfe (n=rgff@h24-207-10-162.dlt.dccnet.com) |
| 03:00.07 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 03:00.22 | moy | im having some problems with UNICALL and E1 with signaling MFC/R2 in Mexico... it seems that sometimes my telco does not detect our tones, any ideas? this is random, some calls goes just fine, but others get stuck in the middle until are timeout |
| 03:00.52 | moy | im not using asterisk for making and receiving the calls so far, just using the testcall program that comes with Unicall |
| 03:00.53 | *** part/#asterisk yxa (n=diablo@58.185.90.101) |
| 03:02.29 | moy | any other channel where i can ask? |
| 03:04.04 | willt | hey guys im seeing this: ast_filehelper: Unable to open file on /var/lib/asterisk/sounds/NN/test/g2.wav |
| 03:04.16 | willt | but the file is there any ideas? |
| 03:05.57 | *** join/#asterisk SibRphrek (n=SibRphre@user-12lccke.cable.mindspring.com) |
| 03:06.09 | *** join/#asterisk forao (n=fjaosdfj@ool-4354d6b4.dyn.optonline.net) |
| 03:10.42 | xmark | exit |
| 03:10.44 | willt | nm i see whats up |
| 03:25.29 | *** join/#asterisk nayyares (n=Nayyar@mbl-65-151-218.dsl.net.pk) |
| 03:25.50 | nayyares | hi guys |
| 03:26.14 | *** join/#asterisk Psyiode (n=psyiode@205.241.238.186) |
| 03:30.40 | *** join/#asterisk viLeR (i=1000@66.128.47.232) |
| 03:31.20 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 03:33.47 | mocker | Is asterisk-addons-1.2.2 compatible w/ asterisk 1.2.5? |
| 03:33.55 | mocker | Specifically the mysql addon. |
| 03:34.59 | russellb | yes |
| 03:35.05 | russellb | just use the latest version of each |
| 03:38.03 | *** join/#asterisk newmember[laptop (n=newmembe@S010600036d1139bb.cg.shawcable.net) |
| 03:39.56 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 03:47.59 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
| 03:49.40 | hollymolly | Can someone pm me if they have project gizmo working with their asterisk system.. thank you |
| 03:51.41 | *** join/#asterisk bmg505 (n=leon@dsl-146-47-223.telkomadsl.co.za) |
| 03:52.00 | littleball | hello, i am using background to play a voice prompt and at the same time to collect the users' DTMF input. Is it possible to end the user input if the user type # key? |
| 03:55.55 | orlock | hey |
| 03:56.05 | orlock | which codecs are bgest for asterisk/grandstreams? |
| 03:56.36 | orlock | i cant seem to see ulaw/alaw/g711 in the grandstream conf |
| 03:56.42 | Octothorpe | littleball: what exactly to you mean "collecting DTMF" |
| 03:57.37 | tsume | well he needs to read the manual and wiki |
| 03:58.06 | tsume | he means reading input like a wake up call time or etcc |
| 03:58.14 | Octothorpe | ok |
| 03:58.22 | Octothorpe | ~thebook |
| 03:58.25 | jbot | [thebook] Asterisk: The Future of Telephony, released under a Creative Commons license and available at http://www.asteriskdocs.org << Read the book online! |
| 03:58.40 | Octothorpe | ~wiki |
| 04:00.35 | Octothorpe | last I heard the wiki is down |
| 04:04.05 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 04:07.43 | *** join/#asterisk r0d3nt|m (i=anonymou@tinfoilhat.net) |
| 04:08.12 | Octothorpe | or not |
| 04:08.19 | Octothorpe | ~asteriskwiki |
| 04:08.21 | jbot | well, asteriskwiki is at http://www.voip-info.org/wiki-Asterisk" |
| 04:08.57 | hollymolly | Help - I'm not able to rec'v incoming calls from my project gizmo number, but can make outgoing, and appear to be registerd. Can anyone help me wade through the sip debug info to see where project gizmo gives up and decides to answer the call for me? |
| 04:11.25 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 04:12.35 | asterboy | ~fuckingecho |
| 04:12.38 | asterboy | ~echo |
| 04:12.39 | jbot | hmm... echo is Displays the given arguments on the screen. Syntax: echo (arg1) (arg2) ..(argN). Where arg1 through argN are the arguments to echo. Example: echo "Hello World" displays the string "Hello World". "Why echo occurs" at http://lists.digium.com/pipermail/asterisk-users/2005-February/088794.html |
| 04:13.13 | asterboy | The requested URL /pipermail/asterisk-users/2005-February/088794.html was not found on this server. |
| 04:13.41 | asterboy | I like this one: http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current/docs-html/x1695.html |
| 04:13.54 | willt | try using the google cache |
| 04:14.04 | Altair256 | two quick questions... anyone know a good CallerID (SIP) program for Win32... and TAPI integration for multiple users on the same PC (ie, terminal services) |
| 04:15.15 | asterboy | wilt, good suggestion. |
| 04:16.02 | willt | I can't get any of their maillist archives to come up |
| 04:19.00 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 04:19.45 | *** part/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 04:20.21 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 04:20.52 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 04:21.31 | littleball | hello, my system call a phone number, when the user answer the call, my system prompts the user to key in a number followed by # key. To implement this, i am using background() to collect the DTMFs. |
| 04:21.36 | asterboy | Anyone have some good txgain or rxgain settings to suggest? |
| 04:21.43 | asterboy | Can't find ztmonitor. |
| 04:21.53 | littleball | but when the users finish key #, the background () not return immediately |
| 04:21.54 | asterboy | Must not have compiled |
| 04:22.05 | Altair256 | <PROTECTED> |
| 04:22.06 | littleball | and need to wait for a few seconds. How to solve this problem? |
| 04:22.13 | Altair256 | then ./ztmonitor |
| 04:23.29 | *** join/#asterisk JunK-Y (n=junky@67.70.34.128) |
| 04:23.51 | asterboy | ah yes, thats why I can't find ztmonitor...have to mount that drive with /usr/src on it. |
| 04:24.13 | Altair256 | lol asterboy |
| 04:24.28 | asterboy | doh! |
| 04:24.45 | Altair256 | how do I make it so a SIP account can receive more than one call? similar to call waiting, except it would ring another line button |
| 04:25.03 | Altair256 | using Linksys SPA-941 and X-Lite softphone... I have a feeling it's a .conf issue though |
| 04:25.58 | asterboy | I have a serious contract for telephones to prepare for tomorow. |
| 04:26.05 | willt | I think it does that for me by default |
| 04:26.21 | Altair256 | hrm willt, happen to know where the setting is? |
| 04:28.14 | willt | xlite is doing it for me I believe. if I pickup line one and dial my extension it rings on line 2 |
| 04:28.21 | *** join/#asterisk Utah_Dave (n=boucha@c-67-166-71-219.hsd1.ut.comcast.net) |
| 04:28.42 | Altair256 | mine says "The person at extension ### is on the phone" |
| 04:29.06 | willt | are you limiting the number of channels in you sip.conf? |
| 04:29.14 | Altair256 | let me check |
| 04:29.55 | Altair256 | does seem so |
| 04:31.01 | asterboy | wow, zmonitor show my TX off the side of the screen. |
| 04:31.17 | asterboy | Now how does that txgain=# work? |
| 04:31.23 | asterboy | 1.0 is current setting |
| 04:31.35 | willt | Altair256: how many channels are you allowing? |
| 04:31.42 | asterboy | so if I set it to 12.0 that lowers it? |
| 04:32.19 | Altair256 | I can't find anything in sip.conf or any of it includes that even mentions channels |
| 04:32.32 | Altair256 | lookig at my macro-dial now, since all dials go through it |
| 04:33.00 | Altair256 | ah, found it |
| 04:33.01 | Altair256 | ; Ring an extension, if the extension is busy or there is no answer send it |
| 04:33.01 | Altair256 | ; to voicemail |
| 04:33.07 | asterboy | Got it: http://www.voip-info.org/wiki-Asterisk+x100p+echotraining |
| 04:33.17 | Octothorpe | m l;, |
| 04:33.18 | asterboy | set it to negative to lower |
| 04:33.19 | Altair256 | at least now I know where I need to work |
| 04:33.27 | willt | Altair256: are you using A@H ? |
| 04:33.36 | Altair256 | >.> |
| 04:33.38 | Altair256 | maybe |
| 04:33.50 | willt | I ask because.... |
| 04:33.51 | Octothorpe | ~aah |
| 04:33.59 | Altair256 | bah Octothorpe |
| 04:34.02 | Altair256 | lol... it's : |
| 04:34.03 | Altair256 | ~amp |
| 04:34.05 | jbot | extra, extra, read all about it, amp is NOT supported here! people using it should join #amportal |
| 04:34.15 | Altair256 | #amportal has like 5 people in it |
| 04:34.20 | Altair256 | this is where the support is... lol |
| 04:34.22 | willt | I setup a server at work the other day with that. and it does the same thing as you are seeing |
| 04:34.35 | Altair256 | I'm willing to learn and rewrite the script, so no elistestism |
| 04:35.09 | Altair256 | willt, to fix it go to extensions.conf in the [macro-dial] section |
| 04:35.36 | Altair256 | ; Ring an extension, if the extension is busy or there is no answer send it |
| 04:35.50 | Altair256 | ... voice mail |
| 04:35.59 | Altair256 | so AAH appears to do it automatically |
| 04:36.07 | Altair256 | sure gotta fix that >.< |
| 04:37.12 | willt | shouldn't call-limit=1 limit the user to one call or channel? |
| 04:37.43 | Altair256 | where are you putting the call-limit=1? |
| 04:38.15 | Octothorpe | lol, it's late, i'm tired |
| 04:38.38 | *** join/#asterisk icyfire0573 (n=icyfire@u1016342.ul.warwick.net) |
| 04:41.26 | icyfire0573 | I'm looking into building an asterisk box in my house. I want to get two lines into the house, and control multiple analog phones in the house. To top it all off I want to do this on a low power (Read VIA EPIA) motherboard, that come with only 1 PCI slot. Is it possible to do this, or do I need to get a motherboard with more PCI |
| 04:42.20 | Altair256 | you can get a TDM400 with 2 FX2 modules |
| 04:43.30 | Altair256 | willt, fyi.. when you create an extension, "call waiting" is disabled in AAH2.7 |
| 04:43.40 | Altair256 | to turn it on, pick up the extension and dial *70 |
| 04:44.15 | icyfire0573 | FX2? So two FXO and two FXS? |
| 04:44.23 | willt | Altair256: thanks for the tip |
| 04:44.37 | Altair256 | do you want to provide 2 analog jacks to plug regular phones into? |
| 04:44.53 | Altair256 | you could also consider getting an extra SPA device |
| 04:44.57 | icyfire0573 | At least two. But I could separate it into two areas |
| 04:45.00 | Altair256 | 1 sec, I'll get you a link |
| 04:45.41 | Altair256 | http://www.linksys.com/servlet/Satellite?c=L_Product_C2&childpagename=US%2FLayout&cid=1139414817158&pagename=Linksys%2FCommon%2FVisitorWrapper |
| 04:45.49 | Altair256 | that has 1 FXO and 1 FXS port |
| 04:46.11 | Altair256 | about the size of a box of cigaretts |
| 04:49.05 | icyfire0573 | Alright, not to be dense. But the computer getting signal from the Telephone company, thats the FXO card. And inside the area, pointing from the computer to the phones, those are FXS right? |
| 04:49.27 | Altair256 | the line coming from the company is the FXO (office) |
| 04:49.32 | Altair256 | you plug it into an FXS port |
| 04:50.17 | icyfire0573 | So in the computer, the telephone company's lines plug into my FXS cards? |
| 04:50.23 | Altair256 | Your FXO port (where you are the office) is where you plug in an FXS device (such as an analog telephone, fax machine, etc)... this is the "station" |
| 04:50.36 | *** join/#asterisk Faithful (n=Faithful@202-6-145-116.ip.adam.com.au) |
| 04:50.45 | Altair256 | correct, the FXS port is the "station" - the end point |
| 04:51.02 | Altair256 | it's hella confusing |
| 04:51.12 | icyfire0573 | haha, yea I know I got em backwards :-D |
| 04:51.23 | Altair256 | just remember that you plug an FXO line plugs into an FXS port, and vice versa |
| 04:51.32 | wasim | fxo is where you receive tone/battery, fxs is where you power the line, provide tone |
| 04:52.19 | icyfire0573 | Alright. |
| 04:52.19 | Altair256 | ~fxo |
| 04:52.21 | jbot | foreign exchange office - type of port you need to connect a POTS (Plain Old Telephone Service) line from your telco to a pbx http://www.digium.com/index.php?menu=fxsvfxo |
| 04:52.24 | Altair256 | ~fxs |
| 04:52.25 | jbot | well, fxs is foreign exchange system - or the type of port you need to connect a analog device (phone, fax machine) to a pbx |
| 04:52.59 | Altair256 | >.> |
| 04:53.09 | icyfire0573 | I gotta look more into semi generic cards. I'm not adverse to spending some money, but I dont want to spend 300 for 2 lines in and 2 inside area phones. |
| 04:53.48 | Altair256 | icyfire0573, double check me... jbot is disagreeing with me |
| 04:53.56 | icyfire0573 | Also, if I got IP Phone, I wouldn't need FXO cards right? |
| 04:54.08 | *** join/#asterisk Eggplant (i=No@dsl-216-155-210-218.cascadeaccess.com) |
| 04:54.22 | Altair256 | hold on, I do have it backwards |
| 04:54.39 | Altair256 | FXO card = receive signal |
| 04:54.46 | Altair256 | FXS card = provide signal |
| 04:55.03 | icyfire0573 | haha, dont mess with me :-) my grasp is tenuous. lol. |
| 04:55.20 | Altair256 | so I suggest buying 1 FXS port.. and getting a multi-handset Uniden phone |
| 04:55.46 | Altair256 | all the phones talk back to the one home base (so the system is only plugged in once), but you can put the charger for additional bases all over the house |
| 04:55.55 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 04:56.13 | *** part/#asterisk brockj49464 (n=brockj49@63.87.56.236) |
| 04:56.42 | icyfire0573 | I actually have that right now, 3 handsets |
| 04:56.43 | Altair256 | icyfire0573, read http://www.voip-info.org/wiki-FXO ... this should clear it up for you |
| 04:56.56 | icyfire0573 | I was kinda hopin to be able to set extensions for each room of the house |
| 04:57.12 | Altair256 | then plan on paying at least $50 per extension for a DTA |
| 04:57.20 | icyfire0573 | + being able to call out when someone else is on the phone/get a phone call (w/o call waiting) (you need 2 phone lines i know) |
| 04:57.31 | icyfire0573 | DTA? |
| 04:57.32 | wasim | you can buy an atcom 4 port fxs ata for like $100 |
| 04:57.34 | Altair256 | and then running an ethernet cable to the DTA wherever you want that extension |
| 04:57.54 | Altair256 | yes wasim, but they'll all be in the same room |
| 04:58.09 | wasim | Altair256: no, you plug wireless, or drag normal telephone wire |
| 04:58.24 | Altair256 | wireless will add an extra 40$ per extension then |
| 04:58.25 | wasim | Altair256: like any normal phone extension |
| 04:58.49 | Altair256 | but if you can get a 4 port FXS for $100 then your initial per extension is only $25 |
| 04:59.27 | Altair256 | dragging normal telephone wire is even more trouble, imo |
| 04:59.53 | Altair256 | with wireless, you mean wireless analog phones? so all the bases should sit in that one room? o.O |
| 05:00.37 | icyfire0573 | Umm, where would I find this 4port FXS card? Because, right now all I'm seeing at Froogle is $1200 Cisco 4ports |
| 05:00.53 | *** join/#asterisk dotslashroot (n=kuntz@mont-cas2-73.dial.mhtc.net) |
| 05:01.09 | wasim | icyfire0573: for FXO you should use PCI cards, digium or sangoma, for FXS the ATA are a much cheaper and extendable option |
| 05:01.09 | Altair256 | voip-supply.com |
| 05:01.45 | Altair256 | ~dta |
| 05:01.46 | Altair256 | ~ata |
| 05:01.48 | jbot | it has been said that ata is Analog Telephone Adapter which is used to put a normal analog phone onto ethernet, see http://www.voip-info.org/tiki-index.php?page=Analog%20Telephone%20Adapters for more info |
| 05:02.32 | icyfire0573 | http://www.digitnetworks.com/store/product_info.php?cPath=22&products_id=45 (For the 2 FXO cards) |
| 05:02.54 | Altair256 | for home, get the OEM intel clones |
| 05:03.01 | Altair256 | the quality isn't that bad |
| 05:03.26 | Octothorpe | I like the ones from x100p.com (I use one myself) |
| 05:03.29 | Altair256 | he's limited to 1 PCI slot though |
| 05:03.30 | icyfire0573 | And I didn't see anything lower than $300 for the FXS at voip-supply |
| 05:03.51 | icyfire0573 | I'm not limited yet, I'd prefer to do this all on the one board |
| 05:03.54 | Octothorpe | sorry, that was fxo, not fxs |
| 05:04.02 | icyfire0573 | but if it isn't feasible Ill buy a different motherboard. |
| 05:04.11 | icyfire0573 | The VIA is just what I'd LIKE to use |
| 05:04.57 | Octothorpe | if you need all 4 ports in one card, 300 is probably the least expensive you get |
| 05:05.38 | Altair256 | wasim, where is this atcom 4 port fxs ata you're talking about? |
| 05:05.44 | icyfire0573 | Whew, alright anything better on multiple ports? FXS cards seem to be the more expensive ones. |
| 05:06.00 | Altair256 | you can get the X100P clones for like 10-15$ each |
| 05:06.06 | wasim | Altair256: on the atcom website? |
| 05:06.07 | Altair256 | just make sure they have their own IRQ |
| 05:06.18 | Altair256 | cat /proc/interrupts |
| 05:07.03 | Altair256 | I'm either at the wrong site, or... something else |
| 05:07.25 | wasim | http://atcom.cn/ |
| 05:07.27 | icyfire0573 | http://www.iareaphone.com/ShoppingCart/catalogue_enterfromstatic.asp?ProductSet=10279 I think this is what he is referring to. But I think its a 4port router with 1 or 2 phone in there |
| 05:07.33 | Altair256 | ty |
| 05:08.39 | Altair256 | this is what he's talking about http://atcom.cn/En_products_AG468.html |
| 05:09.10 | Altair256 | retails for 120$... looking for a $100 |
| 05:10.21 | Altair256 | if I could find that AG-468 for $100, I think I'd buy one myself |
| 05:10.26 | Altair256 | that's a heck of a deal |
| 05:11.32 | wasim | depends on the vol, we pick up 20 packs for $80 from the factory |
| 05:11.47 | Altair256 | http://www.ipchitchat.com/products/AG-468.htm how much is that in USD? lol |
| 05:12.00 | icyfire0573 | Alright, So I can get this box, which runs on Ethernet, so I can hook it up anywhere on the network. And if im willing to run just One phone line in, or pay another $200 for two phone lines in. I can do this all on 1 pci slot? right? |
| 05:12.27 | Altair256 | 1 phone line, 1 OEM X100P ($15) |
| 05:12.42 | wasim | icyfire0573: you might want to try the audiocodes fxo ata if pci slots are a rarity |
| 05:12.45 | icyfire0573 | 95.27 British pounds = 165.045748 U.S. dollars |
| 05:13.12 | Altair256 | 4 analogue extensions, AG468 (for around 150$ maybe, if you look hard enough) |
| 05:14.48 | icyfire0573 | Naa, this works for me, A $200 solution with 4 intenal extensions (Plus any computers that want to connect) and 1 exterior phone number, more if I want to get more PCI card slots (and then cards.) |
| 05:14.51 | icyfire0573 | This is great! |
| 05:15.55 | Altair256 | you can also play with some softphones first as well |
| 05:16.08 | wasim | and tin cans and string ... |
| 05:16.15 | Altair256 | X-Lite, IDEFISK http://www.asteriskguru.com/tools/idefisk_beta.php, etc |
| 05:16.19 | *** join/#asterisk Z-Knight (n=Z-Knight@cpe-67-10-28-100.houston.res.rr.com) |
| 05:16.19 | icyfire0573 | haha, yea, just make sure the string is tight! :-) |
| 05:17.17 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 05:17.20 | Z-Knight | stupid question follows: For the asterisk 1.2.5, when entering the console via: asterisk -cvvvv how do you quit out? I tried QUIT and nothing... |
| 05:17.29 | [TK]D-Fender | "stop now" |
| 05:17.39 | icyfire0573 | Thats a great place to start, to check if I'm good with asterisk to being with, w/ only a $15 layout if I don't like the idea in general. |
| 05:17.52 | [TK]D-Fender | Next!!!!!!!! |
| 05:17.53 | Z-Knight | that closes * completely |
| 05:17.54 | Altair256 | exactly icyfire0573 |
| 05:17.57 | [TK]D-Fender | (tm) BKW |
| 05:18.07 | wasim | ^Z bg |
| 05:18.08 | Z-Knight | I'm talking about leaving the console and leave * running |
| 05:18.28 | wasim | don't use -c to start it, use -r to reattach to it |
| 05:18.31 | [TK]D-Fender | Z-Knight : If you wanted to start * and then be able to quit you should start it as a daemon as "safe_asterisk &" |
| 05:18.32 | Z-Knight | k |
| 05:18.34 | Z-Knight | thanks |
| 05:18.47 | [TK]D-Fender | And then connect to it with "asterisk -r" |
| 05:18.53 | Altair256 | bah, I have to go home |
| 05:19.00 | Altair256 | been at work for 20 hours today -.- |
| 05:19.20 | Z-Knight | ok...thank you...I used the 'c' flag thinking it provided console which I could quit |
| 05:19.25 | Altair256 | only got 3 hours of sleep v.v;; |
| 05:19.26 | icyfire0573 | harsh, hope you get overtime Altair256 |
| 05:19.38 | Altair256 | no, but I take comp time |
| 05:20.26 | *** part/#asterisk alephcom (n=darren@host75.net14.mcsnet.ca) |
| 05:21.55 | *** join/#asterisk bkw__ (n=bkw_@adsl-70-142-62-199.dsl.tul2ok.sbcglobal.net) |
| 05:22.20 | icyfire0573 | Thanks for all the help guys, Its been great. I'm gonna watch some Anime and go to bed. |
| 05:24.37 | Altair256 | later guys |
| 05:34.41 | *** part/#asterisk heath__ (n=heath__@12-215-32-56.client.mchsi.com) |
| 05:35.32 | MikeJ[Laptop] | ~centos |
| 05:35.33 | jbot | well, centos is better than Fedora Core except for that silly bug, see ~centosbug for details |
| 05:35.33 | orlock | goddamn |
| 05:35.40 | MikeJ[Laptop] | ~centosbug |
| 05:35.42 | jbot | well, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 05:35.49 | orlock | i'm getting auth failures when trying to register with * |
| 05:35.51 | *** join/#asterisk vuud (n=vuud@wsip-68-15-62-138.ri.ri.cox.net) |
| 05:35.59 | orlock | username and secret for each extension is right |
| 05:36.51 | vuud | good morning all. |
| 05:38.14 | vuud | I am having a problem connecting my pbx to gizmo... I have outbound calls working, and inbound calls half work... when they call in and I have an undefined context in sip.conf, they hear the voicemail lady and can leave a voicemail... |
| 05:38.45 | vuud | If I define the context in extensions, they hear nothing. On the CLI it says its doing everything it should be (playing sounds) but they hear nothing. |
| 05:39.45 | vuud | There is NAT, but why it would work without a valid context, but will not with one stumps me. Any help would be greatly appreciated |
| 05:42.43 | justinu | canreinvite=no |
| 05:42.54 | justinu | set that, vuud |
| 05:44.43 | *** join/#asterisk lylsir (n=lylsir@222.188.139.82) |
| 05:45.27 | vuud | justinu: have that set in my from-gizmo part of the sip.conf file |
| 05:45.53 | justinu | the fact that it works on a default context tells me that the defaults are correct |
| 05:46.13 | rpm | [2006-03-14 22:40:28] WARNING[13032]: pbx.c:785 pbx_find_extension: Maximum PBX stack exceeded |
| 05:46.15 | CoffeeIV | any one here do faxing through a IAX2 provider ? which one ? |
| 05:46.16 | rpm | hax0red. |
| 05:46.26 | SwK | hah nice |
| 05:46.44 | SwK | coffeeiv: faxing via VoIP in the wild internet sucks |
| 05:47.10 | *** join/#asterisk PBXtech (i=PBXtech@178.sub-70-213-226.myvzw.com) |
| 05:47.13 | SwK | unless you just REALLY REALLY REALLY have to use a hard fax machine, get a web -> fax account somewhere |
| 05:47.16 | vuud | justinu: so its something in my sip.conf or extensions.conf that I am overridding |
| 05:47.36 | PBXtech | do the iaxy not generate ring? |
| 05:47.54 | rpm | exten => ${FWDNUMBER},a,VoiceMailMain(100) |
| 05:47.54 | rpm | exten => ${FWDNUMBER},o,Goto(${FWDNUMBER},1) |
| 05:47.59 | rpm | why does that stuff break? |
| 05:48.56 | SwK | I didnt know a and o were valid priorities |
| 05:49.15 | rpm | isn't that for defining the oper and * key? |
| 05:49.31 | SwK | no |
| 05:49.37 | rpm | orly. |
| 05:49.41 | SwK | not the way you used it there |
| 05:49.45 | SwK | yeah really |
| 05:50.05 | rpm | how do i make it unsuck? |
| 05:50.07 | SwK | exten => EXTEN/CID,PRIORITY,APPLICATION(ARGVS) |
| 05:50.24 | rpm | so a and o are the extensions? |
| 05:50.34 | SwK | look at the stdexten macro in the sample dial plan |
| 05:50.41 | rpm | okee |
| 05:50.45 | CoffeeIV | SwK: I know it will suck, I'm just hoping it will work 50% of the time or so |
| 05:51.20 | SwK | coffeeiv: set your fax for 9600 as the fastest and pretty much any of them with ok bandwidth will work sometimes |
| 05:53.14 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 05:56.45 | *** join/#asterisk frk2 (n=frk2@202.5.145.13) |
| 05:56.51 | *** join/#asterisk Rui (n=rmps@85.138.72.66) |
| 05:57.12 | frk2 | jbalcomb you awake? :) |
| 05:57.19 | frk2 | i did the new firmware |
| 05:57.23 | frk2 | seems much better |
| 05:57.46 | frk2 | however the gxp-2000 still has the one-way voice issue... this ONLY happens for calls incoming from the PRI though.. not for internal calls |
| 05:57.47 | justinu | vuud: that would be my inclination |
| 05:57.56 | vuud | Wait a sec... my default context is "default" which should not roll to voicemail... it should redo the demo menu... wtf?!?! (and I have reloaded and restarted many times) |
| 05:58.12 | frk2 | so basically for some calls my gxp-2000 cannot SEND voice to calls incoming from the PRI |
| 05:58.45 | frk2 | can hear them just fine |
| 05:59.04 | justinu | vuud: pare your dialplan down to a minimum |
| 05:59.12 | justinu | build slowly |
| 05:59.18 | justinu | you'll get it straightened out |
| 05:59.20 | justinu | same thing for sip.conf |
| 06:00.12 | vuud | justinu: I had it down to just forward to my x-lite, which I could answer, but no sound either way. I will see what I can take out |
| 06:00.25 | frk2 | damn- my saviour jbalcomb aint here |
| 06:00.37 | justinu | make sure your xlite isn't doing STUN or anything |
| 06:00.42 | Rui | Hello. I've 10 IP phones (Grandstream GXP-2000) and one of them appends "90" before the number I dial (or so it seems). I've already reset that phone and the configuration is the same on all other 9. Is that "90" any special number? What it could be? |
| 06:00.43 | justinu | that could be part of the problem |
| 06:00.48 | justinu | xlite tries to be cute |
| 06:01.19 | frk2 | could be a dialplan setting |
| 06:01.20 | justinu | rui: factory reset? |
| 06:01.30 | frk2 | rui- do you experience any one-way voice issues? |
| 06:01.33 | vuud | justinu: I can check but it does not address the gizmo -> astricks problem. |
| 06:01.40 | justinu | heh |
| 06:01.41 | Rui | justinu: Yes. That MAC address thingy |
| 06:01.52 | justinu | rui: then it's in the asterisk config |
| 06:02.30 | Rui | justinu: Weird. That extension is configured just like any other... |
| 06:02.42 | justinu | sip debug to figure out |
| 06:02.52 | justinu | see exactly what to: the phone invites |
| 06:03.30 | *** join/#asterisk nain (n=nain@202.59.73.36) |
| 06:04.06 | Rui | frk2: No... Everything else works fine. |
| 06:07.00 | frk2 | what firmware you have? |
| 06:07.11 | Z-Knight | with Asterisk 1.2.5, do you still need to compile with PROC=i586 for VIA motherboards? |
| 06:07.20 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:09.03 | Rui | frk2: Program 1.0.1.9, Bootloader 1.0.1.2 |
| 06:10.48 | frk2 | hmmm |
| 06:11.08 | frk2 | i have one way voice issues some times from calls coming in through the PRI |
| 06:11.31 | Z-Knight | Does anyone know with Asterisk 1.2.5, do you still need to compile with PROC=i586 for VIA motherboards? |
| 06:12.57 | frk2 | do you use it with u-law or a-law? |
| 06:15.08 | Rui | frk2: u-law |
| 06:19.12 | *** join/#asterisk nain (n=nain@202.59.73.36) |
| 06:20.31 | *** join/#asterisk sternn (n=sternn@user-0c938ku.cable.mindspring.com) |
| 06:20.34 | exten123 | how can we overcome the extensions when AGI fail? |
| 06:20.55 | *** part/#asterisk marktt (n=marktt@203.217.18.2) |
| 06:22.16 | nain | Among Predictive dialing which one is best and stable among both astguiclient and gnudialer |
| 06:24.06 | frk2 | ive had good luck with astguiclient |
| 06:24.15 | frk2 | just bad luck with phones |
| 06:24.16 | frk2 | :) |
| 06:24.41 | exten123 | frk2:what u mean with phones? |
| 06:26.04 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 06:28.07 | frk2 | man... i bought ATCOM phones-- they suck in larger deployments |
| 06:28.13 | frk2 | then I got the grandstream GXP2000s |
| 06:28.16 | frk2 | the gxp-2000s |
| 06:28.43 | Z-Knight | how do you like the gxp-2000s? Are they easy to configure? |
| 06:29.03 | frk2 | they are nice.. decent features too |
| 06:29.09 | frk2 | my only issue is this one-way voice |
| 06:29.19 | frk2 | sometimes calls coming in from my PRI cannot hear the GXP-2000 |
| 06:29.28 | frk2 | rebooting the phone fixes the issue |
| 06:29.32 | Z-Knight | hmm |
| 06:29.51 | frk2 | however |
| 06:29.59 | frk2 | while this PRI issue is happening, local calls work JUST fine |
| 06:30.06 | frk2 | voip to voip calls i mean |
| 06:30.37 | Z-Knight | i just got one of those phones recently and I will hopefully be testing it out soon...we'll see what happens on my end |
| 06:31.25 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:31.28 | *** join/#asterisk Kumbang (n=kumbang@167.205.24.5) |
| 06:31.42 | frk2 | i think i need to setup a daily reboot |
| 06:32.33 | wasim | frk2: eww |
| 06:36.21 | *** join/#asterisk hansin321 (n=chatzill@c-67-174-182-21.hsd1.co.comcast.net) |
| 06:37.42 | *** join/#asterisk xilo (n=xilo@80.97.188.20) |
| 06:46.09 | frk2 | what do i do man |
| 06:46.17 | frk2 | the GXP also randomly becomes unreachable |
| 06:47.17 | frk2 | i wonder if its an issue with asterisk 1.2.1 now |
| 06:48.13 | blitzrage | uhh. 1.2.1 is old :) |
| 06:48.44 | frk2 | i know |
| 06:48.53 | frk2 | but i cant pinpoint the issue before i start blaming it on asteirsk |
| 06:49.11 | blitzrage | I don't know what the issue is -- so I can't comment :) |
| 06:54.08 | frk2 | issue is 1) GXP 2000 not responding to SIP INVITE.. so the asterisk consol say Called SIP/100 and then goes to sleep |
| 06:54.28 | frk2 | 2) GXP 2000 recieves an incoming call from a PRI and sometimes the calling party cannot hear the GXP |
| 07:00.12 | *** join/#asterisk tuxinator_linux (n=tuxinato@142.131.189.11) |
| 07:01.38 | nain | frk2: I am stuck with ntp (time synconization) while compiling astguiclient, the ip of time server is not live given in astguiclient scratch install |
| 07:04.05 | Rui | nain: I don't know if you can set the NTP server. If so, try time.nist.gov |
| 07:05.51 | nain | Rui: well, i will try it , |
| 07:06.21 | nain | Rui: one of my friend tried to setup astguiclient but he was unable to load leads in vicidial and custom format too.... |
| 07:07.56 | nain | so i am not sure that it will work too for me |
| 07:09.55 | *** join/#asterisk pengyong (n=lala@222.188.139.82) |
| 07:10.36 | *** join/#asterisk Djeli (n=vircuser@59.167.207.2) |
| 07:16.25 | *** part/#asterisk Djeli (n=vircuser@59.167.207.2) |
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| 07:17.25 | *** join/#asterisk vandien (i=www-data@aditu.dahltronics.de) |
| 07:18.08 | *** join/#asterisk Qwell[laptop] (n=Qwell[]@unaffiliated/qwell) |
| 07:19.09 | *** join/#asterisk medusaXX (n=medusaxx@p54A9869A.dip0.t-ipconnect.de) |
| 07:25.05 | *** join/#asterisk shifter (n=shifter@flotsam.internetofdeath.com) |
| 07:30.02 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 07:30.14 | shifter | anyone have pointers on getting ekiga to register with asterisk ? |
| 07:31.26 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 07:34.45 | Qwell[laptop] | shifter, It's just SIP |
| 07:36.31 | [av]bani | qwell, any luck with 7960/7970 sip yet |
| 07:36.35 | shifter | Qwell[laptop]: sure, but i get an auth error after i've set things up in sip.conf |
| 07:36.40 | Qwell[laptop] | [av]bani, next week |
| 07:36.58 | shifter | i saw a hint that i needed to set the realm, but that didn't seem to fix things |
| 07:45.44 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
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| 07:53.28 | *** join/#asterisk FreezeS (n=Gladius@86.35.81.54) |
| 07:53.40 | FreezeS | hello |
| 07:53.46 | FreezeS | I have a problem with a PRI card |
| 07:53.55 | FreezeS | hangup doesn't work |
| 07:55.26 | *** join/#asterisk file[laptop] (n=jcolp@142.131.190.116) |
| 07:55.56 | *** part/#asterisk hansin321 (n=chatzill@c-67-174-182-21.hsd1.co.comcast.net) |
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| 08:07.24 | *** join/#asterisk TonyM (n=TonyM@adsl-solo-80-168-225-203.claranet.co.uk) |
| 08:12.16 | *** join/#asterisk boos_ (n=boos@adsl-51-191.37-151.net24.it) |
| 08:12.18 | boos_ | hello |
| 08:12.57 | FreezeS | hello :) |
| 08:15.03 | *** join/#asterisk Becky75 (n=pirch@dsl-165-221-124.telkomadsl.co.za) |
| 08:15.08 | Becky75 | hey guys |
| 08:16.18 | *** join/#asterisk Vitux (n=LNX@cable-63-135-21-193.sudbury.dyn.personainc.net) |
| 08:16.29 | Becky75 | can some one help me pls. I have a PRI that is used by 3 companies. there is 3 numbers asigned to the same pri. how can i rename the caller ID using DNID? |
| 08:18.33 | florz | Anyone in here who can tell me how a SIP UAC, when its INVITE is being forked and is subsequently accepted by two peers and both peers start sending RTP data (which they should, shouldn't they?!) - how does the UAC decide which stream to play back?! |
| 08:19.10 | florz | erm, well the end doesn't quite fit the beginning of that sentence - I hope you can figure out what I meant nonetheless ;-) |
| 08:19.48 | wasim | Becky75: SetCallerID() |
| 08:19.58 | wasim | Becky75: doc/README.var |
| 08:22.14 | *** join/#asterisk chapeaurouge (n=chap@vilhost1.vision.lu) |
| 08:22.42 | wasim | Becky75: and nice chase btw ... |
| 08:23.08 | Becky75 | wasim : hmmm so i can get the number the guy has dialed and then change the caller id to the company name sa ABC stationary?.. |
| 08:23.23 | wasim | Becky75: oui |
| 08:23.46 | Becky75 | wasim cause there is 3 companies using the same telco line but it has 3 different hunting numbers |
| 08:24.27 | Becky75 | i have to basicaly get the number the guy has dialed not his caller ID and then replace his caller id with the company name |
| 08:24.35 | {zombie} | Becky75: if your phone displays callerID name as well as number, then I would suggest using SetCIDName |
| 08:24.49 | Becky75 | its a snom360 |
| 08:24.50 | {zombie} | that's what I do |
| 08:25.07 | {zombie} | can't remember if that does or not, I know the 320 doesn't |
| 08:25.10 | Becky75 | think it will support it... be gentle i am a newbie to asterisk.. still use to mitel *spit* |
| 08:25.38 | {zombie} | either that, or do SetCIDName(Company $CALLERIDNUM) |
| 08:25.48 | {zombie} | that way you get both the company name and the number on the same line |
| 08:26.33 | Curus | Some people put in a speak saying which company the call is for |
| 08:27.56 | Becky75 | {zombie} : there is 3 companies using the same telco line but the line has 3 different phone numbers what i can see on the voip-info is that i have to use ${dnid} to get the dialed number then i have to use that to replace the called id or the $CALLERIDNUM ?... |
| 08:28.14 | Becky75 | Curus i thought of that but its 3 seperate companies in one building trying to do things on the cheap |
| 08:28.27 | {zombie} | Becky75: you don't have to do that |
| 08:28.30 | Becky75 | the 3 has nothing in common at all accept the same receptionist |
| 08:28.32 | {zombie} | this is what I do: |
| 08:28.37 | {zombie} | exten => 0733379988,4,SetCIDName(PCBNE ${CALLERIDNAME}); |
| 08:28.38 | {zombie} | exten => 0893223444,4,SetCIDName(PCare ${CALLERIDNAME}); |
| 08:29.57 | Becky75 | {zombie} so if the caller has dialed 0733379988 then it will display the PCBNE name on the phone with the phone number of the caller?.. |
| 08:30.01 | {zombie} | right |
| 08:30.14 | {zombie} | well actually it shows the PCBNE and then the company name (if available) |
| 08:30.16 | {zombie} | and the number on the next line |
| 08:30.20 | {zombie} | because my phone shows both |
| 08:30.29 | {zombie} | but if yours doesn't just replace the CALLERIDNAME with CALLERIDNUM |
| 08:31.05 | {zombie} | by company name I mean company name of the person calling (reverse search based on their phone number) |
| 08:31.45 | Becky75 | {zombie} so where do i put this in in [default] ?. |
| 08:31.56 | {zombie} | you put that wherever your calls come in |
| 08:32.07 | {zombie} | usually not in [default] |
| 08:32.11 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 08:32.15 | {zombie} | anyway I'm afraid I have to run, chat later |
| 08:32.23 | Becky75 | so my call hits the [incomming] context form the pri |
| 08:32.37 | Becky75 | arrgg {zombie} just one second pls :> |
| 08:32.51 | wasim | Becky75: right, so put in there |
| 08:32.58 | Becky75 | aha |
| 08:33.00 | wasim | Becky75: wherever your zapata.conf throws the calls |
| 08:33.01 | Becky75 | now it makes sence |
| 08:33.08 | {zombie} | sorry, I have a windoze 2000 exchange server to recover :( |
| 08:33.15 | Becky75 | so where 089blahblah is i put the number of the company they dialing |
| 08:33.22 | wasim | Becky75: thats the DNIS |
| 08:33.35 | Becky75 | if that matches then it will show PCBNE |
| 08:35.27 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 08:36.08 | Becky75 | wasim {zombie} thanks you the guys that makes it worth asking questions here *mwah* |
| 08:38.28 | Becky75 | wasim i hope it rusts and falls to his anckles heh |
| 08:38.50 | wasim | only if he's in durban |
| 08:38.56 | *** join/#asterisk sulex (n=sulex@pdpc/supporter/active/sulex) |
| 08:40.08 | *** join/#asterisk medusaXX (n=medusaxx@p54A9869A.dip0.t-ipconnect.de) |
| 08:41.18 | Becky75 | wasim u in ZA ?... |
| 08:41.27 | wasim | Becky75: no, no |
| 08:42.28 | *** join/#asterisk P0L0 (n=tekn0@62-43-65-175.user.ono.com) |
| 08:43.06 | Becky75 | {zombie} hehe good luck i hate exchange with a pasion... |
| 08:43.20 | Becky75 | ok well lemme get back to work before i asult the recptionist :| |
| 08:45.51 | FreezeS | I can't hangup on my PRI card. Is this a usual problem ? |
| 08:46.05 | wasim | FreezeS: no |
| 08:46.54 | FreezeS | hmm... are there more ways to hangup ? |
| 08:47.04 | FreezeS | what is the default cause used ? |
| 08:47.16 | wasim | what does pri debug show? |
| 08:47.42 | FreezeS | ccserver*CLI> pri debug |
| 08:47.42 | FreezeS | No such command 'pri debug' (type 'help' for help) |
| 08:48.04 | FreezeS | zap debug isn't eighter |
| 08:48.28 | FreezeS | this is weird, because I remember I used it once... |
| 08:48.34 | wasim | [Mar 15 13:49:37] DEBUG[7012]: NEW_HANGUP DEBUG: Calling q931_hangup, ourstate Null, peerstate Release Request |
| 08:48.38 | wasim | [Mar 15 13:49:37] DEBUG[7012]: > Protocol Discriminator: Q.931 (8) len=9 |
| 08:48.40 | wasim | [Mar 15 13:49:37] DEBUG[7012]: > Call Ref: len= 2 (reference 31336/0x7A68) (Terminator) |
| 08:49.08 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 08:50.02 | FreezeS | wasim: so how do I enable debugging ? |
| 08:50.13 | *** join/#asterisk nagl (n=nagl@rih.zid-nw.wu-wien.ac.at) |
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| 08:54.17 | chapeaurouge | hi all.. I would like to have distinctive rings on internal calls.. using polycom phones. I tried adding a exten => _1XX,1,SipAddHeader(ALERT_INFO="INTERNAL_RING") in my extensions.conf, and define the alertInfo in the polycom sip.cfg, but it hangs. |
| 08:54.23 | chapeaurouge | does anyone have pointers? |
| 08:54.25 | *** join/#asterisk Zeeek (n=icechat5@pdpc/supporter/active/Zeeek) |
| 08:54.32 | Zeeek | hej! |
| 08:56.19 | Zeeek | what does anyone know about Astricon europe? |
| 08:56.36 | oej | Noone in europe knows anything about it |
| 08:56.55 | Zeeek | why not? |
| 08:57.18 | Zeeek | I hear you're coming here in May btw? |
| 08:57.28 | Zeeek | what's that about? |
| 08:57.30 | oej | Yes, I'll be all over Europe |
| 08:57.37 | oej | Training |
| 08:57.39 | oej | Training |
| 08:57.39 | Zeeek | ewww... car accident? |
| 08:57.52 | Zeeek | MeetAsterisk? |
| 08:57.59 | oej | Sure |
| 08:58.04 | Zeeek | ok |
| 08:58.12 | Zeeek | hey I'll buy you a beer |
| 08:58.19 | oej | Cool |
| 08:58.33 | Zeeek | or wine or fruit juice |
| 08:58.46 | oej | :-) |
| 08:59.40 | Zeeek | olle where is the info about what you're doing? I can post it to our mailing list |
| 09:00.02 | oej | http://edvina.net |
| 09:00.05 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 09:00.09 | oej | THe french meeting coming up later this week |
| 09:00.22 | Zeeek | ok, cool. Someone just announced the May thing but with no details |
| 09:00.40 | oej | On the mailing list? |
| 09:00.45 | oej | Rumours go fast |
| 09:01.02 | Zeeek | yeah ours. Just 20 people or so, but all "important" actors in the local asterisk scene |
| 09:01.16 | Zeeek | except me |
| 09:01.22 | Zeeek | I'm just a peasant |
| 09:01.39 | Zeeek | I formed the group though :) |
| 09:02.17 | MGSsancho | lol |
| 09:02.26 | Zeeek | are there French speakers involved in your training? |
| 09:02.49 | oej | Not yet. looking for trainers in various places in Europe |
| 09:03.28 | Zeeek | I'll tell our guys to check it out. MOst of them are competent engineers with telcom backgrounds |
| 09:03.34 | Zeeek | except me... |
| 09:03.51 | Zeeek | I'm self taught ignoramus |
| 09:03.52 | chapeaurouge | oej, we might be providing training for Asterisk in the future |
| 09:04.19 | oej | Chapeaurouge: Where are you based? |
| 09:04.21 | Zeeek | chapeaurouge what country? |
| 09:04.23 | chapeaurouge | luxembourg |
| 09:04.26 | Zeeek | like minds and allthat |
| 09:04.28 | FreezeS | so, how do I enable zap debugging ? |
| 09:07.24 | Zeeek | chapeaurouge I know one of your princes |
| 09:07.45 | chapeaurouge | not mine... im not luxembourgish. I just work here ;) |
| 09:08.07 | Zeeek | still... you have to swear fidelity |
| 09:08.10 | chapeaurouge | na |
| 09:08.24 | Zeeek | to digium :) |
| 09:08.27 | chapeaurouge | heh |
| 09:10.13 | Zeeek | why do some people say X100P and others X101P? |
| 09:10.25 | Zeeek | my digium cards are reported as 101 |
| 09:10.41 | oej | And why don't most people want to talk about X10x at all? |
| 09:10.57 | chapeaurouge | Zeeek, bc of a possible daugther card? (/me not using digium cards) |
| 09:11.01 | chapeaurouge | well |
| 09:11.02 | chapeaurouge | wait |
| 09:11.35 | chapeaurouge | yea. |
| 09:11.40 | chapeaurouge | with sangoma, it would be bc of the daugther card i guess |
| 09:11.55 | Zeeek | well I bought my first cards as a devel kit |
| 09:12.19 | Zeeek | I then added a second X101P and a couple of FXS boards for the TDM400 |
| 09:12.52 | Zeeek | I hate to fix what works, so I've never replaced this stuff with the newer TDM400 and FXO daughter boards I bought 2 years ago |
| 09:13.25 | Zeeek | I have had the three cards in a Pentium III for two years and they work fine (low traffic) |
| 09:15.47 | Zeeek | is anyone having problems with 800 numbers being called by pollsters and other commercial crap? |
| 09:16.29 | *** join/#asterisk fourcheeze (n=rich@westbury.doilywood.org.uk) |
| 09:16.51 | *** join/#asterisk exten123 (n=exten@60.49.6.190) |
| 09:16.58 | tzafrir | Is there any way to get a stereo recording of a zaptel channel? |
| 09:17.18 | Zeeek | is the source stereo? |
| 09:17.30 | oej | I think you can mix like that in sox, tzafrir |
| 09:17.35 | wasim | tzafrir: sox it |
| 09:17.42 | Zeeek | hey wasim |
| 09:17.45 | *** join/#asterisk lojikop (i=lojikop@c-67-184-110-194.hsd1.il.comcast.net) |
| 09:17.48 | Zeeek | long time! |
| 09:17.50 | wasim | bonjour monsieur Zeeek |
| 09:17.51 | tzafrir | can I be guaranteed that the two files are in sync? |
| 09:18.06 | wasim | tzafrir: oh you mean inbound in left channel, and outbound in right? |
| 09:18.26 | tzafrir | Zeeek, the call in Asterisk is always stereo |
| 09:18.43 | Zeeek | you mean the two parties? |
| 09:18.44 | x86 | tzafrir: you are mis-informed if you think that ;) |
| 09:18.54 | wasim | tzafrir: no, its mono always, afaik |
| 09:19.07 | Zeeek | or like the bass and drums are on the left and the saxophone on the right? |
| 09:19.13 | tzafrir | I mean: there are two mono channels. |
| 09:19.20 | x86 | no |
| 09:19.24 | x86 | one mono channel |
| 09:19.25 | Zeeek | two mono channelms != stereo |
| 09:19.40 | x86 | err, one in each direction i guess ;) |
| 09:19.45 | x86 | aka full duplex |
| 09:20.07 | lojikop | sup d00ds, who's got time for a dumb question? |
| 09:20.22 | wasim | lojikop: jbot |
| 09:20.27 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 09:20.36 | tzafrir | lojikop, ask anyway |
| 09:20.50 | tzafrir | Gunnar has |
| 09:21.02 | exten123 | hey what version of MySQL is the best in Fedora Core 3? I try to install dependency checker keep said perl(DBI) require event been install. |
| 09:21.05 | lojikop | heh, i'm on aah v2.5, trying to create a DID to point to a custom extension that in turn points to *98 |
| 09:21.18 | lojikop | idea ... get users to get the generic comedian mailbox/password prompt |
| 09:23.05 | wasim | lojikop: use Goto() |
| 09:23.43 | lojikop | well the way AMP is setup it asks for a parameter for Dial() |
| 09:23.53 | wasim | lojikop: use vi |
| 09:23.56 | X-Rob | lojikop: |
| 09:23.57 | X-Rob | ~amp |
| 09:23.59 | jbot | i guess amp is "amp is, like, NOT supported here! people using it should join #freepbx (the new name of amp)" |
| 09:24.07 | tzafrir | wasim, so I guess my question is: if I monitor a call, how much are those two sound files (-in and -out) guaranteed to be in sync? |
| 09:24.13 | lojikop | good point jbot ;) |
| 09:24.36 | wasim | tzafrir: fairly in sync afaik |
| 09:25.00 | tzafrir | lojikop, generally AMP is nice to get started and troublesom to customize |
| 09:25.09 | Zeeek | how can I batch convert a zillion MS Word file to PDF? |
| 09:25.31 | lojikop | tzafrir: you're right, i'm kinda done with the startup, now i'm running into customization issues |
| 09:25.36 | tzafrir | antiword can convert them to SP |
| 09:25.40 | tzafrir | PS, that is |
| 09:25.45 | tzafrir | So can word-view |
| 09:25.52 | lojikop | i mean i could do the Goto() in extensions_custom i think |
| 09:25.55 | tzafrir | That will probably not preserve links, though |
| 09:26.02 | Zeeek | no links |
| 09:26.09 | X-Rob | lojikop, really, join #freepbx. |
| 09:26.10 | wasim | http://www.xml.com/lpt/a/2006/01/11/from-microsoft-to-openoffice.html |
| 09:26.15 | Zeeek | text and an image |
| 09:26.19 | wasim | might contain a tip or two |
| 09:26.33 | Zeeek | I don't have openoffice |
| 09:26.41 | Zeeek | well, I do on WIndows but not linux |
| 09:26.47 | tzafrir | openoffice and batch conversion? bah. abiword is nicer for batch operetions |
| 09:27.01 | tzafrir | And is also much smaller |
| 09:27.04 | Zeeek | love the name anyway |
| 09:29.55 | *** join/#asterisk lorinc (n=ang@caracas-2528.adsl.interware.hu) |
| 09:32.26 | blkremedy | anybody here use weather underground script? |
| 09:35.21 | X-Rob | the weather underground is easy 'Dark. Hot. You might get eaten by a grue' |
| 09:36.31 | X-Rob | commonwealth games are starting.. *wanders off to watch the ceremony* |
| 09:43.41 | chapeaurouge | is the native MOH supporting mp3 in 1.2.5? |
| 09:49.39 | blkremedy | how do you configure weather.agi for japan? |
| 09:56.09 | *** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it) |
| 09:56.35 | tzafrir | chapeaurouge, if you don't want to stream mp3-s from an external source, convert them to wavs and save CPU cycles |
| 09:56.50 | chapeaurouge | native will play .wav? |
| 09:56.51 | tzafrir | wavs, or sln, or whatever |
| 09:57.35 | tzafrir | Asterisk is known to be able to play .wav files, in the proper encoding, yeah. Take a look at the voicemail box |
| 09:57.43 | chapeaurouge | kool, thx |
| 10:16.23 | *** join/#asterisk DingBack (n=dingbat@host-81-191-147-145.bluecom.no) |
| 10:24.32 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 10:25.09 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 10:25.46 | astra^^ | hello all |
| 10:28.36 | *** join/#asterisk backblue (n=igor@82.102.1.42) |
| 10:28.38 | backblue | hi |
| 10:28.48 | Becky75 | how can i pusch voice though a speaker to a noisy factory to call some one to pick up a phone extention?.. |
| 10:30.00 | x86 | you cant |
| 10:30.21 | astra^^ | wen i use wget cmd i am gettin an error permission denied ... |
| 10:31.53 | frk2 | wasim- are you the same wasim from convergence? |
| 10:35.10 | backblue | anyone knows libassman? |
| 10:35.21 | *** join/#asterisk Aze` (n=aze@85-18-136-114.ip.fastwebnet.it) |
| 10:36.10 | *** join/#asterisk areski (n=areski@polar.es6.egwn.net) |
| 10:38.13 | *** join/#asterisk pif (n=ldm@zenon.apartia.fr) |
| 10:40.34 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 10:42.27 | markit | Hi, I've set FWD and other VoIP provider context to "incoming-voip", where first line is exten => s,1,Dial(${I_SIP_IN_RING,20,t), but today one called me and I've got this message |
| 10:42.29 | markit | Mar 15 10:20:03 NOTICE[28264]: chan_iax2.c:7213 socket_read: Rejected connect attempt from 192.246.69.186, request '635131@incoming-voip' does not exist |
| 10:42.46 | markit | any clue? |
| 10:43.30 | markit | (using asterisk 1.2.x svn, just updated and recompiled yesterday) |
| 10:43.32 | x86 | sounds like you dont have a matching extension in sip.conf |
| 10:43.41 | fourcheeze | markit: missing a closing } |
| 10:43.49 | markit | argh! |
| 10:43.53 | x86 | that too ;) |
| 10:44.16 | markit | fourcheeze: I could have spent months looking at this.. thanks a lot :)) |
| 10:44.16 | fourcheeze | use emacs ;-) |
| 10:44.30 | markit | btw, is there a way to make asterisk parse the dialplan and tell about these kinds of errors? |
| 10:46.50 | ambriento | well, at the CLI with some level of verbosity you'll see asterisks complaining about |
| 10:55.45 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 10:56.38 | frk2 | jbalcomb - You there? |
| 10:59.06 | astra^^ | i need some help to set up asterisk on my system .. |
| 10:59.15 | astra^^ | anyone can help me please.. ? |
| 10:59.35 | x86 | whats your question? |
| 10:59.59 | astra^^ | wen i untar it i get errors |
| 11:00.24 | astra^^ | asterisk-1.2.5/configs/features.conf.sample |
| 11:00.25 | astra^^ | tar: asterisk-1.2.5/configs/features.conf.sample: Cannot open: No such file or directory |
| 11:01.28 | vgster | why are yopu trying to untar a non tar file? |
| 11:01.59 | astra^^ | its a tar file asterisk-1.2.5.tar.gz |
| 11:02.00 | astra^^ | ? |
| 11:02.06 | msw | astra^^: out of disk space? |
| 11:02.11 | vgster | are you doing tar zxvf or just tar? |
| 11:02.24 | astra^^ | tar -zxvf |
| 11:02.25 | *** join/#asterisk X-Rob (n=rob-x@dsl-202-173-151-24.qld.westnet.com.au) |
| 11:02.44 | astra^^ | naa nt out of space.. |
| 11:02.49 | vgster | does it stop on that file? |
| 11:03.01 | x86 | sounds like a corrupt tarball |
| 11:03.04 | x86 | re-download it |
| 11:03.07 | vgster | yes |
| 11:03.53 | astra^^ | i have installe dthe same on other machine and it worked . . |
| 11:04.00 | astra^^ | i used pscp to transfer |
| 11:04.14 | astra^^ | the same copy . |
| 11:06.03 | x86 | it's corrupt |
| 11:06.03 | x86 | get over it ;) |
| 11:06.20 | x86 | re-download and stop arguing with us |
| 11:06.22 | mogorman | or you are out of diskspace... |
| 11:06.30 | Zeeek | corruption is rampant these days |
| 11:06.40 | x86 | out of disk space would give insufficient disk space errors |
| 11:06.46 | mogorman | i blame the schools Zeeek |
| 11:06.54 | astra^^ | hmmm.. i'll try reinstalling .. and one more thing.. |
| 11:06.55 | Zeeek | the parents too |
| 11:07.30 | astra^^ | weni use wget i get 404 error |
| 11:07.33 | x86 | that would give permission denied errors |
| 11:07.42 | astra^^ | exactly |
| 11:07.44 | x86 | are you all linux newbies here? :P |
| 11:07.55 | astra^^ | yes enterprice edition 4 |
| 11:07.55 | x86 | corrupt tar file, deal with it ;) |
| 11:07.57 | *** join/#asterisk arcy (n=arcanum@ppp88-adsl-138.ath.forthnet.gr) |
| 11:08.08 | astra^^ | wht abt cvsroot |
| 11:08.14 | x86 | mogorman: eh? |
| 11:08.17 | Zeeek | some version sof tar have trouble with long file names |
| 11:08.30 | mogorman | gnite folks |
| 11:08.32 | x86 | mogorman: my point was valid |
| 11:08.33 | x86 | :P |
| 11:08.53 | mutilator | anyone know why answering machines don't like ata's? |
| 11:08.56 | mutilator | and is there any way to fix it? |
| 11:09.00 | msw | mogorman: maybe you'll be awake by the time I land in CA for VON |
| 11:09.06 | mutilator | the machine records the call after it's done |
| 11:09.11 | mutilator | like busy signal for quite a while |
| 11:11.12 | *** join/#asterisk powerchip (i=powerchi@197.80-202-229.nextgentel.com) |
| 11:19.55 | *** join/#asterisk slePP (n=slepp@S0106000f663692da.ed.shawcable.net) |
| 11:19.59 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 11:24.16 | astra^^ | x86: got the problm |
| 11:24.35 | astra^^ | am nt on # am in $ |
| 11:25.01 | astra^^ | cvs [checkout aborted]: cannot make directory zaptel: Permission denied |
| 11:25.10 | astra^^ | [admin@enterprise src]$ cvs checkout zaptel libpri asterisk |
| 11:25.11 | astra^^ | cvs [checkout aborted]: cannot make directory zaptel: Permission denied |
| 11:27.11 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 11:29.13 | *** join/#asterisk bmrocha (n=bruno@82.102.1.42) |
| 11:29.54 | *** join/#asterisk florz (i=nobody@2001:1a50:503c:0:0:0:0:1) |
| 11:31.17 | *** join/#asterisk MacWinner (n=amit@rso.suspicious.org) |
| 11:32.26 | MacWinner | hi, if I have a VoIP gateway setup on an asterisk box, does the provider leave it up to me with how to deal with multiple callers to my number? |
| 11:33.00 | MacWinner | for example, if more than one person calls, do they just forward the call to me anyway and it's up to me to send back a busy signal or not? |
| 11:33.35 | MacWinner | my goal is to setup a simple VoIP conference bridge. |
| 11:38.50 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 11:39.00 | Dr-Linux | hi |
| 11:39.15 | Zeeek | hey doc |
| 11:39.24 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 11:39.44 | Dr-Linux | one of my US client wants recommended Digium card for T1 |
| 11:40.13 | Dr-Linux | should i recommend >> Wildcard TE110P |
| 11:40.19 | Dr-Linux | or what? |
| 11:41.34 | Dr-Linux | anybody acitve? :S |
| 11:42.25 | Zeeek | nobody knows (the trouble I've seen) |
| 11:43.06 | Dr-Linux | they are all sleeping, except jbot |
| 11:47.11 | glm2k | Dr-Linux: how many phones? |
| 11:48.44 | *** join/#asterisk f7950qs0 (n=redhotte@61.17.213.99) |
| 11:48.58 | f7950qs0 | has anyone tried pbxnsip? |
| 11:50.15 | f7950qs0 | nobody interested |
| 11:51.19 | f7950qs0 | PBXNSIP? anyone heard bout it? |
| 11:51.23 | *** join/#asterisk Nag (n=NetAdmin@LSt-Amand-152-31-11-135.w82-127.abo.wanadoo.fr) |
| 11:51.26 | Nag | Hi ! |
| 11:52.56 | Zeeek | no never hoid of it |
| 11:54.07 | Dr-Linux | glm2k: it will be using for IVR production IVR |
| 11:54.36 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 11:54.43 | Dr-Linux | glm2k: users will get check their balance etc using asterisk IVR |
| 11:55.16 | Nag | can someone tell me if it's possible to set a MusicOnHold when a call whit a softphone is etablished ? (example --> i call my number, Answer ---> Music 1, the personn on Softphone 100 Take The Call, he push the Hold Button of is softphone ---> Music 2) ?? |
| 11:58.23 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 11:58.47 | sternn | Nag: Could you use SetMusicOnHold() to do that? |
| 11:59.12 | Nag | sternn ---> i don't how to do that.. |
| 11:59.24 | Nag | i have made many classes in my musiconhold.conf |
| 12:00.02 | Nag | but i always have the same music |
| 12:00.24 | Nag | when i push the hold button, or on the automatic Answer when calling |
| 12:01.30 | *** join/#asterisk Abbas (n=Abbas@203.81.222.169) |
| 12:03.01 | *** join/#asterisk f7950qs0 (n=redhotte@61.17.213.99) |
| 12:03.06 | sternn | on the incoming call context SetMusicOnHold(blues), then in a hold context (not sure if that is right) SetMusicOnHold(jazz) |
| 12:03.59 | sternn | Not sure exactly how to write the dialplan for it, but I think the SetMusicOnHold() is what you would use to change the music classes. |
| 12:04.22 | Nag | yes , i think it's that, but it did not working |
| 12:05.35 | f7950qs0 | is there an alternative to G729 |
| 12:05.41 | f7950qs0 | I dont want to purchase licences |
| 12:05.52 | f7950qs0 | cause I am not sure how long will I use those licences |
| 12:06.45 | fourcheeze | f7950qs0: lots of alternatives |
| 12:06.51 | fourcheeze | depends what youwant |
| 12:07.03 | fourcheeze | have you tried gsm? |
| 12:07.14 | f7950qs0 | but my voip provider does not support gsm |
| 12:07.24 | fourcheeze | does your voip provider support g729? |
| 12:07.29 | f7950qs0 | yes he does |
| 12:07.34 | fourcheeze | and ulaw/alaw ? |
| 12:07.35 | f7950qs0 | G711 and G729 |
| 12:07.37 | fourcheeze | ok |
| 12:07.38 | f7950qs0 | yes |
| 12:07.49 | *** join/#asterisk giggles (n=chatzill@ool-18bb0d86.dyn.optonline.net) |
| 12:07.50 | fourcheeze | well if you are sending g729 straight through there is no license required |
| 12:08.13 | f7950qs0 | i am not using asterisk but dont get offended guys i wanna use a third party pbx which is also free |
| 12:08.23 | fourcheeze | which is that? |
| 12:08.34 | f7950qs0 | I dont know if their software works as a pass through i sent an e mail to them and dint get the response |
| 12:08.41 | f7950qs0 | Axon |
| 12:08.51 | fourcheeze | don't know that one, might have to have a look |
| 12:08.57 | fourcheeze | is it properly free or just beer free? |
| 12:09.08 | fugitivo | free or opensource? |
| 12:09.17 | f7950qs0 | it's free |
| 12:09.17 | fourcheeze | yeah, is it open source |
| 12:09.22 | f7950qs0 | i dont think it's opensource |
| 12:09.56 | fourcheeze | got a url? |
| 12:10.24 | f7950qs0 | nch.com.au |
| 12:10.50 | fourcheeze | uch, it needs windows <shudder> |
| 12:11.21 | f7950qs0 | i think they have linux version as well |
| 12:12.11 | *** join/#asterisk apardo (n=apardo@87.218.44.116) |
| 12:13.47 | f7950qs0 | extensions Must support symmetric UDP signalling and audio with reINVITEs |
| 12:13.59 | f7950qs0 | does that mean that they WILL USE G711? |
| 12:14.22 | f7950qs0 | I need this for simple callshop billing |
| 12:14.32 | Nag | nobody for my question please ? |
| 12:15.04 | fourcheeze | nag: show application dial |
| 12:15.10 | f7950qs0 | devices dont monitor minutes and the website doesn't provide instant minute usage |
| 12:15.21 | f7950qs0 | any ideas? |
| 12:15.38 | fourcheeze | f7950qs0: I would think about using something like SER |
| 12:15.59 | f7950qs0 | is it technical like asterisk? |
| 12:16.03 | fourcheeze | hehe |
| 12:16.07 | fourcheeze | all VOIP is technical |
| 12:16.18 | fourcheeze | the things that make you think it isn't are the worst |
| 12:16.23 | Zeeek | that's why there are consultants |
| 12:16.32 | f7950qs0 | is that from iptel? |
| 12:16.35 | f7950qs0 | or sipfoundry? |
| 12:16.35 | fourcheeze | yes |
| 12:16.39 | fourcheeze | iptel |
| 12:16.52 | fourcheeze | there is a standard config if you just want to use it to generate cdrs I think |
| 12:17.10 | f7950qs0 | oh cool and it works as a G729 pass through? |
| 12:17.14 | f7950qs0 | looks like opensource |
| 12:17.21 | fourcheeze | the good thing about it is that it won't handle rtp at all - so your endpoints are negotiating codecs between them |
| 12:17.39 | fourcheeze | how many simultaneous calls? |
| 12:18.06 | *** join/#asterisk puzzled (n=patrick@puzzled.xs4all.nl) |
| 12:21.24 | giggles | anyone know how to program a mediatrix 2102? |
| 12:22.02 | f7950qs0 | Seven simultaneous calls |
| 12:22.20 | fourcheeze | f7950qs0: ok, then I would use asterisk |
| 12:22.34 | f7950qs0 | is that a lot for SER? |
| 12:22.38 | fourcheeze | no |
| 12:22.46 | fourcheeze | no way near enough ;-) |
| 12:23.18 | f7950qs0 | i mean is that a lot or is that a lot less |
| 12:23.22 | fourcheeze | f7950qs0: if all your clients are g729 and your termination is, why not just use asterisk in pass through? |
| 12:23.38 | fourcheeze | 7 simultaneous calls is practically nothing |
| 12:24.24 | f7950qs0 | cause I dont know how to use asterisk |
| 12:24.32 | fourcheeze | yeah, we've all been there ;-) |
| 12:24.50 | f7950qs0 | I dont even know how to create a dial plan |
| 12:24.56 | fourcheeze | ~wiki |
| 12:25.05 | f7950qs0 | I read the tutorial |
| 12:25.20 | Zeeek | read this: |
| 12:25.21 | Zeeek | The dialplan is the heart of an Asterisk system, as it defines how Asterisk should handle calls. |
| 12:25.21 | Zeeek | http://www.asteriskdocs.org/modules/tinycontent/content/docbook/current_v1/docs-html_one/vm1.html#AEN650 |
| 12:25.55 | Zeeek | even though it's old it gives the background |
| 12:26.04 | Abbas | hello zeeek |
| 12:26.10 | Zeeek | hi Abbas |
| 12:26.29 | Abbas | i need ur help in an issue |
| 12:26.49 | Abbas | frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 12:27.22 | Abbas | i am using xpro >> asterisk >> voipswitch |
| 12:27.27 | Zeeek | I think I've seen that before |
| 12:27.33 | Abbas | asterisk is 1.2.4 |
| 12:27.47 | fourcheeze | f7950qs0: how are you planning on billing people? |
| 12:27.53 | f7950qs0 | I have the whole tutorial of asterisk printed out in four books ! |
| 12:27.59 | Abbas | i have tried canreinvite = yes and no both for peers |
| 12:28.05 | Zeeek | now you only have to read one |
| 12:28.34 | Zeeek | <PROTECTED> |
| 12:28.36 | fourcheeze | f7950qs0: I fully understand that it looks impossible - I was in your shoes about 9 months ago |
| 12:28.46 | f7950qs0 | the scenario is : customer walks in and makes a call he comes out and I look it up in the web interface his number and tell him the minutes |
| 12:28.49 | fourcheeze | you just take the plunge |
| 12:29.15 | fourcheeze | ok, so there's no calling credit |
| 12:29.15 | MacWinner | when a VoIP provider gives you a phone number, and multiple people call it from their regular PSTN phones, is it up to you to decide how to handle the multiple calls? |
| 12:29.22 | Abbas | zeeek can u please help me in tht issue ? |
| 12:29.36 | f7950qs0 | is SER for windows? |
| 12:29.36 | fourcheeze | MacWinner: yes |
| 12:29.40 | Zeeek | Abbas no I don't know anything about it other than the fact that I've seen the error message |
| 12:29.52 | *** join/#asterisk sergeus (n=s@195.112.98.13) |
| 12:30.04 | Abbas | any one else can help me in this issue?? |
| 12:30.05 | Abbas | frame.c:179 __ast_smoother_feed: Dropping extra frame of G.729 since we already have a VAD frame at the end |
| 12:30.25 | MacWinner | fourcheeze, thanks. so if i have a broadvoice trunk, I can setup call bridges between 3+ people and i don't need to do anything special with broadvoice? |
| 12:30.57 | f7950qs0 | fourcheeze: http://www.yes-tele.com this is a perfect thing for my cafe but i can't afford it |
| 12:31.05 | I-MOD | Abbas: disable silence suppression on your end device |
| 12:31.12 | fourcheeze | MacWinner: well, I guess that it's down to Broadvoice |
| 12:31.25 | fourcheeze | but I've never encountered a problem with more than one incoming call |
| 12:31.47 | fourcheeze | f7950qs0: SER runs on Linux AFAIK |
| 12:31.52 | tzanger | jesus that yes-tele thing is using OLD technology |
| 12:31.53 | MacWinner | fourcheeze, do you use only IP trunks? which service provider? |
| 12:32.11 | Zeeek | f7950qs0 can't you just use the cdr-csv file of asterisk for billing times? |
| 12:32.12 | tzanger | you cold replace almost all of that digital stuff with a single $7 PIC |
| 12:32.13 | fourcheeze | MacWinner: Gamma Telecom right now |
| 12:32.19 | noky | hi |
| 12:32.27 | *** join/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 12:32.28 | noky | can i paste 4 lines of h323.log ? |
| 12:32.44 | fourcheeze | Zeeek: yeah, he could but he doesn't have an asterisk running :/ |
| 12:32.56 | tzanger | I mean they're using an external full-blown 16450 for christ's sake. What modern microcontroller doesn't have an onboard UART (and RAM/ROM for that matter)?? |
| 12:32.58 | noky | my asterisk's unregistering with my gatekeeper =( |
| 12:33.20 | fourcheeze | f7950qs0: are you familiar with hacking on any particular thing? |
| 12:33.37 | f7950qs0 | be a bit more specific please fourcheeze |
| 12:33.49 | fourcheeze | I mean coding, you know writing software |
| 12:34.06 | f7950qs0 | no i dont but i know some people who might |
| 12:34.09 | f7950qs0 | know |
| 12:34.15 | f7950qs0 | why? |
| 12:34.22 | MacWinner | fourcheeze, is your gamma connection a VoIP trunk? |
| 12:34.31 | fourcheeze | MacWinner: yes, pure SIP |
| 12:35.16 | Zeeek | fourcheeze it was an easy assumption to make since this is the asterisk channel :) |
| 12:35.18 | fourcheeze | f7950qs0: just that with basic skills you could write some stuff with DB access that made your job a bit easier |
| 12:35.27 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 12:35.43 | MacWinner | cool, so each incoming call looks like a seperate SIP session? and then you can use asterisk MeetMe to bridge them all? |
| 12:36.02 | MacWinner | just trying to get an idea of the pieces needed to be put together |
| 12:36.17 | fourcheeze | MacWinner: sure - we have a single number that goes to a meetme conference |
| 12:37.03 | fourcheeze | any number of people can join it |
| 12:37.07 | MacWinner | sweet! that's exactly what i need :) |
| 12:37.28 | MacWinner | does gamma charge any extras for handling more than one call at a time? |
| 12:37.35 | fourcheeze | f7950qs0: however evenwithout any coding skills you could open a CSV file in a spreadsheet I'm sure |
| 12:38.11 | Zeeek | what I do is to wget the asterisk csv, load it into mysql on another server and access that from the web |
| 12:38.34 | fourcheeze | I just store cdrs in mysql |
| 12:38.34 | Zeeek | it took about 5 minutes to write a php script to show the results |
| 12:39.00 | Zeeek | and of the 5 minutes, 4 wezre remembering what the mysql password was :) |
| 12:39.03 | fourcheeze | :-) |
| 12:39.10 | MacWinner | fourcheeze, just to clarify, your meetme solution is entirely IP based? |
| 12:39.20 | fourcheeze | MacWinner: how can I say this |
| 12:39.41 | MacWinner | i know, i'm asking simple/stupid questions :) |
| 12:39.45 | fourcheeze | If I ever have to touch another telecoms FXO/FXS/PRI/BRI or whatever card again it would be too soon |
| 12:39.57 | fourcheeze | I only ever do IP |
| 12:40.01 | Zeeek | give your soul to SIP |
| 12:40.04 | fourcheeze | indeed |
| 12:40.16 | fourcheeze | although we prefer IAX when we can get it |
| 12:40.25 | fourcheeze | which isn't often :-( |
| 12:40.35 | Zeeek | so you're in GoogleTalk denial |
| 12:40.54 | fourcheeze | are they doing IAX? |
| 12:41.00 | f7950qs0 | does anyone know if yahoo voip can be configured in any device or asterisk :P |
| 12:41.12 | Zeeek | people keep inviting me to googleTalk and I keep saying "get a SIP phone and I'll hook you into our system" |
| 12:41.30 | fourcheeze | google were going to open things up |
| 12:41.41 | Zeeek | someone has a page on yahoo SIP but it can stop anytime |
| 12:41.53 | Zeeek | stop working I mean |
| 12:41.56 | f7950qs0 | wow really |
| 12:42.03 | noky | it works now |
| 12:42.06 | fourcheeze | yeah, I mean Yahoo wouldn't want to create anything *useful* |
| 12:42.14 | Zeeek | <PROTECTED> |
| 12:42.20 | f7950qs0 | you're right bout that fourcheeze |
| 12:42.42 | Zeeek | . Yahoo is using TCP SIP and RTP codec SPEEX. |
| 12:42.42 | Zeeek | 2. For audio its using UDP |
| 12:42.43 | Zeeek | 3. The registrations is normal like SIP registration but via TCP |
| 12:42.44 | MacWinner | fourcheeze, does each incoming call look like a different SIP session? |
| 12:43.24 | Zeeek | better yet: http://nerdvittles.com/index.php?p=70 |
| 12:43.42 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F25CF.dip0.t-ipconnect.de) |
| 12:43.58 | Zeeek | this is dated sept 2005 and claims it works |
| 12:44.30 | Zeeek | but then you have to have a paid account |
| 12:44.30 | f7950qs0 | good thanks zeeek |
| 12:44.36 | Zeeek | $10/mo |
| 12:44.44 | f7950qs0 | i hope they have one cent a minute to australia or UK |
| 12:44.55 | Zeeek | unlimited North Am |
| 12:45.14 | f7950qs0 | north america meaning if I call texas is it gonna cost me? |
| 12:45.33 | Zeeek | no meaning N.A. unlimited for $10 |
| 12:45.44 | *** join/#asterisk saftsack (n=saftsack@p54A7E08A.dip.t-dialin.net) |
| 12:46.00 | saftsack | hi |
| 12:46.12 | saftsack | the asterisk mathfunction gives me an double (15.0000) back and not a normal 15. howto change this? |
| 12:46.22 | saftsack | couldnt find anything in voip-info.org |
| 12:46.43 | Zeeek | and the answers yoiu got yesterday here didn't work? |
| 12:48.08 | *** join/#asterisk __AK__ (n=ak@ns2.necstar.fr) |
| 12:48.58 | *** join/#asterisk rpm (i=russell@S010600111155e117.cg.shawcable.net) |
| 12:49.21 | f7950qs0 | other than yahoo's dialpad they have one more voip service |
| 12:49.31 | Zeeek | who does? |
| 12:49.34 | Zeeek | yahoo? |
| 12:49.34 | __AK__ | hello all, |
| 12:50.01 | __AK__ | anyone knows how i can test the pickup command before executing it |
| 12:50.19 | __AK__ | i'd like to test it, and if it return an error code do something else |
| 12:51.11 | f7950qs0 | yes yahoo |
| 12:51.15 | f7950qs0 | let me go check |
| 12:51.22 | f7950qs0 | they allow one cent a minute calls to many countries |
| 12:51.29 | f7950qs0 | i mean they provide :P |
| 12:51.56 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 12:52.08 | Zeeek | for you @hole afficionatos : http://nerdvittles.com/index.php?cat=3 |
| 12:52.22 | saftsack | Zeeek, didnt get any answers |
| 12:52.36 | Zeeek | someone mentioned CUT - did you look it up or try it? |
| 12:52.46 | saftsack | no i didnt |
| 12:52.51 | Zeeek | why not? |
| 12:52.52 | saftsack | CUT after getting the variable? |
| 12:53.01 | saftsack | because i didnt received the message |
| 12:53.07 | Zeeek | will you fucking go rzead some of this stuff? |
| 12:53.11 | fourcheeze | can asterisk handle any complicated variables natively - like lists? |
| 12:53.41 | *** join/#asterisk htims (n=htims@Vc97c.v.pppool.de) |
| 12:53.48 | saftsack | Zeeek, can you be a little bit more friendly? |
| 12:53.57 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 12:53.57 | *** mode/#asterisk [+o anthm] by ChanServ |
| 12:54.09 | Zeeek | safsack, you are on my ignore list. Why? Because you ignore the answers people give you |
| 12:54.37 | Zeeek | You were given a link to the CUT page on the wiki. You only have to reazd about 5 lines |
| 12:54.41 | saftsack | is it a crime to oversight an answer? |
| 12:54.53 | saftsack | and yes im on this page atm |
| 12:57.21 | fourcheeze | saftsack: does cut solve your problem? |
| 12:57.45 | fourcheeze | f7950qs0: much as I hate to recommend a@h you might want to give that a try |
| 12:58.07 | f7950qs0 | I tried it and remember you helped me start asterisk@home? |
| 12:58.16 | fourcheeze | ahh yeah |
| 12:58.29 | fourcheeze | ok then, have you tried xorcom ? |
| 12:59.12 | fourcheeze | f7950qs0: or freepbx |
| 12:59.16 | *** join/#asterisk FreezeS (n=Gladius@86.35.81.54) |
| 12:59.44 | fourcheeze | f7950qs0: xorcom is pretty straightforward, although basic |
| 12:59.50 | f7950qs0 | have you seen pbxnsip i found that really easy for my kind of a guy. I mean I am not afraid of trying things or spending time for it. for the cafe at the rates I provide the services it's not worth doing all the hard work |
| 13:00.00 | Zeeek | so is the yahoo messenger voip thing the same as dialpad? |
| 13:00.11 | f7950qs0 | xorcom is the first asterisk i downloaded |
| 13:00.29 | fourcheeze | ok |
| 13:00.35 | fourcheeze | how did you get on? |
| 13:00.52 | fourcheeze | shold be hard to go wrong with that |
| 13:00.53 | *** join/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 13:00.56 | FreezeS | I've got a problem. The 's' extension isn't run when I dial out from a sip account |
| 13:01.11 | chapeaurouge | got a pb with MOH... http://pastebin.ca/45798 |
| 13:01.34 | chapeaurouge | the files are there... *.mp3, from the * installation |
| 13:01.44 | MacWinner | thanks for the info guys.. have a great day! |
| 13:01.58 | FreezeS | chapeaurouge: check the rights |
| 13:02.07 | f7950qs0 | how did I get on? |
| 13:02.13 | chapeaurouge | FreezeS, yea, they seem ok. |
| 13:02.34 | FreezeS | and the rights on the folder |
| 13:02.49 | chapeaurouge | dont work with 777 either |
| 13:03.51 | chapeaurouge | bleh |
| 13:03.54 | chapeaurouge | works now |
| 13:04.02 | chapeaurouge | had to completely stop and restart asterisk |
| 13:04.04 | chapeaurouge | :\ |
| 13:04.05 | chapeaurouge | thx |
| 13:04.44 | FreezeS | :) |
| 13:05.22 | chapeaurouge | now on to distinctive ringing, which i cant make the polycom ring different :P |
| 13:05.58 | fourcheeze | what's the best way to iterate over a list in a dialplan? |
| 13:06.09 | f7950qs0 | www.pbxnsip.com |
| 13:06.14 | FreezeS | so, my problem is the following: I have a sip user, with the default context [phone]. The first line on [phone] is exten => s,1,Set(CALLERID(num)=207938) followed by some includes. The 's' extension isn't running |
| 13:06.32 | f7950qs0 | fourcheeze try that website i gave you |
| 13:06.58 | f7950qs0 | it has linux and windows both versions, costs 200 dollars plus 10 dollars for every G729 licence purchased |
| 13:07.04 | f7950qs0 | and it doesn't work as a passthrough |
| 13:07.17 | Zeeek | FreezeS the SIP user runs in the context when a call is received for her |
| 13:07.17 | fourcheeze | hehe |
| 13:07.18 | FreezeS | fourcheeze: exten => x,1,Dial(sip/user) exten => x,2,Dial(sip/user2) ... etc |
| 13:07.44 | FreezeS | but how can I run something everytime somebody dials out ? |
| 13:07.45 | fourcheeze | FreezeS: yeah I want a list of numbers to call |
| 13:07.54 | Zeeek | FreezeS in the dialplan |
| 13:07.58 | FreezeS | fourcheeze: that's the way to do it |
| 13:08.00 | FreezeS | like I told you |
| 13:08.24 | FreezeS | just replace sip/user with zap/line/number_to_dial |
| 13:08.24 | fourcheeze | FreezeS: I don't see how I get that out of a list |
| 13:08.37 | FreezeS | you edit extensions.conf by hand |
| 13:08.39 | fourcheeze | I want to write a macro that I can give a list of numbers |
| 13:08.44 | FreezeS | oh |
| 13:08.49 | FreezeS | learn AGI :) |
| 13:08.51 | fourcheeze | and it goes through them |
| 13:08.53 | fourcheeze | I know agi |
| 13:09.02 | fourcheeze | I'd rather use a macro if at all possible |
| 13:09.02 | Zeeek | how many numbers max? |
| 13:09.06 | fourcheeze | not many |
| 13:09.07 | f7950qs0 | did you see the website i gave you fourcheeze |
| 13:09.11 | f7950qs0 | oh you're busy with your own thing |
| 13:09.12 | fourcheeze | f7950qs0: yes |
| 13:09.23 | Zeeek | there is already a macro that does this... superdial or some such |
| 13:09.25 | fourcheeze | f7950qs0: I wouldn't pay money for something worse than astiersk |
| 13:09.28 | f7950qs0 | i find that the closest to my heart :) |
| 13:12.35 | fourcheeze | Zeeek / FreezeS: I'm trying to implement hunt groups based on DB lookups - if anyone knows of a Better Way (TM) |
| 13:12.52 | fourcheeze | Zeeek: looking at superdial |
| 13:13.44 | Zeeek | fourcheeze what about ael ? |
| 13:13.52 | fourcheeze | dunno what that is |
| 13:13.59 | Zeeek | seems like it'd be tailor-made for your situation |
| 13:14.10 | Zeeek | what version asterisk? |
| 13:14.14 | fourcheeze | 1.2.4 |
| 13:14.29 | Zeeek | in that case... http://www.voip-info.org/wiki/view/Asterisk+AEL |
| 13:14.48 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 13:15.03 | Zeeek | this would allow you to loop thru a list AFAIK |
| 13:15.13 | Zeeek | also http://scottstuff.net/blog/articles/2005/10/10/asterisk-extension-language |
| 13:15.24 | *** part/#asterisk caio1982 (i=caio1982@CAcert-br/caio1982) |
| 13:15.33 | fourcheeze | Zeeek: well I'm thinking if I can get to the first item in a list I can recurse using the Local channel |
| 13:15.45 | FreezeS | fourcheeze: I'll need to make something like that myself |
| 13:15.57 | fourcheeze | I suppose I can use a pipe or soemthing in a string |
| 13:16.00 | Zeeek | switch(${DIALSTATUS}) { |
| 13:16.00 | Zeeek | <PROTECTED> |
| 13:16.02 | Zeeek | look, this is perfect, switch(${DIALSTATUS}) { |
| 13:16.18 | fourcheeze | hmm |
| 13:16.27 | Zeeek | except it's backwards in my paste |
| 13:16.30 | fourcheeze | does it coexist with ordinary dialplans? |
| 13:16.35 | Zeeek | I think so |
| 13:16.43 | Zeeek | but I may be wrong about that |
| 13:16.58 | fourcheeze | looks a bit experimental |
| 13:17.02 | fourcheeze | otherwise ideal |
| 13:17.25 | *** join/#asterisk kardecallan (n=kardecal@ns1.pcma.com.br) |
| 13:17.45 | jsharp | processed cheeze food |
| 13:17.52 | fourcheeze-food | nor processed |
| 13:17.54 | Zeeek | actually you could easily write a macro too as you planned |
| 13:17.54 | fourcheeze-food | not |
| 13:18.12 | fourcheeze-food | how would I slice the first number off a string? |
| 13:18.18 | fourcheeze-food | is that a job for cut? |
| 13:18.22 | noky | hi |
| 13:18.25 | Zeeek | you're pulling mly leg |
| 13:18.37 | fourcheeze-food | no |
| 13:18.43 | fourcheeze-food | well maybe |
| 13:18.47 | FreezeS | but, how can I set the CALLERID when a user dials out ? |
| 13:18.48 | Zeeek | heh |
| 13:18.55 | noky | i need to set in extension.conf some dialplan to comunicate my Asterisk with a Gatekeeper |
| 13:18.59 | noky | how can i do this?? |
| 13:19.01 | noky | extend => ? |
| 13:19.07 | noky | in [default] ? |
| 13:19.11 | fourcheeze-food | the point is that if you recurse you need to send the top of the string to the function and save the rest |
| 13:19.26 | fourcheeze-food | I'll have to give it some thought |
| 13:19.37 | Zeeek | FreezeS whatever they dial, go thru a dialplanextension or macro that does all that |
| 13:19.49 | *** join/#asterisk mover (n=dlu@gw-dus-net.dus.de.ncore.net) |
| 13:20.02 | mover | hi |
| 13:20.32 | mover | how i can set the Accountcode in svn trunk? before there was an app SetAccount |
| 13:20.35 | *** join/#asterisk scubasteve (n=steve@ns1.misel.com) |
| 13:20.43 | Zeeek | still is |
| 13:20.47 | mover | no |
| 13:20.50 | mover | isnt |
| 13:20.54 | *** part/#asterisk scubasteve (n=steve@ns1.misel.com) |
| 13:20.54 | FreezeS | my problem is that for dialout I have a context [dialout] exten => _X.,1,Dial(Zap/g1/${EXTEN}) |
| 13:21.25 | FreezeS | and I include that context for all the users that need to dial out using the PRI line |
| 13:21.25 | Zeeek | is so! |
| 13:21.52 | FreezeS | but before the dial, I must set the CALLERID |
| 13:21.54 | mover | Zeeek: res_agi.c:1095 handle_exec: Could not find application (SetAccount) |
| 13:22.06 | Zeeek | no it changed name |
| 13:22.26 | Zeeek | what versikn? I have that in 1.2 |
| 13:22.37 | mover | zeek to what? |
| 13:22.38 | *** join/#asterisk Pix (i=pix@crazyfrogs.org) |
| 13:22.51 | jsharp | exten => _X.,1,SetCIDNum(8675309) |
| 13:22.58 | Zeeek | <PROTECTED> |
| 13:23.02 | mover | sterisk SVN-trunk-r12430M |
| 13:23.05 | Zeeek | try this^^^^ |
| 13:23.06 | jsharp | exten => _X.,2,Dial(Zap/g1/${EXTEN}) |
| 13:23.15 | RoyK | jsharp: set(CALLERID(number)=1234) |
| 13:23.20 | mover | zeek you are the best!!!!!!!!! |
| 13:23.22 | jsharp | Or that too. |
| 13:23.23 | *** join/#asterisk h4mm3r` (n=h4mm3r@85-18-14-10.fastres.net) |
| 13:23.24 | mover | thank you!!! |
| 13:23.30 | Zeeek | mover one google brought that up |
| 13:23.41 | Zeeek | google asterisk setaccount |
| 13:23.56 | Zeeek | the answer was in the google summaryt! |
| 13:23.57 | mover | hehe i will remember |
| 13:23.58 | FreezeS | jsharp, I was thinking to modify the [dialout] context to change the priority to 2 |
| 13:24.10 | mover | i goggled to Accountcode |
| 13:24.27 | jsharp | Priority 1: Set CID. Priority 2: Dialout. |
| 13:24.28 | Zeeek | setaccount has been deprecated |
| 13:25.30 | *** join/#asterisk jimmy_deanPB (n=jhodapp@indianalifesciences.com) |
| 13:26.14 | _Paulo_ | ~seen coppice |
| 13:26.32 | jbot | coppice <n=chatzill@211.155.168.153> was last seen on IRC in channel #asterisk, 1d 3h 25m 30s ago, saying: '"Nun"'. |
| 13:28.07 | FreezeS | it worked |
| 13:29.47 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 13:29.54 | Zeeek | ael and extensions.conf can cohabit nicely |
| 13:30.17 | Zeeek | so you could write a small ael to solve the hunt group thing |
| 13:30.40 | *** part/#asterisk Pix (i=pix@crazyfrogs.org) |
| 13:33.14 | kardecallan | Is there anybody that can help me with configuration of the Asterisk behind the Firewall? |
| 13:33.35 | Zeeek | ask away |
| 13:35.59 | kardecallan | I only configured asterisk with sip, when I receive a call external I do not obtain to establish the audio one. |
| 13:36.19 | Zeeek | describe the network, client and asterisk |
| 13:36.27 | *** join/#asterisk [ProB]CrazyMan (n=Tobias@p549F12C1.dip0.t-ipconnect.de) |
| 13:38.05 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool138-128.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 13:40.39 | kardecallan | Asterisk is with IP 10.75.2.30 mine firewall has ip 10.75.2.1 (LAN) and ip 201.45.22.140(WAN) the external customer meets in the InterNet. |
| 13:42.40 | *** part/#asterisk sandos (n=sandos@me-0-50-da-e0-bb-36.2.cust.bredband2.com) |
| 13:45.25 | kardecallan | Asterisk is with IP 10.75.2.30 mine firewall has ip 10.75.2.1 (LAN) and ip 201.45.22.140(WAN) |
| 13:45.25 | kardecallan | the external customer accesses our server via InterNet. |
| 13:45.29 | tzafrir | I asked here earlier about stereo and such. After some googling I landed on a page from pkg-voip: the 2wav2mp3 script... |
| 13:46.39 | RoyK | tzafrir: I have a small script that uses sox to mix monitor in stereo |
| 13:47.01 | kardecallan | Zeeek. Sorry! I'm brazilian |
| 13:47.13 | kippi | anyone used asterisk on freebsd? any ideas where it puts the files? eg, extension.conf |
| 13:47.40 | kardecallan | I have difficulty to write in english |
| 13:47.56 | kardecallan | but I understand well |
| 13:48.50 | exonic | kippi, I imagine /etc/asterisk/ |
| 13:49.00 | exonic | kippi, if its your first install, run 'make samples' |
| 13:49.53 | *** join/#asterisk Bambr (n=Bambr@213-35-239-33-dsl.end.estpak.ee) |
| 13:50.20 | kippi | make: don't know how to make samples. Stop |
| 13:51.11 | Zeeek | kardecallan are you forwarding ports to asterisk from the firewall? |
| 13:51.27 | kardecallan | yes |
| 13:51.39 | *** join/#asterisk shiznatix (n=Bambr@213-35-239-33-dsl.end.estpak.ee) |
| 13:51.54 | Zeeek | ports 10,000-11,000 ? |
| 13:52.03 | RoyK | kippi: cd samples && for i in *; do cp $i /etc/asterisk/`echo $i | sed s/.sample//`; done |
| 13:52.36 | kardecallan | The rule of firewall is to repass everything for the server asterisk. |
| 13:52.53 | Zeeek | what phone is client? |
| 13:53.02 | shiznatix | Hello, I am trying to send a fax from asterisk then save it back onto the asterisk server. I am able to start the send (I think) but I want the fax to be saved on the asterisk server under a specified directory instead of dialing the phone |
| 13:53.31 | shiznatix | I have of course read the wiki 5 times over and tried just about everything without success |
| 13:55.17 | kardecallan | I observed in the debug of the Asterisk that, |
| 13:55.17 | kardecallan | when it goes to send solicitation RTP it is sending for the local IP of my remote customer |
| 13:56.34 | Zeeek | then the client is screwed up! |
| 13:56.36 | fourcheeze-food | what do people do to avoid race conditions? |
| 13:56.47 | fourcheeze | in dialplans |
| 13:56.48 | Zeeek | live in white trash neighborhoods? |
| 13:56.54 | glm2k | lol |
| 13:57.08 | fourcheeze | e.g. |
| 13:57.09 | Zeeek | move away from NASCAR? |
| 13:57.30 | fourcheeze | exten => 100,1,Dial(Local/101) |
| 13:57.30 | fourcheeze | exten => 101,1,Dial(Local/100) |
| 13:57.59 | fourcheeze | I thinkthat could easily grind my system to a halt |
| 13:58.06 | Zeeek | it's pretty ugly |
| 13:58.32 | fourcheeze | I suppose I could always Wait(1) before doing anything |
| 13:58.48 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:59.08 | *** join/#asterisk doughecka_ (n=Miranda@unaffiliated/doughecka) |
| 13:59.12 | Zeeek | there must be easier ways |
| 13:59.15 | fourcheeze | or is there something that will kill runaway things like that? |
| 13:59.28 | Zeeek | kill -p asterisk_pid |
| 13:59.33 | doughecka_ | raid |
| 13:59.35 | fourcheeze | if you can type |
| 13:59.44 | fourcheeze | I mean if your shell responds |
| 14:02.49 | fourcheeze | maybe something counts the number of applications per second that a thread opens or something |
| 14:03.06 | Zeeek | why re-write the shell in a dialplan? |
| 14:03.09 | RoyK | http://www.bbspot.com/News/2005/01/bush_countdown.html |
| 14:03.19 | fourcheeze | I'm not going to |
| 14:03.38 | Zeeek | big deal, his brother will take over |
| 14:03.39 | fourcheeze | ahh you mean system limits might do it |
| 14:04.26 | Zeeek | I mean there's likely a better way to do what you are so painfully trying to invent |
| 14:04.55 | kardecallan | Zeek, I have read that the solution for this problem would be the implementation of STUN. Is it correct? |
| 14:05.00 | *** join/#asterisk mellw (n=mell@146.17.227.87.j.siw.siwnet.net) |
| 14:05.07 | Zeeek | on the client end, you might try STUN, yes |
| 14:05.19 | Zeeek | what is the client phone? |
| 14:05.31 | *** part/#asterisk mellw (n=mell@146.17.227.87.j.siw.siwnet.net) |
| 14:05.50 | fourcheeze | it always amazes me when I look at a picture of Bush and see that this guy really is leader of some state |
| 14:05.56 | Zeeek | I know! |
| 14:05.58 | shiznatix | Is there a way to send a fax to a phone but to instead just bypass the phone itself and save the file to the asterisk server instead? |
| 14:06.06 | kardecallan | I'm using a softphone AdoreSoftPhone. |
| 14:06.15 | fourcheeze | and it's not the Congo or Nigeria or some other no-hope place |
| 14:06.33 | Zeeek | kardecallan I never heard of that, sorry |
| 14:06.44 | Zeeek | shiznatix a received fax? |
| 14:07.05 | glm2k | shiznatix: the fax sould be a tif file at one point. |
| 14:07.11 | glm2k | er, should |
| 14:07.19 | Zeeek | an incoming fax is a file anyway |
| 14:07.26 | glm2k | agreed |
| 14:07.38 | Zeeek | in /var/spool |
| 14:07.38 | astra^^ | how do i to route the calls from your server without any peer |
| 14:07.47 | astra^^ | *my server |
| 14:07.54 | kardecallan | May you suggest an specific phone to be used? |
| 14:08.14 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 14:08.17 | Zeeek | kardecallana no, but I know that Grandstream and X(Lite both have configsd for STUN |
| 14:08.26 | *** join/#asterisk ToTo (n=ToTo@81-174-33-2.f5.ngi.it) |
| 14:08.37 | Zeeek | I have used both WITHOUT STUN though by forwarding ports on the client side |
| 14:08.47 | *** join/#asterisk rigas (n=rigas@adsl-220-176-42.mob.bellsouth.net) |
| 14:08.50 | Zeeek | in fact I never use STUN |
| 14:09.39 | fourcheeze | scary |
| 14:11.01 | *** join/#asterisk zaf (n=tfournet@wsip-68-228-9-79.br.br.cox.net) |
| 14:11.47 | kardecallan | My client is be an ADSL modem. |
| 14:12.01 | kardecallan | Behind |
| 14:12.06 | noky | i need to set in extension.conf some dialplan to comunicate my Asterisk with a Gatekeeper |
| 14:12.08 | noky | how can i do this?? |
| 14:12.11 | noky | extend => ? |
| 14:12.14 | noky | in [default] ? |
| 14:12.23 | Zeeek | kardecallan what modem? |
| 14:12.37 | kardecallan | And I can't control it. |
| 14:13.03 | Zeeek | I guess STUN is the best idea for you then |
| 14:13.34 | astra^^ | how do i to route the calls from my * server without any peer? |
| 14:16.31 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 14:17.22 | __AK__ | anyone knows how i can test the pickup command before executing it |
| 14:17.24 | __AK__ | i'd like to test it, and if it return an error code do something else |
| 14:17.24 | chapeaurouge | what's the correct syntax for SIPAddHeader(Alert-Info: Internal) |
| 14:17.27 | chapeaurouge | ? |
| 14:17.33 | chapeaurouge | i currently have exten => _1XX,1,SIPAddHeader(Alert-Info: Internal) |
| 14:19.25 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 14:21.16 | shiznatix | About my fax, I am sending a .tif file from the asterisk server then trying to have asterisk ring a phone for about 5 seconds and if nobody picks up then i want it to save the fax to a new tif file on the server |
| 14:24.50 | *** join/#asterisk Eimann (i=eimann@linoa.etherkiller.de) |
| 14:24.51 | Eimann | hi |
| 14:25.34 | _Paulo_ | shiznatix, are you using a callfile? |
| 14:26.17 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 14:27.01 | Eimann | i've a asterisk and a digitalk node with ipt card and some E1 lines. i can set the clip feature with my telephone connected to a sipura box. when i'm calling via the digitalk, screening is disabled and my endpoint does not see my telephone number. digitalk says "we don't touch this flag". now they want some ngrep output or so between the * and the digitalk. but how can i see, if the flag is set? |
| 14:27.23 | _Paulo_ | shiznatix, I would advice you into using hylafax+iaxmodem to handle the fax jobs |
| 14:27.26 | fu3 | Well guys.. I'm running FreeBSD 6.1-PRERELEASE w/ Asterisk 1.2.5 and it works PERFECTLY. |
| 14:27.34 | fu3 | So.. Asterisk works on FreeBSD just fine ;) |
| 14:30.39 | shiznatix | _Paulo_: what is a callfile? I don't have a choice right now on any of my hardware |
| 14:31.14 | shiznatix | _Paulo_: also, the quality does not matter at all right now, all that matters is that I get at least a partial fax saved onto the asterisk server |
| 14:31.51 | _Paulo_ | are you sending the fax from or to * ? |
| 14:32.39 | shiznatix | both |
| 14:33.38 | shiznatix | sending it from asterisk by putting a callfile thing (i think thats a call file) into the outgoing folder, that file then calls the sip phone next to me. i dont want it to keep ringing on the sip phone, I want it to wait 5 seconds and if I dont answer to save it as a file on the asterisk server |
| 14:34.48 | _Paulo_ | you shoud use the "T" extension |
| 14:36.35 | _Paulo_ | shiznatix, set the timeout and then call another extension where you will call RxFAX(/path/${uniqueid}.tif) |
| 14:37.16 | *** join/#asterisk denisbr (n=c887e701@yossman.net) |
| 14:37.23 | *** join/#asterisk butzke (n=c887e701@yossman.net) |
| 14:37.32 | butzke | Hi |
| 14:37.34 | fu3 | hi |
| 14:37.41 | jsharp | lo |
| 14:37.46 | butzke | Anyone can tell me if Asterisck |
| 14:37.52 | jsharp | yes |
| 14:37.57 | shiznatix | when i try to call a extension that is no currently logged in it give me a error saying it could not call that extension |
| 14:38.05 | fu3 | ok |
| 14:38.14 | butzke | may be use to make calls to normal phones |
| 14:38.30 | butzke | ? |
| 14:38.32 | fu3 | yes it can |
| 14:38.39 | fu3 | with the right setup of course |
| 14:38.45 | butzke | yes |
| 14:39.01 | fu3 | dont goddamn message me |
| 14:39.10 | fu3 | (wow, I feel like im a part of this channel now!) |
| 14:39.49 | fu3 | hey jsharp.. just so you know, i got my shit all working perfectly now.. thanks for your help and advice |
| 14:40.03 | [TK]D-Fender | butzke : There is all sorts of equipment to let Asterisk use phone lines of every kind (POTS, BRI ISDN, T1, E1, etc) |
| 14:40.23 | fu3 | he is busy typing messages to me |
| 14:40.29 | fu3 | for some reason :) |
| 14:40.38 | vuud | good morning all. |
| 14:40.42 | fu3 | morning |
| 14:40.44 | vuud | I am having a problem connecting my pbx to gizmo... I have outbound calls working, and inbound calls half work... when they call in and I have an undefined context in sip.conf, they hear the voicemail lady and can leave a voicemail... |
| 14:40.51 | vuud | If I define the context in extensions, they hear nothing. On the CLI it says its doing everything it should be (playing sounds) but they hear nothing. |
| 14:41.04 | vuud | There is NAT, but why it would work without a valid context, but will not with one stumps me. Any help would be greatly appreciated |
| 14:41.08 | fu3 | hmm.. I wish I knew something about gizmo, but I do not |
| 14:41.37 | vuud | fu3, it gets configured just like any other SIP thing mostly |
| 14:41.54 | fu3 | oh |
| 14:42.35 | *** part/#asterisk denisbr (n=c887e701@yossman.net) |
| 14:42.38 | vuud | fu3: the voice client is pretty funny also... you can insert funny noises into your conversation... like "boooiiinnnngggg" |
| 14:42.56 | shiznatix | _Paulo_: when i try to call a extension that is no currently logged in it give me a error saying it could not call that extension (sorry for the double post) |
| 14:43.01 | *** join/#asterisk [Airwolf] (n=airwolf@attilla.nl) |
| 14:43.04 | fu3 | really?? fuck.. I had to hire the sound effects guy from Police Academy to jump on the other line to do that. |
| 14:43.26 | jsharp | I just fart into the mic. |
| 14:43.35 | vuud | fu3: sadly, you have wasted your money :) |
| 14:43.55 | _Paulo_ | shiznatix, you should make a "fax" extention containing the RxFax |
| 14:44.08 | fu3 | I dont know man.. he IS pretty good. |
| 14:44.48 | Hmmhesays | na na na na, don't phunk with my heart |
| 14:44.57 | jsharp | Michael Winslow. He's a riot at a live show. |
| 14:45.26 | vuud | So sound effects aside, could my problem be due to NAT? |
| 14:45.44 | shiznatix | _Paulo_: I have in the callfile first line (Channel: SIP/fax) then in the extensions.conf () |
| 14:45.45 | vuud | I can also call out to Gizmo fine |
| 14:45.51 | fu3 | brb |
| 14:45.56 | Hmmhesays | ahh the gizmo |
| 14:46.03 | *** join/#asterisk Skarmeth (n=Skarmeth@200164213103.user.veloxzone.com.br) |
| 14:46.28 | shiznatix | _Paulo_: I have in the callfile first line (Channel: SIP/fax) then in the extensions.conf (exten => fax,1,RxFAX(/tmp/ffax/tiffax.tif)) |
| 14:46.37 | _Paulo_ | shiznatix, and on the receiving * ? |
| 14:46.54 | Hmmhesays | vuud pastebin your dp |
| 14:47.12 | vuud | Hmmhesays: okay, wait one |
| 14:47.13 | shiznatix | _Paulo_: It's the same asterisk server. |
| 14:47.54 | Hmmhesays | little pig little pig let me in |
| 14:48.00 | Skarmeth | What's a recomended ATA (with T.30 or T.38 fax protocols) for using to connect Fax machines to Asterisk? |
| 14:48.08 | _Paulo_ | shiznatix, this fax extension is also your sip phone? |
| 14:48.43 | shiznatix | _Paulo_: No my SIP phone is set to 300 as the number |
| 14:49.51 | _Paulo_ | shiznatix, what isnt working, then? |
| 14:49.54 | jsharp | Skarmeth: t.38 on asterisk is still hit or miss. I wouldn't rely on it. |
| 14:50.42 | *** join/#asterisk file[laptop] (n=jcolp@142.131.190.116) |
| 14:50.56 | Ikarus | Anyone know of a cheap (sub 100 euro) VoIP speakerphone (as in, dedicated for speakerphone use in a conference room) |
| 14:50.59 | shiznatix | _Paulo_: when I get asterisk to start the fax it just dials the phone instead of saving the fax as a file even though I don't have any Answer() or Dial() in the extensions file |
| 14:51.57 | shiznatix | _Paulo_: Really I dont want it to ring the phone at all, I jsut want it to save the file automatically but if I don't have a connected phone it just says 'No such host: fax' or whatever |
| 14:52.47 | _Paulo_ | shiznatix, you can use the context instead of channel in the call file |
| 14:53.19 | _Paulo_ | shiznatix, put a rxfax context on your extensions.conf |
| 14:53.23 | *** join/#asterisk firejon (n=firejon@206-169-48-226.gen.twtelecom.net) |
| 14:53.38 | Skarmeth | jsharp, I know, but by what I read on forums and mailing lists, ATA's with T.30/T.38 support seens to make thinks more easy |
| 14:53.48 | *** join/#asterisk brodiem (i=1000@cpe-66-69-222-36.austin.res.rr.com) |
| 14:53.49 | Skarmeth | and I need to buy one now for a project |
| 14:54.21 | vuud | Hmmhesays: http://pastebin.com/603472 |
| 14:54.37 | shiznatix | _Paulo_: Alright that got me a little bit farther, now i get the error (apply_outgoing: At least one of app or extension must be specified) |
| 14:54.48 | vuud | Hmmhesays: wierd part is the CLI shows it doing everything it should |
| 14:54.53 | brodiem | can someone recommend a hard phone around or a bit more than the grandstream gxp2000? I wasn't completely satisfied with its build (huge footprint, plasticy) and call quality |
| 14:55.29 | Ikarus | ah well, I'll just toss another BudgeTone in the conference room, it worked good enough in tests |
| 14:55.48 | _Paulo_ | shiznatix, then put Extension: s |
| 14:55.52 | jsharp | Skarmeth: I've used Grandstream HT-286s with T.38, but not against Asterisk...only talking to my Quintum gateway. |
| 14:56.08 | *** join/#asterisk Micetto (n=k@217-133-98-121.b2b.tiscali.it) |
| 14:56.09 | _Paulo_ | shiznatix, Priority: 1 |
| 14:56.09 | jsharp | And they worked exceptionally well, even over crappy, bursty satellite links. |
| 14:56.12 | Micetto | hi :) |
| 14:56.34 | firejon | has anyone gotten presence hints to work in ael? |
| 14:56.34 | tzanger | that's because T38 removes a LOT of the timing issues with faxing |
| 14:56.42 | Micetto | anyone have tested mISDN and fax ? |
| 14:56.45 | fu3 | god I can hear some JUST HORRIBLE 1980's techno music playing |
| 14:56.47 | _Paulo_ | shiznatix, then exten => s,1,RxFax(.... |
| 14:56.47 | fu3 | ugh |
| 14:57.33 | Micetto | how to detect a fax signal with mISDN driver ? |
| 14:57.34 | shiznatix | _Paulo_: Same error |
| 14:57.47 | Micetto | I use mISDN v.3.1 |
| 14:58.09 | Micetto | please help me :(( |
| 14:58.18 | Hmmhesays | vuud, which is the default context |
| 14:59.03 | _Paulo_ | shiznatix, pb your callfile |
| 14:59.08 | vuud | well, default is in general, but when they come in from gizmo its the "from-gizmo" |
| 14:59.11 | Skarmeth | jsharp, thank's for the tip about HT-286 |
| 14:59.30 | shiznatix | _Paulo_: pb? |
| 14:59.43 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 14:59.53 | Hmmhesays | vuud so if dial fails you want to playback moo2? |
| 15:00.02 | noky | how is the character * in extension.conf for a dial plan ? |
| 15:00.31 | _Paulo_ | ~pb |
| 15:00.33 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 15:00.48 | Micetto | ... |
| 15:01.21 | vuud | Hmmhesays: thats just one of the ways... If they call here from gizmo, it picks up - the CLI says its playing moo or whatever else I tell it too - but they here nothing. If I roll them right to voicemail I get a CLI error about "no audio from SIP". But if I let them go into the default context, they get and hear voicemail... which is odd since they should not get that I dont think |
| 15:02.15 | jsharp | Skarmeth: Gotta make sure the Grandstream is running latest & greatest firmware. T.38 support has been very recent. |
| 15:02.30 | shiznatix | _Paulo_: http://pastebin.com/603482 |
| 15:02.46 | *** join/#asterisk Teeli (n=Tili@82-217-236-131.cable.quicknet.nl) |
| 15:02.50 | Hmmhesays | vuud answer the call first |
| 15:03.01 | Skarmeth | ok, I was looking on Sipura's, but it has a "Pending" on it's data sheet... does news fw has T.38 ready? |
| 15:03.21 | Skarmeth | s/news/newer |
| 15:03.39 | Micetto | .... |
| 15:03.43 | Micetto | :( |
| 15:04.03 | Skarmeth | I need to go out for a while |
| 15:04.19 | Micetto | help...me...(cry) :'( |
| 15:07.18 | vuud | Hmmhesays: crap. I swear it was not working on ones that we had that with. Let me see if that works though |
| 15:07.48 | vuud | Hmmhesays: I was trying crap all last night - I'm gonna whack myself with a keyboard if that fixes it |
| 15:08.18 | Hmmhesays | make sure you have canreinvite set as no in your general sip conf too |
| 15:09.23 | _Paulo_ | shiznatix, use Channel: local/s@term_fax |
| 15:09.24 | vuud | Hmmhesays: I have that, and I also added externip also recently... my guy with Gizmo seems to have disappeared too. Thanks much! |
| 15:09.43 | Hmmhesays | if you have externip you should have your localnet set also |
| 15:09.51 | _Paulo_ | shiznatix, instead of SIP/fax |
| 15:10.04 | vuud | Hmmhesays: thanks! |
| 15:10.19 | *** join/#asterisk redondos (n=redondos@190.48.36.29) |
| 15:10.19 | brodiem | has anyone used the Polycom IP301? any optinions? |
| 15:10.22 | brodiem | er opinions |
| 15:10.25 | shiznatix | _Paulo_: should i use Channel: or Context: ??? |
| 15:10.41 | redondos | What ports should I forward in my firewall to my asterisk server if I want to connect with a softphone remotely using SIP? Just 5060 TCP/UDP? |
| 15:11.06 | _Paulo_ | shiznatix, Channel |
| 15:11.14 | vuud | redondos: not sure if its accurate, but I've been reading http://www.voip-info.org/wiki-Asterisk+firewall+rules |
| 15:11.23 | RoyK | redondos: just configure the softphone to register with asterrisk. that should do by itself |
| 15:11.25 | redondos | vuud: Thank you. |
| 15:11.58 | [TK]D-Fender | redondos : softphone behind NAT contacting a public IP *? |
| 15:12.18 | _Paulo_ | shiznatix, seems you always have to use Channel, but you can choose between Application or Context. |
| 15:12.48 | _Paulo_ | shiznatix, sorry for the wrong tip before. |
| 15:13.23 | redondos | [TK]D-Fender: Yeah. |
| 15:13.39 | shiznatix | _Paulo_: don't worry about it. Ok this is working as far as no errors but the new file (/tmp/ffax/tifftmp.tif) is not created but i do not get any errors |
| 15:13.45 | redondos | [TK]D-Fender: What ports does the softphone will need forwarded, then? |
| 15:13.50 | file[laptop] | Micetto: ...rrrrrright |
| 15:14.04 | MikeJ[Laptop] | file[laptop], having fun? |
| 15:14.10 | file[laptop] | always! |
| 15:14.15 | MikeJ[Laptop] | your missed |
| 15:14.28 | file[laptop] | why am I missed? :( |
| 15:14.28 | redondos | s/will// |
| 15:14.51 | redondos | Nice script. I love it. Please tell me what's its name. |
| 15:15.04 | shiznatix | _Paulo_: the last thing asterisk outputs is (> Launching txfax(/tmp/tiff.tif|caller) on Local/s@term_fax-cccc,1) |
| 15:16.03 | _Paulo_ | shiznatix, hum... that -cccc is strange... |
| 15:17.06 | Micetto | file[laptop]: what ?!?!? |
| 15:17.43 | Micetto | I have a problem and file[laptop] have fun....file is not good...:'( |
| 15:17.56 | Micetto | ops...sorry...life is not good |
| 15:18.29 | file[laptop] | Micetto: well, it's just... you have a problem... but you haven't said what the problem is |
| 15:19.17 | MikeJ[Laptop] | file[laptop], maybe he doesn't really have a problem? |
| 15:19.30 | file[laptop] | MikeJ[Laptop]: maybe! |
| 15:19.43 | MikeJ[Laptop] | so don't worry about it.. |
| 15:19.50 | *** join/#asterisk mkl1525 (n=daniel@pD9533837.dip0.t-ipconnect.de) |
| 15:20.58 | Micetto | file[laptop]: |
| 15:21.07 | Micetto | file[laptop]: my problem is fax over mISDN driver |
| 15:21.08 | shiznatix | _Paulo_: http://pastebin.com/603504 that is the exact output when i run the callfile |
| 15:21.26 | file[laptop] | Micetto: then say exactly what your issue is, and maybe someone will respond |
| 15:21.34 | file[laptop] | but I know of nobody off the top of my head who uses mISDN - so good luck |
| 15:21.54 | Becky75 | mISDN r u guys nuts who will use that?.. |
| 15:22.22 | MikeJ[Laptop] | just passinng fax over an mISDN to mISDN calll.? |
| 15:22.24 | *** join/#asterisk linville (n=linville@azure.tuxdriver.com) |
| 15:22.30 | MikeJ[Laptop] | or misdn to rxfax, or what? |
| 15:22.51 | Micetto | misdn to rxfax |
| 15:22.59 | mkl1525 | Hi, didn't find it in http://www.voip-info.org/tiki-index.php?page=Asterisk+config+zapata.conf so could anybody clarify what the difference between bri_net, bri_net_ptmp and bri_net_ptp is? |
| 15:23.23 | Micetto | file[laptop]: what kind of isdn you use with Asterisk 1.2.5 ? |
| 15:23.30 | fugitivo | ~seen coppice |
| 15:23.33 | jbot | coppice <n=chatzill@211.155.168.153> was last seen on IRC in channel #asterisk, 1d 5h 22m 31s ago, saying: '"Nun"'. |
| 15:23.42 | file[laptop] | I don't use ISDN |
| 15:23.48 | Micetto | :( |
| 15:24.46 | backblue | mkl1525: ptp does not exists. bri_net bri_net_ptmp bri_cpe bri_cpe_ptmp |
| 15:24.55 | Micetto | Can I use bristuff-0.3.0-PRE-1l without problem ? |
| 15:25.00 | zigman | no |
| 15:25.02 | zigman | use m |
| 15:25.04 | MikeJ[Laptop] | Micetto, rxfax is very time sensitive |
| 15:25.06 | backblue | Micetto: use k |
| 15:25.14 | Micetto | ah ok |
| 15:25.18 | MikeJ[Laptop] | if you have any timing issues in the driver, it just wont work |
| 15:25.21 | backblue | l have some patch's in iax implementations |
| 15:25.24 | backblue | k it's fine |
| 15:25.30 | Micetto | and with qozap can I normaly receive fax? |
| 15:25.34 | *** join/#asterisk Winkie (n=urmom@cpc2-stre1-6-0-cust119.bagu.cable.ntl.com) |
| 15:25.38 | backblue | dunno |
| 15:25.59 | Winkie | any of you use CDR with incoming calls transferred back out? |
| 15:26.02 | MikeJ[Laptop] | qozap? |
| 15:26.05 | zigman | use rc2o for those |
| 15:26.10 | zigman | quadbri drivers |
| 15:26.11 | backblue | MikeJ[Laptop]: quad bri driver |
| 15:26.20 | zigman | use only that module |
| 15:26.27 | MikeJ[Laptop] | dunno |
| 15:26.38 | MikeJ[Laptop] | I don't know details of mISDN |
| 15:26.43 | zigman | but bbristuff-0.3.0-PRE-1k drivers |
| 15:26.52 | MikeJ[Laptop] | I just know that anything with faxing is very time sensitive |
| 15:26.58 | backblue | it's not misdn, its bristuff |
| 15:27.05 | MikeJ[Laptop] | yes, I know |
| 15:27.17 | MikeJ[Laptop] | but it was mISDN a minute ago |
| 15:27.22 | backblue | yes |
| 15:27.45 | Micetto | now I'm going to install bristuff-0.3.0-PRE-1k.tar.gz package |
| 15:28.04 | Micetto | with qozap driver form my quadBRI card |
| 15:28.18 | MikeJ[Laptop] | ok |
| 15:28.25 | Micetto | and I hope that fax works fine! |
| 15:28.42 | mkl1525 | backblue, thanks, so bri_net and bri_cpe is for point-to-point? |
| 15:28.48 | backblue | mkl1525: y |
| 15:29.01 | backblue | i think so |
| 15:29.13 | backblue | never used ptp |
| 15:29.20 | noky | w |
| 15:29.21 | mkl1525 | backblue, will try it thanks for the help |
| 15:29.26 | backblue | mkl1525: np |
| 15:29.58 | *** join/#asterisk umay (n=chris@70-101-61-50.dsl2-plymouth.roc.ny.frontiernet.net) |
| 15:30.42 | kmilitzer | Hello everyone. |
| 15:31.05 | kmilitzer | Is there any way to terminate t.38 on asterisk? |
| 15:31.45 | jsharp | Yes, but its hit or miss on whether or not it will actually work. Its not production ready yet. |
| 15:32.19 | mutilator | is it hit and miss per connection |
| 15:32.30 | mutilator | or does it tend to hit and miss per person |
| 15:32.35 | jsharp | Per device. |
| 15:32.39 | mutilator | user a will work 90% of the time |
| 15:32.47 | mutilator | user b works 50 |
| 15:32.47 | kmilitzer | jsharp: I don't care ... I want to test it |
| 15:33.08 | jsharp | Oh. Well, then "yes" is the answer to your question. |
| 15:33.27 | kmilitzer | I need it ... I need to build a ss7-test-gateway and this must have t.38 in some way |
| 15:33.40 | kmilitzer | So it's woth testing befor I give up with empty hands ... |
| 15:33.53 | kmilitzer | jsharp: where can I find more info? Do I need svn? |
| 15:35.54 | jsharp | I tested it wish Asterisk 1.2.4, the T.38 patch from bugs.digium.com. You'll also need app_txfax & rxfax, plus the DSP tools. |
| 15:38.43 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 15:39.20 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 15:39.59 | shiznatix | _Paulo_: are you still there? |
| 15:40.36 | Hmmhesays | well lcdial is giving me hell this morning |
| 15:40.37 | Hmmhesays | weeeee |
| 15:41.08 | fu3 | Hmm |
| 15:41.34 | kmilitzer | jsharp: do you mean this patch: http://bugs.digium.com/view.php?id=5090 |
| 15:41.47 | jsharp | That's the one. |
| 15:42.35 | shiznatix | I have a problem when trying to send a fax. Basically it says that it completed it but in reality no file was created. Here is a pastebin of my information http://pastebin.com/603534 |
| 15:43.22 | _Paulo_ | shiznatix, yes, I'm here. |
| 15:44.44 | _Paulo_ | shiznatix, do you run your * server as root? |
| 15:45.00 | *** join/#asterisk SpaceBass (n=sp@static-71-251-230-2.rcmdva.fios.verizon.net) |
| 15:45.01 | SpaceBass | morning |
| 15:45.24 | SpaceBass | I've finally had it with broadvoice...been waiting on them to port my number since december....anyone have a provider they recomend? |
| 15:45.25 | shiznatix | _Paulo_: I don't know but I made the folder /tmp/ffax/ and chmod it to 777 |
| 15:45.40 | shiznatix | _Paulo_: So it should be able to write to that folde3r |
| 15:46.07 | *** join/#asterisk gambolputty (n=root@64.74.225.131) |
| 15:46.35 | Hmmhesays | it works now |
| 15:46.36 | Hmmhesays | sweet |
| 15:46.52 | Hmmhesays | SpaceBass asterlink? |
| 15:46.53 | Octothorpe | ~sixtel |
| 15:46.54 | jbot | i guess sixtel is "a SIP / IAX origination and termination service for US48. DIDs cost like $1.50 per month with like 1.6 cents per minute incoming. They also provide toll-free USA DIDs for like 20 cents a months at 2 cents a minute. |
| 15:47.02 | Hmmhesays | i've had decent luck with sixtel too |
| 15:47.13 | asteriskmonkey | tja, tjats ex[emsoce |
| 15:47.22 | asteriskmonkey | thats expensive |
| 15:47.24 | _Paulo_ | shiznatix, in my setup asterisk:asterisk is the dir owner |
| 15:47.41 | asteriskmonkey | .0018 cents per min is going us48 rate |
| 15:47.54 | SpaceBass | Hmmhesays, thanks, I'll check it out |
| 15:48.04 | shiznatix | _Paulo_: I get 'Invalid User' when trying to chown ffax to asterisk:asterisk |
| 15:48.12 | SpaceBass | as long as they have good rates and can port my number....oh and reliability |
| 15:48.21 | _Paulo_ | shiznatix, I use Debian |
| 15:48.39 | shiznatix | _Paulo_: Same here |
| 15:48.58 | jsharp | try asterisk.asterisk |
| 15:49.01 | jsharp | not asterisk:asterisk |
| 15:49.11 | Hmmhesays | anyone else in here using lcdial? |
| 15:49.26 | shiznatix | jsharp: same error |
| 15:49.37 | iDunno | try |
| 15:49.41 | iDunno | chown asterisk file |
| 15:49.43 | iDunno | then |
| 15:49.47 | iDunno | chgrp asterisk file |
| 15:50.15 | austinnichols101 | SpaceBass: telasip.com |
| 15:51.17 | shiznatix | iDunno: I am trying to create a brand new file, should I actally be trying to overwrite the existing file? |
| 15:51.19 | SpaceBass | so far broadvoice had some of the best rates, but the fact that they will not port my number is killing me |
| 15:51.55 | _Paulo_ | shiznatix, no, you do not. |
| 15:52.07 | *** join/#asterisk Eitch (n=hugo@unaffiliated/eitch) |
| 15:52.10 | tzafrir | group asterisk? maybe group dialout? |
| 15:52.34 | *** join/#asterisk AlexCTI (n=alex@pembrkfl-bellsouth-24-53-200-134.miamfl.adelphia.net) |
| 15:53.00 | _Paulo_ | shiznatix, In my setup * creates the file. I put then into /var/spool/asterisk-fax/${exten}/${uniqueid}.tif |
| 15:54.11 | _Paulo_ | shiznatix, but with 1.2.4, txfax doesnt work for me. |
| 15:54.33 | _Paulo_ | shiznatix, I have to use iaxmodem+hylafax. |
| 15:54.44 | wasim | ugh |
| 15:55.16 | vuud | Hmmhesays: No work. Plus a bad grep left my last post incomplete... Here is the chunk of the extensions I have http://pastebin.com/603574 along with the CLI output from the call. Thanks for any help! |
| 15:55.43 | shiznatix | _Paulo_: Crap, I am running 1.2.4, what is this iaxmodem+hylafax?? |
| 15:55.49 | vuud | Hmmhesays: Also, I added in the timeout after the cli copy and paste |
| 15:56.01 | *** join/#asterisk bweschke (n=bweschke@sjcc28x184.sjccnet.com) |
| 15:56.17 | _Paulo_ | shiznatix, hylafax is a very good oss fax server |
| 15:56.19 | *** join/#asterisk wunderkin (n=kev@ip68-226-113-228.ph.ph.cox.net) |
| 15:56.38 | Hmmhesays | that is pretty odd vuud |
| 15:57.19 | vuud | Hmmhesays: Yeah. I'm stumped |
| 15:57.38 | shiznatix | _Paulo_: A iaxmodem is a specific type of hardware modem? |
| 15:58.03 | _Paulo_ | shiznatix, iaxmodem is a pure software modem that uses Steve Underwood libspandsp (use the last version) |
| 15:58.12 | vuud | Hmmhesays: I am wondering about it being the firewall... not sure why it would work with the default (which does not really run the default I guess) but not once I add a context |
| 15:58.20 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 15:58.20 | *** mode/#asterisk [+o anthm] by ChanServ |
| 15:58.55 | shiznatix | _Paulo_: Ok I shall take a look at these things. Do they have to be installed on the asterisk server or can I use them from my machine? |
| 15:59.01 | *** join/#asterisk SplasPood (n=jwb@206.252.198.100) |
| 15:59.49 | SpaceBass | so broadvoice tells me that my port has been approved but they are waiting on a date from my carrier |
| 15:59.51 | SpaceBass | is that bs or real? |
| 16:00.15 | _Paulo_ | I had the same txfax problems with the iaxmodem that cames with debian. |
| 16:00.21 | AlexCTI | Hi. I have questions about codec g279 and g711, actually I don't have any license and all the calls are using g711, and my band with is high, and i'm using x-lite softphones, and I have PRI lines attach to my server, so in order migrate a g729 do I need purchase g729 licenses on the server and that's it? |
| 16:00.27 | _Paulo_ | you can run iaxmodem from any machine |
| 16:01.32 | _Paulo_ | shiznatix, hylafax has a nice virtual printer port monitor called winprint, that let you create a "fax" printer and fax from any windows machine. |
| 16:01.32 | SwK[Work] | actually iaxmodem comes with the correct versions of libIAX and SpanDSP |
| 16:02.24 | shiznatix | _Paulo_: That would be good if I had a computer with windows but sadly we are all Ubuntu and Debian here. |
| 16:02.33 | *** join/#asterisk rajiv|work (n=rajiv@gentoo/developer/rajiv) |
| 16:02.48 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 16:03.01 | _Paulo_ | shiznatix, bether iet, hylafax works very well with linux clients. |
| 16:03.22 | AlexCTI | Anyone can explain me how the license works? |
| 16:03.38 | SpaceBass | Hmmhesays, you put freepbx over your a@h install, right? |
| 16:03.47 | shiznatix | _Paulo_: excellent. Alright Im going for a smoke then to figure this hylafax and iaxmodem out, Thank you very much for your help and if I need more help I hope you are back in the channel. Thanks! |
| 16:04.03 | Hmmhesays | spacebass I haven't yet, but others have been successful at it |
| 16:04.14 | SpaceBass | wonder if it overwrites all the .conf files |
| 16:06.41 | Hmmhesays | god I hate excel |
| 16:08.59 | *** join/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 16:09.19 | SpaceBass | ohhhhh |
| 16:10.12 | fu3 | I wish to configure Asterisk's dialplan to do different things at different times of the day, can anyone point me in the right direction? |
| 16:10.31 | fu3 | nevermind |
| 16:10.31 | jsharp | time based includes |
| 16:10.33 | fu3 | foudnb it |
| 16:10.43 | fu3 | http://www.voip-info.org/wiki-Asterisk+tips+openhours |
| 16:10.46 | fu3 | that seems to be it |
| 16:10.51 | *** join/#asterisk ^HeLL^ (n=admin@232.Red-83-42-51.dynamicIP.rima-tde.net) |
| 16:10.56 | ^HeLL^ | hello all |
| 16:10.58 | fu3 | hi |
| 16:12.10 | Hmmhesays | Spacebass any good way I can go through and take all the numbers in a column that start with 011 and remove it? |
| 16:12.25 | *** join/#asterisk _maydayjay_ (n=maydayja@gimel.nas.net) |
| 16:12.41 | ^HeLL^ | I just put a web titled AsteriskCounter to keep contact and count how many asterisk users there are on the world... |
| 16:12.49 | *** join/#asterisk shido6 (n=shido6@d221-68-216.commercial.cgocable.net) |
| 16:12.51 | *** join/#asterisk fulgas (n=fulgas@82.102.2.199) |
| 16:13.44 | *** part/#asterisk Eimann (i=eimann@linoa.etherkiller.de) |
| 16:17.07 | [TK]D-Fender | Hmmhesays : Just sort by the colum and mass select them in a row... |
| 16:18.21 | Hmmhesays | i don't want to remove them completely |
| 16:18.25 | Hmmhesays | just the 011 part |
| 16:18.35 | Hmmhesays | voipjets rate tables are retarded |
| 16:18.48 | *** join/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 16:19.04 | astra^^ | fu.. fu .. fu rescue.. team... :( |
| 16:19.32 | astra^^ | can i place call tru* without a dial peer |
| 16:19.46 | Hmmhesays | yes |
| 16:20.09 | astra^^ | how do i do tat |
| 16:20.20 | Hmmhesays | with the dial command |
| 16:20.20 | astra^^ | rgt nw i have 1001 as peer |
| 16:21.15 | vader-- | have any of you guys setup asterisk where it requires user codes to make long distance calls? |
| 16:21.40 | astra^^ | as like.. |
| 16:21.54 | astra^^ | => _127966XXXXXXXXX,2,Dial(SIP/${EXTEN:3}@mypbx) |
| 16:21.58 | jsharp | App_authenticate |
| 16:22.32 | brettnem | of course you can't determine if a call is LD my only looking at the Area Code.. |
| 16:22.54 | vader-- | well we can say any call that isn't a specific area code |
| 16:22.56 | vader-- | needs a number first |
| 16:22.57 | vader-- | right |
| 16:23.19 | *** join/#asterisk DeeJay[2] (n=bleh@office.abi.ca) |
| 16:23.25 | astra^^ | without a dial peer? |
| 16:23.34 | vader-- | dial peer? |
| 16:23.39 | astra^^ | yes ... |
| 16:23.54 | astra^^ | i mean can i place call without the dial peer |
| 16:23.56 | DeeJay[2] | We often see the zap driver disabling the echo canceller when it shouldn't do it... is it a problem with our telco (PRI) or with our channel banks? We're using a Digium Quad T1 card. |
| 16:24.06 | a1fa | HELO |
| 16:24.13 | a1fa | :P |
| 16:24.14 | jsharp | Pleased to meet you. |
| 16:24.14 | a1fa | EHLO |
| 16:24.20 | [TK]D-Fender | Hmmhesays : Sort by it and do =mid(a1,4,255) |
| 16:24.22 | a1fa | MAIL FROM: <> |
| 16:24.27 | a1fa | 250 ok? |
| 16:24.32 | exonic | DeeJay[2], Sure they're not fax calls? |
| 16:24.33 | jsharp | 500 GO POUND SAND |
| 16:24.36 | DeeJay[2] | exonic: sure |
| 16:24.36 | astra^^ | vader--? |
| 16:24.47 | a1fa | RCPT TO: <*@*> |
| 16:25.00 | a1fa | 200 SLAP |
| 16:25.03 | exonic | a1fa, do you need help? |
| 16:25.06 | a1fa | :P |
| 16:25.13 | a1fa | exonic : no, thanks.. |
| 16:25.23 | a1fa | i am just saying hi to my friends |
| 16:25.28 | a1fa | [TK]D-Fender : respek' |
| 16:25.37 | astra^^ | exonic: i need help |
| 16:25.38 | a1fa | biggy up from down & under |
| 16:25.44 | [TK]D-Fender | a1fa : y0 |
| 16:25.45 | a1fa | astra^^ : just ask |
| 16:25.54 | astra^^ | i mean can i place call without the dial peer |
| 16:25.55 | a1fa | [TK]D-Fender : i hate/love nat |
| 16:25.57 | astra^^ | and hw |
| 16:26.04 | exonic | DeeJay[2], i've never seen it so i'm sure i'd be of no help. Sorry. |
| 16:26.09 | [TK]D-Fender | a1fa : Doesn't bother me anywhere.... |
| 16:26.11 | DeeJay[2] | exonic ok thank you |
| 16:26.12 | exonic | DeeJay[2], I am usign the exact same card as well |
| 16:26.28 | a1fa | [TK]D-Fender : that netgear router kept making my remote sip unreachable |
| 16:26.31 | brodiem | has anyone used polycom IP301's? |
| 16:26.54 | a1fa | [TK]D-Fender : the work around, I had to enable port forwarding.. nat=yes;qualify=2000;canreinvite=no; |
| 16:27.02 | [TK]D-Fender | brodiem : plenty of people, whats your question? |
| 16:27.21 | a1fa | brodiem: has anyone used a butt plug |
| 16:27.23 | [TK]D-Fender | a1fa : thats NORMAL.... |
| 16:27.29 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 16:27.46 | a1fa | [TK]D-Fender : :P not really.. my other phones (behind a linksys) didnt need port forwarding |
| 16:28.05 | brodiem | [TK]D-Fender, just looking for a couple opinions (looks ok so far from reviews I've read), and I wanted to know if anyone has tested them using headsets without using the amplifier |
| 16:28.07 | *** join/#asterisk Fedoracore6 (n=deddd@60.50.132.131) |
| 16:28.08 | [TK]D-Fender | a1fa : Actually yeah.. you shouldn't need to do taht... bad NAT I guess... |
| 16:28.19 | astra^^ | i still dint gt help |
| 16:28.22 | AlexCTI | Hi Everyone, Someone can explain me how the g729 works, I mean if I purchase some lic for the server, Do I need purchase lic for my x-ten too? |
| 16:28.43 | [TK]D-Fender | brodiem : It works (having tried with a Plantronics "striaght" cable, but I would do that in a call-center. Amplifiers make the difference... |
| 16:28.51 | tzanger | AlexCTI: x-ten has the license already (you paid for it) |
| 16:28.57 | [TK]D-Fender | brodiem : For home / light use, its "ok" I guess |
| 16:29.10 | brodiem | [TK]D-Fender, are there any phones in that price range that have an amplified headset port? |
| 16:29.14 | AlexCTI | Sorry, I'm using x-lite |
| 16:29.17 | [TK]D-Fender | AlexCTI : Which X-Ten? |
| 16:29.32 | tzanger | x-lite does not have g729, you buy x-ten to get g729 |
| 16:29.38 | [TK]D-Fender | X-Lite doesn't come with G729 at all. Only their PAYED versions (X-Pro / eyeBeam)_ |
| 16:29.46 | brodiem | [TK]D-Fender, the headset seemed to be ok on a GXP2000 but it isn't amplified either is it? |
| 16:29.58 | [TK]D-Fender | brodiem : What kind of use? |
| 16:30.20 | brodiem | [TK]D-Fender, it willbe in a call center environment |
| 16:30.34 | AlexCTI | ok, so If I purchase x-ten with g729, do I need buy g279 on the server? |
| 16:30.41 | [TK]D-Fender | brodiem : IP301 + AMP. I mean it.. be nice to your agents..... |
| 16:30.48 | brodiem | lol |
| 16:30.55 | a1fa | AlexCTI : you can just download xten.. its free |
| 16:31.00 | a1fa | X-Lite |
| 16:31.02 | austinnichols101 | what's up a1fa? |
| 16:31.04 | fu3 | hey, will Playback play .wav or .raw files? |
| 16:31.06 | fu3 | or just .gsm? |
| 16:31.09 | a1fa | austinnichols101 : chillen dude... |
| 16:31.10 | *** join/#asterisk saftsack (n=oliver@p54A7DAD3.dip.t-dialin.net) |
| 16:31.16 | austinnichols101 | are you nat happy? |
| 16:31.19 | brodiem | [TK]D-Fender, we have headsets already that are amplified, but the manager doesn't like the sloppiness of all the cables... that's why I asked |
| 16:31.22 | a1fa | austinnichols101 : 156ms delay, but it still sounds awesome |
| 16:31.28 | austinnichols101 | suh-weet |
| 16:31.32 | [TK]D-Fender | brodiem : I ended up having to get my guys H261 binaural headseds, M12 amps for their IP601's |
| 16:31.35 | a1fa | austinnichols101 : its ok.. i dont like port forwarding |
| 16:31.42 | *** join/#asterisk salviadud (n=ralfalfa@201.137.161.31) |
| 16:31.42 | a1fa | austinnichols101 : script kiddies can call you on 5060 |
| 16:31.45 | austinnichols101 | right |
| 16:31.49 | [TK]D-Fender | brodiem : They clarity will suffer without them... |
| 16:31.56 | AlexCTI | a1fa, if I download x-ten, i just need buy lic on the server right? |
| 16:32.01 | a1fa | i updated the firmware.. still nothing |
| 16:32.09 | a1fa | AlexCTI : yes |
| 16:32.14 | brodiem | [TK]D-Fender, are you referring to the polycom phones only, or are there others with an amplified port? |
| 16:32.17 | [TK]D-Fender | a1fa : port forwarding doesn't do that.... accepting un-auth'd calls does :) |
| 16:32.30 | a1fa | you will need license on both ends in order to utilize the protocol |
| 16:32.34 | [TK]D-Fender | brodiem : No phone worthy mentioning even-if. |
| 16:32.47 | a1fa | [TK]D-Fender : hahaha.. all those SIP ATAs can recieve calls.. no way to disable that |
| 16:32.57 | a1fa | IP calling.. damn it |
| 16:33.01 | [TK]D-Fender | My SPA-941 seemed "ok", but takes a 2.5mm jack. |
| 16:33.01 | a1fa | powerchip:5060, ring ring! |
| 16:33.14 | austinnichols101 | aifa: here's another couple of solutions. 1. Get a phone with keepalive. 2. Make your own keepalive by setting the register expiration to something low like 15 seconds (the phone will re-register every 15 seconds). Then you can turn off qualify=yes on the server side |
| 16:33.35 | austinnichols101 | still kinda ugly, IMO |
| 16:33.37 | backblue | anyone -> http://lists.digium.com/pipermail/asterisk-dev/2006-March/019309.html ? |
| 16:33.37 | a1fa | too much chatter :P |
| 16:33.38 | AlexCTI | I got it.. can you send me the x-ten link? |
| 16:33.44 | a1fa | www.x-ten.com |
| 16:34.09 | AlexCTI | thnks |
| 16:34.43 | Abydos313 | why not .net client :)) |
| 16:34.53 | a1fa | why not .com client, biznutch! |
| 16:35.01 | gaupe | why not XUL? |
| 16:35.05 | a1fa | i want to learn ajax |
| 16:35.08 | a1fa | looks neat |
| 16:35.16 | Abydos313 | lots of talk lately about that a1fa.. haven't heard anything on .net |
| 16:35.22 | gaupe | ajax is just a hype, nothing much to it |
| 16:35.39 | a1fa | gaupe : well, it is the future |
| 16:35.44 | a1fa | think of it that way |
| 16:35.48 | austinnichols101 | web 2.0 baby! |
| 16:35.52 | a1fa | all office applications will be outsourced on the internet |
| 16:35.59 | a1fa | so no more pirateZ |
| 16:36.03 | brodiem | [TK]D-Fender thanks for the info |
| 16:36.06 | gaupe | yeah right, it's one or three javascript functions that has been there for ages |
| 16:36.07 | austinnichols101 | a1fa: use flex |
| 16:36.08 | a1fa | you will need to buy a login in order to use office |
| 16:36.28 | a1fa | twisted maybe |
| 16:36.31 | brodiem | [TK]D-Fender you think an IP301 would serve best for that price range? I got a GXP2000 in to try it and wasn't overly impressed, and I need about 20 more of them, so I wanted an alternative |
| 16:36.31 | a1fa | http://twistedmatrix.com/trac/ |
| 16:36.35 | austinnichols101 | a1fa: we already offer that (hosted ms office) |
| 16:36.53 | austinnichols101 | you can 'rent' ms project for a week |
| 16:37.13 | a1fa | austinnichols101 : you work for ms? |
| 16:37.22 | *** join/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 16:37.39 | a1fa | or macro? |
| 16:37.44 | austinnichols101 | a1fa: nope - mbs partner for Dynamics AX (Axapta) |
| 16:38.09 | austinnichols101 | but then we do hosting so under their Service Provider License Agreement (SPLA) you can basically 'rent' any of their software |
| 16:38.31 | austinnichols101 | and a McAfee Elite partner |
| 16:38.35 | queuetue | Hello. Could anyone point me towards a resource where I can learn more about T1 lines? How they are used, how flexble they are, just a trusted general resource so I understand what I'm researching. |
| 16:38.59 | Fedoracore6 | hai all |
| 16:39.04 | Fedoracore6 | can i use CDR using PHPAGI |
| 16:39.16 | Fedoracore6 | its ok to my system |
| 16:39.40 | [TK]D-Fender | brodiem : IP301 is a perfect choice for a call center. low-budget, high wuality |
| 16:39.58 | Fedoracore6 | cos i try do the update code using agi but not success |
| 16:40.02 | brodiem | great |
| 16:40.39 | a1fa | austinnichols101: flex any god? |
| 16:40.40 | a1fa | good? |
| 16:40.40 | Fedoracore6 | its ok or nor using CDR using PHPAGI |
| 16:41.16 | austinnichols101 | a1fa: I think it's a great product. Think of it as being able to describe a UI using XML + Actionscript and then have flex generate the flash results |
| 16:41.21 | AlexCTI | a1fa: That link doesn't let download anything |
| 16:41.35 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 16:41.37 | a1fa | AlexCTI : yes it does |
| 16:41.51 | a1fa | google.com |
| 16:41.52 | a1fa | search for |
| 16:41.54 | a1fa | X-PRO_Install.exe |
| 16:41.55 | austinnichols101 | aifa: then they have lots of screen widgets (grids, drop-down boxes, radio buttons, tabs, etc) |
| 16:42.00 | a1fa | site:.pk |
| 16:42.07 | a1fa | search "X-PRO_Install.exe site:.pk" |
| 16:42.20 | a1fa | austinnichols101 : i will think about making a sip client :p |
| 16:42.29 | zoa | a1fa: please dont spread out links to warez on this channel |
| 16:42.54 | a1fa | its not |
| 16:43.04 | a1fa | damn it dude |
| 16:43.08 | zoa | l :) |
| 16:43.09 | a1fa | stop pointing fingers |
| 16:43.17 | zoa | it looks kinda suspicious :p |
| 16:43.34 | a1fa | he still needs to pay for the license key |
| 16:44.07 | Skarmeth | brodiem, I got 20 IP 301 las friday |
| 16:44.16 | a1fa | Skarmeth : damn dude |
| 16:44.19 | a1fa | Skid: how much? |
| 16:44.58 | a1fa | Skarmeth : how much>? |
| 16:45.00 | Skarmeth | it was in Brazil, R$ 550,00 (about $ 253,00 with taxes) |
| 16:45.28 | jarrod | I'm receiving quite a few 'Avoiding initial deadlock' on different channel types... is this normal? |
| 16:45.29 | a1fa | thats it? |
| 16:45.32 | a1fa | $253 USD? |
| 16:45.40 | a1fa | per phone? |
| 16:45.44 | a1fa | or for 20 of them? |
| 16:46.02 | Skarmeth | it was a good business, your market average price is R$ 611,00 (about $ 281.00) |
| 16:46.16 | brodiem | Skarmeth, how are they fairing out? |
| 16:46.17 | a1fa | IP301 is $134 USD |
| 16:46.21 | a1fa | per phone :P |
| 16:46.23 | a1fa | you got robbed |
| 16:46.31 | *** join/#asterisk iq (n=iq@71-214-5-20.omah.qwest.net) |
| 16:46.33 | Skarmeth | a1fa, not in brazil |
| 16:46.41 | a1fa | man, next time, give me a haller |
| 16:46.45 | a1fa | I will ship you the goods |
| 16:46.47 | brodiem | I saw the 301's for $110 |
| 16:46.54 | a1fa | i will charge you $180 per phone |
| 16:46.56 | Skarmeth | get your $ 134 USB and put a 60% of taxes (including transportation) |
| 16:47.02 | a1fa | free shipping |
| 16:47.07 | a1fa | who pays taxes anyway? |
| 16:47.08 | a1fa | :P |
| 16:47.12 | a1fa | maybe customs |
| 16:47.22 | a1fa | you will have to pay customs down there... no sales tax needed |
| 16:47.53 | brodiem | yeah customs is paid by the receiver upon delivery |
| 16:47.59 | a1fa | right :P |
| 16:48.11 | a1fa | i will sell them for $180 with shipping included |
| 16:48.47 | Skarmeth | if you have a cost of $ 15 USD to send it, then your will have $ 146.00 + 60% that results in $ 233,60 plus additional local taxes and transportation costs |
| 16:48.48 | a1fa | !calc 180*20*(180*60/100) |
| 16:48.50 | brodiem | but if you declare the items as a gift (at least shipping to canada anyway) you can get away without paying the duty |
| 16:49.05 | a1fa | yes |
| 16:49.09 | a1fa | i will send you a gift |
| 16:49.14 | a1fa | <PROTECTED> |
| 16:49.17 | brodiem | haha |
| 16:49.22 | a1fa | it is your late christmass and easter present |
| 16:49.27 | willt | LOL |
| 16:49.28 | a1fa | fromt he time since you where born until today |
| 16:49.47 | Skarmeth | a1fa, if you send more that one same object to brazil, it's considered commerce |
| 16:49.49 | a1fa | so what, you are like ~25, i can send you 50 phones |
| 16:49.50 | Skarmeth | :) |
| 16:49.54 | brodiem | a1fa but it happens all the time, you really think customs and border protection are going to put out an investigation to find out if it's a gift? lol |
| 16:50.04 | a1fa | brodiem : i know.. |
| 16:50.10 | Skarmeth | and gifts are limited to $ 50.00 USD (without transportation) |
| 16:50.25 | brodiem | Skarmeth, so each phone costs $2 :) |
| 16:50.30 | a1fa | hehee |
| 16:50.32 | a1fa | of course |
| 16:50.36 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 16:50.36 | a1fa | they are "DEFECTIVE" |
| 16:50.44 | a1fa | so they only cost "2" |
| 16:51.05 | Skarmeth | not here, all packages will be opened and then the local market value will be applied, plus duty and law enforcements |
| 16:51.18 | a1fa | sucks man |
| 16:51.22 | a1fa | i read about brazil |
| 16:51.29 | a1fa | how they steel merchandise from docks |
| 16:51.31 | brodiem | crazy |
| 16:51.36 | Skarmeth | hehehe |
| 16:51.40 | a1fa | so i am sure you can smugle it out |
| 16:51.47 | a1fa | i will send you container |
| 16:51.59 | Skarmeth | We always pay about $ 50.00 more for each thing |
| 16:52.00 | Skarmeth | :) |
| 16:52.24 | jsharp | Smuggle a bunch of phones in a pack of zebras. |
| 16:52.26 | a1fa | its ok |
| 16:52.33 | a1fa | you get to play with babes in g-strings down @ the beach |
| 16:52.44 | brodiem | jsharp, I hear totem poles work good to stuff things with too :) |
| 16:52.45 | a1fa | $50 is a cheap price to pay for p00tang |
| 16:52.47 | vuud | Hmmhesays: well, just found out the voicemail my tester has been reporting is Gizmo's voice mail. So I get no outgoing sound when someone calls in regardless of the context... which makes a lot more sense. Any thoughts in this case? |
| 16:52.53 | a1fa | $50 is your p00tang tax! |
| 16:53.51 | Skarmeth | a1fa, where your work? are a resaller? |
| 16:55.05 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 16:55.13 | a1fa | no man |
| 16:55.16 | a1fa | private contractor |
| 16:56.44 | Micetto | :(( |
| 16:56.49 | *** join/#asterisk ropeguru_work (n=ropeguru@65-121-222-5.dia.static.qwest.net) |
| 16:57.08 | ropeguru_work | Hi all... |
| 16:57.34 | ropeguru_work | I just tried googling but cannot find an answer to a question that I have. |
| 16:57.51 | a1fa | just ask |
| 16:57.58 | salviadud | damn! |
| 16:58.05 | salviadud | my mixmonitor thing |
| 16:58.08 | salviadud | it gets cut off! |
| 16:58.12 | salviadud | why? |
| 16:58.29 | ropeguru_work | I just installed openSUSE 10.0 and have asterisk up and running fine. But I cannot get the zaptel script running correctly to load the drivers automatically. Seems there is a problem with the "functions" routine. |
| 16:58.45 | salviadud | its weird... |
| 16:58.53 | salviadud | i have a long conversation |
| 16:59.09 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 16:59.09 | salviadud | and the output file shows like 14 secs or so of sound.. |
| 16:59.59 | a1fa | hey.. when you call between extensions, do they establish a direct connection? |
| 17:00.00 | ropeguru_work | It is looking for a functions script in the rc.d and there isn't one. Found one in the /etc/sysconfig/hardware/scripts/ and /etc/sysconfig/network/scripts/ but neither are the correct functions script |
| 17:00.11 | a1fa | i have 2 extensions in europe |
| 17:00.18 | a1fa | and it is pointless for them to talk back to mamma |
| 17:00.29 | a1fa | in order to talk to eachother |
| 17:00.34 | a1fa | anyway to move them off the network? |
| 17:00.40 | a1fa | in a direct connection? |
| 17:01.01 | ropeguru_work | r they sip and behind NAT |
| 17:01.09 | a1fa | yeup |
| 17:01.23 | a1fa | ropeguru_work : i will enable port forwarding tho.. |
| 17:01.27 | ropeguru_work | Probably not going to happen then because os SIP not liking NAT ver well.. |
| 17:01.41 | *** join/#asterisk spunz_ (n=spunz@h081217096096.dyn.cm.kabsi.at) |
| 17:01.41 | ropeguru_work | And no, I cannot type today. :-) |
| 17:02.06 | a1fa | ropeguru_work : No NAT? |
| 17:02.07 | a1fa | yes? |
| 17:02.08 | ropeguru_work | If you want to test allow re-invites between the two extensions. |
| 17:02.18 | a1fa | ok |
| 17:02.20 | a1fa | will do that |
| 17:02.30 | ropeguru_work | But I am sure you will probably run into one-way audio issues. |
| 17:03.10 | ropeguru_work | To verify thaey are talking directly do an rtp debug in the asterisk console to make sure it isn't in the middle of the rtp stream/ |
| 17:03.30 | a1fa | will do that |
| 17:03.38 | a1fa | that will be perfect |
| 17:03.44 | a1fa | because if they dont reinvite |
| 17:03.50 | a1fa | they create a delay of 300ms |
| 17:03.52 | a1fa | or more |
| 17:03.58 | ropeguru_work | Right.. |
| 17:04.02 | a1fa | if they reinvite, the delay between the two hosts is 50ms |
| 17:04.02 | a1fa | :P |
| 17:04.08 | ropeguru_work | Nice |
| 17:04.22 | a1fa | yeah.. |
| 17:04.22 | a1fa | the * is in .US |
| 17:04.29 | salviadud | hey, any dude here using mixmonitor? |
| 17:04.49 | a1fa | no dude |
| 17:04.54 | a1fa | only chiX |
| 17:05.18 | salviadud | chiX? |
| 17:05.31 | a1fa | chicks |
| 17:05.33 | a1fa | females |
| 17:05.35 | a1fa | girls |
| 17:05.36 | a1fa | women |
| 17:05.39 | mkl1525 | is there a way to show all atm used channels of the zap devices? |
| 17:05.48 | salviadud | ¬¬ |
| 17:05.52 | a1fa | zap show peers |
| 17:06.19 | salviadud | well then, i would like to get in contact with the girls using mixmonitor |
| 17:06.23 | a1fa | or not |
| 17:06.31 | a1fa | salviadud : perver |
| 17:06.37 | salviadud | hahaha |
| 17:06.50 | salviadud | duuude, it's a real issue here |
| 17:07.03 | salviadud | damn thing cuts off... |
| 17:07.07 | salviadud | makes no sense |
| 17:07.07 | *** join/#asterisk docelm0 (n=docelmo@66.239.192.34.ptr.us.xo.net) |
| 17:07.15 | docelm0 | WHADUP! |
| 17:07.18 | salviadud | i blame the chameleon. damn suse |
| 17:08.07 | mkl1525 | alfa, is this a 1.2 feature - my current 1.0 just knows zap show channel(s)? |
| 17:08.29 | Damin | jbot centosbug |
| 17:08.34 | jbot | well, centosbug is a problem with the latest Centos 4.2 kernel. To fix it, edit the file /usr/src/kernels/2.6.9-34.EL-i686/include/linux/spinlock.h and change 'rw_lock_t' on line 407 to 'rwlock_t'. Make sure you're NOT using gcc4, as that doesn't work either. |
| 17:08.34 | a1fa | yeah do zap help |
| 17:09.04 | mogorman | Damin, are you here? |
| 17:09.17 | mogorman | in san jose that is |
| 17:09.17 | Damin | mogorman: Here as in VON? No. Here as in the channel? Yes. |
| 17:10.05 | *** join/#asterisk htims (n=htims@Vc97c.v.pppool.de) |
| 17:10.22 | *** join/#asterisk justinu (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 17:12.18 | astra^^ | Mar 15 11:11:27 WARNING[12997]: channel.c:2333 set_format: Unable to find a codec translation path from ulaw to g729 |
| 17:12.27 | jarrod | you dont have g729 licenses? |
| 17:12.56 | astra^^ | asterisk supports g723 pass tru right..? |
| 17:13.03 | jarrod | yes |
| 17:13.06 | jarrod | but that says g729 |
| 17:13.13 | Damin | astra^^: Yep. Works great. |
| 17:13.19 | *** join/#asterisk MikeJ__ (n=vircuser@c-71-228-12-47.hsd1.il.comcast.net) |
| 17:13.24 | astra^^ | ok .. bt hw do i do it without transcoding it |
| 17:13.51 | jarrod | channel.c: Avoiding initial deadlock for XXX |
| 17:13.55 | jarrod | why do i receive a lot of these? |
| 17:14.12 | *** join/#asterisk kc5cqm (n=michael@puffin.tamucc.edu) |
| 17:14.46 | kc5cqm | has anyone here played with Phil Zimmermann's zfone utility yet? |
| 17:14.55 | kc5cqm | or for that matter...got it to build? |
| 17:15.47 | kc5cqm | for those of y'all who don't know, zfone encrypts the rtp datastream |
| 17:16.28 | *** join/#asterisk ToTo (n=ToTo@host33-161.pool870.interbusiness.it) |
| 17:16.39 | *** part/#asterisk kc5cqm (n=michael@puffin.tamucc.edu) |
| 17:17.14 | *** join/#asterisk kc5cqm (n=michael@puffin.tamucc.edu) |
| 17:17.37 | kc5cqm | Anyone play with zfone yet? |
| 17:17.53 | Abydos313 | not yet |
| 17:21.43 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
| 17:23.42 | *** join/#asterisk _Thor (i=CS@user-vc8fl7l.biz.mindspring.com) |
| 17:24.03 | _Thor | No contaban con mi astusia! |
| 17:25.07 | ^HeLL^ | chavo del 8? |
| 17:25.17 | *** part/#asterisk nusse (i=nusse@mega2000.de) |
| 17:25.31 | _Thor | Hell: wrong, Chapulin Colorado |
| 17:26.00 | ^HeLL^ | ohh right! right! :) |
| 17:26.08 | ^HeLL^ | bad memory... |
| 17:26.49 | _Thor | Hell: mmm, slap on wrists! |
| 17:26.57 | ^HeLL^ | xD |
| 17:28.35 | _Thor | I have a question... for calls extension to extension, why in the world I can not force g729? |
| 17:29.56 | ^HeLL^ | 90 users registered... uau!... :D |
| 17:30.14 | ^HeLL^ | _Thor : use canreinvite=yes |
| 17:30.47 | ^HeLL^ | or buy g729 licenses... :) |
| 17:31.00 | ^HeLL^ | if you need transcoding... |
| 17:31.06 | _Thor | I can't, extensions are in 2 different networks, they will not be able to listen to each oterh |
| 17:31.47 | ^HeLL^ | use gateway phone option |
| 17:31.56 | domingues | Hello All, I am getting some problem using Codec G729 trought ooh323 drive, when I send the call, it s completeto the calling but the Caller still hear calling tone, does anyone have the same probllems or have some idea? |
| 17:32.07 | ^HeLL^ | to add the router that connects both phones |
| 17:34.26 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 17:37.58 | vuud | Is there a way to tell what Codec a connection has settled on? I am getting no sound at all. |
| 17:39.01 | Fedoracore6 | hai all |
| 17:39.04 | *** join/#asterisk mover (n=dlu@213.9.46.7) |
| 17:39.26 | Fedoracore6 | i doing the code for update student databases but when i run this code have error |
| 17:39.37 | Fedoracore6 | like this |
| 17:39.39 | Fedoracore6 | http://pastebin.com/603774 |
| 17:40.17 | Fedoracore6 | its i have put something else or have other code for update |
| 17:42.31 | *** join/#asterisk scubasteve (n=steve@ns1.misel.com) |
| 17:42.36 | *** part/#asterisk scubasteve (n=steve@ns1.misel.com) |
| 17:42.47 | *** join/#asterisk Utah_Dave (n=boucha@0-1pool139-55.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 17:44.24 | SpaceBass | anyone had problems with faxes coming across as blank pages? |
| 17:44.33 | brodiem | is there a CLI command to view the contents of variables? |
| 17:44.33 | SpaceBass | i recieve the correct number of pages, but they are always blank |
| 17:44.49 | *** join/#asterisk dippo (n=cwage@quietlife.net) |
| 17:45.04 | dippo | hi. this is a weird one: when I log into an adhoc device with *11 and enter my extension & password, everything appears to work |
| 17:45.15 | dippo | except while that user is logged in, DTMF via my IAX2 trunk stops working |
| 17:45.19 | dippo | it starts working again if I log out via *12 |
| 17:45.33 | dippo | DTMF appears to work via a Zap channel during that time |
| 17:45.34 | dippo | wtf? |
| 17:46.12 | SpaceBass | can anyone help with some dring and fax issues? |
| 17:46.32 | SpaceBass | I have installed fax and pdf support, but I only get blank pages |
| 17:47.13 | *** join/#asterisk MstlyHrmls (n=mh@66.193.14.132) |
| 17:48.23 | SpaceBass | my dring problem is quite complex... 2 zap lines: zap/1 is personal zap/2 is business and has a dring.... everything works fine in terms of routing until someone calls zap/2 then ALL calls come in as if from zap/2....if I comment out the dring, it works fine...personal calls to person extens, work calls get a cID prefix and go to work extensions |
| 17:50.08 | *** join/#asterisk zgor (n=zgor@61.Red-80-36-3.staticIP.rima-tde.net) |
| 17:50.11 | zgor | hi :) |
| 17:51.35 | zgor | (sorry for my bad english), i trying to link an Avaya PBX to Asterisk by PRI. Using a E1/T1 crossover cable, but i have always RED light ... |
| 17:51.58 | willt | Does gotoiftime only transfer to contexts or can I run macros based on it? |
| 17:51.59 | *** join/#asterisk Lino` (n=Lino@i577BDC76.versanet.de) |
| 17:52.54 | zgor | i build myself the cable (1->4 , 2->5 , 4->1,5->2). before next step, i think first is to have green light, isnt it ? |
| 17:53.45 | *** join/#asterisk Lino` (n=Lino@i577BDC76.versanet.de) |
| 17:54.48 | noky | my asterisk unregister with my gatekeeper |
| 18:04.59 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
| 18:06.52 | *** join/#asterisk spatulamaan (n=ggilmore@ip66-107-33-196.z33-107-66.customer.algx.net) |
| 18:11.07 | *** join/#asterisk FuriousGeorge (n=Brian@ool-43536ea8.dyn.optonline.net) |
| 18:12.35 | fu3 | quick.. someone send me $200 |
| 18:13.32 | starlein | go working |
| 18:13.44 | mutilator | paypal address? |
| 18:14.15 | starlein | hehe mutilator you can get mine |
| 18:14.41 | diLLec | wait. i will register for that :-) |
| 18:15.52 | SpaceBass | I have installed fax and pdf support, but I only get blank pages |
| 18:16.02 | ruza | any idefisk ebuild in the wild ? |
| 18:16.15 | *** join/#asterisk gbodemantv (n=gbodeman@216.142.38.154) |
| 18:16.20 | gbodemantv | hi all |
| 18:16.34 | gbodemantv | anyone using Flash Operator Panel?? |
| 18:16.58 | justinu | man, i'm kinda frustrated with AMI |
| 18:17.00 | ^HeLL^ | gbodemantv : me for example... :) |
| 18:17.02 | justinu | trying to originate calls |
| 18:17.05 | *** join/#asterisk kpettit (n=keith@69.15.174.114) |
| 18:17.24 | justinu | but the new channel events with the channel id, unique id don't have any association with the originate action commands! |
| 18:18.34 | exonic | justinu, what do you need them associated with? |
| 18:18.41 | exonic | justinu, on a bigger picture, what's it used for? |
| 18:19.21 | justinu | i need to originate calls, then track the events that occur on them |
| 18:19.27 | justinu | while ignoring calls I didn't originate |
| 18:20.52 | Fedoracore6 | hai all |
| 18:25.20 | *** join/#asterisk jbot_ (i=ibot@pdpc/supporter/active/TimRiker/bot/apt) |
| 18:25.20 | *** topic/#asterisk is Asterisk: The Open Source PBX -=- http://www.asterisk.org/ |
| 18:26.17 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 18:27.26 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 18:27.53 | Fedoracore6 | hemm |
| 18:28.05 | Fedoracore6 | i wanna do update from my databases |
| 18:28.16 | kpettit | I'm trying to setup a que. Is there anyway to have phones auto-login to a queue? |
| 18:28.40 | Fedoracore6 | http://pastebin.com/603774 |
| 18:28.43 | kpettit | The phones are only used in the queue and I'm using the "ringall" strategy, so it would be easuer if there always loggged in |
| 18:28.53 | Fedoracore6 | so i try this code but feild |
| 18:28.58 | *** part/#asterisk UlbabraB (n=salama@host241-43.pool8172.interbusiness.it) |
| 18:29.14 | Fedoracore6 | its have some code or eample for update |
| 18:29.22 | Fedoracore6 | cos i look in my code is wrong |
| 18:29.38 | MikeJ__ | kpettit, you can just make them static members... |
| 18:29.45 | MikeJ__ | so they are always logged in |
| 18:29.53 | kpettit | that would be perfect. |
| 18:30.01 | MikeJ__ | just don't use chan_agent |
| 18:30.04 | kpettit | how do I do that. All the tutorials I've seen focus on doing the login |
| 18:30.14 | MikeJ__ | just put the sip or zap or whatever in the members |
| 18:30.25 | MikeJ__ | like you would AGENT\blah |
| 18:30.49 | kpettit | ah so instead of member => Agent/100 do member => Sip/100 ?? |
| 18:30.53 | MikeJ__ | any valid peer can be a queue member, not just agents |
| 18:30.59 | MikeJ__ | yeah, somthin like tha |
| 18:31.11 | kpettit | sweet, that'lll be perfect. |
| 18:31.14 | MikeJ__ | assuming SIP/100 is setup and such |
| 18:33.43 | Fedoracore6 | its my code wrong |
| 18:33.58 | Fedoracore6 | or i have modified the code |
| 18:34.13 | Fedoracore6 | <PROTECTED> |
| 18:34.19 | kpettit | MikeJ__, Just gave it a try, it just hangs up after it answers. |
| 18:34.52 | kpettit | http://pastebin.com/603878 |
| 18:34.55 | kpettit | that's what I'm using |
| 18:35.11 | kpettit | The Sip/XXX are all valid extensions |
| 18:35.55 | Fedoracore6 | i think fo my :) |
| 18:37.02 | kpettit | I can see the Answer, Wait, and the Queue executing. It then starts, then stops music on hold then hangs up |
| 18:37.08 | gbodemantv | hell: I am trying to implement FOP but use REALTIME |
| 18:37.14 | gbodemantv | any idea how I can connect the 2 |
| 18:37.40 | kpettit | write a cron job to update FOP |
| 18:37.51 | justinu | exonic: looks like I can query the channel variables per new call |
| 18:38.00 | justinu | that's one way for me to set some kind of tag. |
| 18:38.06 | justinu | just requires another server roundtrip :( |
| 18:39.17 | *** join/#asterisk Zodiacal (i=hehehe@bdsl.66.14.242.199.gte.net) |
| 18:39.21 | Zodiacal | anyone know of a way to tell if a phone number is a local call? (im not at the location of the phone line tho) |
| 18:39.50 | fu3 | call the operator |
| 18:40.27 | *** part/#asterisk ropeguru_work (n=ropeguru@65-121-222-5.dia.static.qwest.net) |
| 18:40.41 | austinnichols101 | ask the caller where they're at |
| 18:41.22 | *** join/#asterisk epablo (n=epablo@201.242.75.16) |
| 18:41.33 | epablo | Hi people |
| 18:41.43 | fu3 | hi |
| 18:41.47 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 18:41.53 | PakiPenguin | hello :) |
| 18:41.55 | fu3 | OH MY GOD. #ASTERISK IS MADE OF PEOPLE!!!!!! |
| 18:41.55 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 18:42.14 | *** join/#asterisk DaveHope (n=dave@62.69.60.24) |
| 18:42.18 | PakiPenguin | anyone here used sangoma's single t1/e1 card? |
| 18:42.22 | *** join/#asterisk klasstek (n=nunyobiz@sta-206-168-96-67.rockynet.com) |
| 18:42.27 | fu3 | im using a sangoma a104d |
| 18:42.52 | PakiPenguin | fu3, i want just a single t1 , very less useage , is there an echo cancelation module available? |
| 18:42.53 | SpaceBass | I have installed fax and pdf support, but I only get blank pages |
| 18:42.58 | DaveHope | Hey all. Quick question, is there anywhere home users can obtain paid asterisk support ? - I'd really like to get asterisk going (cant get outgoing SIP calls to work) and don't expect help for free. Any ideas of where I can look ? |
| 18:43.15 | fu3 | I dont know. I sprung for the echo cancellation for the A104 but I sure had to pay for it. |
| 18:43.21 | epablo | I've been reading about res_data and Asterisk RealTime Architecture. But I can't find the modules in the 1.2.x dist |
| 18:43.37 | austinnichols101 | davehope: a@H? |
| 18:43.42 | epablo | <PROTECTED> |
| 18:43.50 | fu3 | you should call Sangoma PakiPenguin. |
| 18:43.52 | fu3 | or email them. |
| 18:43.55 | PakiPenguin | i see |
| 18:44.30 | DaveHope | austinnichols101: Don't really want to go down that route. |
| 18:44.48 | DaveHope | epablo: Asked before, got one part resolved but not the other. No luck on the forum/mailing list either. |
| 18:44.51 | justinu | anyways, if you want vanilla asterisk support (no AMP/AAH) you're int he right place |
| 18:44.56 | austinnichols101 | DaveHope: for A@H there are several people. I've used http://baldwintechsolutions.com/aahsupport.php (ask for Tom Vile) |
| 18:45.14 | austinnichols101 | and there are a LOT of people here who will do paid asterisk stuff |
| 18:45.20 | epablo | DaveHope: What do you wan't to do' |
| 18:45.23 | DaveHope | austinnichols101: Cool. Will ask a bit later then :) |
| 18:45.29 | justinu | yep, you might find your answer for free too |
| 18:45.34 | *** join/#asterisk x86 (n=x86@p3m/member/x86) |
| 18:45.35 | justinu | depends on how you ask ;) |
| 18:45.41 | epablo | LoL |
| 18:45.48 | austinnichols101 | great group here as long as you've done your homework first |
| 18:46.00 | austinnichols101 | otherwise, prepare yourself for the smackdown |
| 18:46.04 | justinu | lol |
| 18:46.29 | DaveHope | epablo: Just make outgoing sip calls using my sipgate number, have incomming calls go through a menu (simple) through asterisk so I can have multiple users. |
| 18:46.31 | epablo | Yeap.. been there. Have been smaked.. LOL |
| 18:46.45 | austinnichols101 | I've been bitch-slapped a couple of times |
| 18:46.49 | DaveHope | lol |
| 18:47.00 | DaveHope | I'll ask you all after I've had my tea. Just got in from work :) |
| 18:47.02 | austinnichols101 | ouch - get off. it hurts! |
| 18:47.52 | epablo | DaveHope: I don't know sipgate but that setup sounds simple |
| 18:48.40 | *** part/#asterisk oden (n=oden@194-237-146-22.customer.telia.com) |
| 18:48.46 | exonic | justinu, what's the command to show channel variables? I could use this also. |
| 18:49.24 | exonic | justinu, are you refering to 'ACtion: Status' ? |
| 18:49.29 | justinu | ACTION: GetVar |
| 18:49.29 | justinu | Channel: SIP/5060-44d225d0 |
| 18:49.29 | justinu | Variable: <whatever> |
| 18:49.53 | exonic | justinu, ahh. Take a look at status. it lets you pass a ActionID parameter. |
| 18:50.21 | epablo | I've been reading about res_data and Asterisk RealTime Architecture. But I can't find the modules in the 1.2.x dist. Any one know is that is still Alive? I'm looking into setting up SER-Asterisk for an Asterisk@large setup |
| 18:50.41 | PakiPenguin | can anyone suggest me something for six fxs lines? cards? channelbank or what? |
| 18:50.41 | justinu | exonic: yeah, i was noticing that also |
| 18:51.17 | justinu | PakiPenguin: TDM2400 w/ 3 modules? |
| 18:51.18 | epablo | PakiPenguin: I saw some 8 port ATA's on voipsupply |
| 18:51.26 | justinu | 8 port SIP gateways are $$$ |
| 18:51.34 | justinu | but would be a nice solution |
| 18:52.34 | *** join/#asterisk mxmasster (n=mxmasste@ppp-71-138-117-215.dsl.irvnca.pacbell.net) |
| 18:52.45 | mxmasster | in the asterisk console, what is the command that i use to test the dialpan? |
| 18:52.52 | mxmasster | i'e what matches a pattern |
| 18:53.01 | epablo | DaveHope: You should be able to do this with a simple SIP friend user setup and an IVR |
| 18:53.03 | gbodemantv | kpettit: what kind of cronjob? |
| 18:53.53 | exonic | mxmasster, I'm not aware of one, Although you could originate a call in whatever context/exten and watch ;) |
| 18:55.09 | *** join/#asterisk asterboy (n=Snake@S01060204ee2b6007.ed.shawcable.net) |
| 18:55.39 | kpettit | gbodemantv, something to query the database to get all the sip info then just spit that in the format you want for FOP |
| 18:55.53 | justinu | exonic: i'm gonna take a look thru the code, see if it's very difficult to associate the ActionID w/ the NewChannel event. |
| 18:56.00 | justinu | just would make my life alot easier. |
| 18:56.00 | asterboy | When using pots lines for a rotary, the line provider sets up the Rotary right? Not * |
| 18:56.16 | justinu | asterboy: all POTS lines work with DP (dial pulse) |
| 18:56.18 | kpettit | FOP can't grab info from a db (that I know of) so you'll have to find some way to grab the info out of the db and re-write the op_buttons.cfg file |
| 18:56.21 | justinu | no special config required. |
| 18:56.36 | mxmasster | what is a tool that i can use on linux to run a traceroute and calculate jitter? |
| 18:56.42 | kpettit | gbodemantv, and restart that op_server process as well |
| 18:56.48 | kpettit | mxmasster, use mtr |
| 18:56.56 | asterboy | If I want a rotary though, do I get the line provider to setup or can I do it in *? |
| 18:56.59 | kpettit | mtr domainname.com |
| 18:57.00 | *** part/#asterisk epablo (n=epablo@201.242.75.16) |
| 18:57.08 | asterboy | this is for pots lines. |
| 18:57.09 | mxmasster | kpettit: mtr will show jitter? |
| 18:57.14 | kpettit | mxmasster, you can also adjust the outpuit of thst do display different fields |
| 18:57.42 | justinu | asterboy: by rotary, you mean routing to different numbers based on the status of the number dialed? |
| 18:57.46 | kpettit | mxmasster, yes I use it all the time. |
| 18:57.50 | justinu | not like a rotary telephone |
| 18:57.53 | asterboy | no incoming |
| 18:58.11 | mxmasster | kpettit: i'm looking at the man page - i don't see the jitter option, what should i pass it? |
| 18:58.13 | asterboy | like I have a line coming in on an FXO via pots... |
| 18:58.24 | kpettit | mxmasster, after you get into mtr you can press "O" to order the fields to get the data displayed the way you want |
| 18:58.35 | asterboy | I want that line to be *not* ring busy if someone else calls... |
| 18:58.38 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 18:58.40 | *** join/#asterisk japerry (n=japerry@seattle.asemblon.com) |
| 18:58.47 | justinu | asterboy: yes, that's up to your phone company at this point. |
| 18:58.48 | asterboy | so it goes to the next available FXO port. |
| 18:58.54 | asterboy | ya that is what I thought. |
| 18:59.05 | mxmasster | kpettit: are you talking about the command line or the gui? |
| 18:59.13 | kpettit | mxmasster, press "O" when in mtr and jhoose the J options for Jitter or M for Mean jitter, or X for Worst jitter or all the above |
| 18:59.14 | asterboy | cause there is no way for * to magically switch the call to another phone line. |
| 18:59.29 | Hmmhesays | is stubblecanoe.com a good name for a website? or too hard to spell |
| 18:59.32 | kpettit | mxmasster, those are the commandline otions, but there is also a GUI to mtr, depending on how it was compiled |
| 18:59.35 | justinu | asterboy: no, because the call won't ever be sent to you |
| 18:59.43 | justinu | no way to control it if you never get it |
| 18:59.48 | *** join/#asterisk lemmy (n=lemmy@developer.g2gui.net) |
| 18:59.50 | kpettit | mxmasster, I always use the console though so I'm not sure what the GUI options are |
| 18:59.56 | lemmy | hi |
| 18:59.57 | asterboy | now on the other side, if I'm calling out, * can use a group to line hunt the next available. |
| 18:59.58 | mxmasster | hmm |
| 19:00.02 | justinu | asterboy: correct. |
| 19:00.08 | asterboy | ok |
| 19:00.17 | tzafrir | Is there any way to get zaptel to do echo cancelation to FXS extensions? |
| 19:00.40 | mxmasster | kpettit: i don't seem to have that option - what version of mtr are you using? mtr-0.54-10 |
| 19:01.10 | *** part/#asterisk astra^^ (n=muhajir_@59.145.104.74) |
| 19:01.13 | tzanger | mxmasster: of course |
| 19:01.34 | salviadud | does anyone here have any probs with mixmonitor? |
| 19:01.41 | mxmasster | tzanger: ? |
| 19:01.54 | kpettit | mxmasster, net-analyzer/mtr-0.69-r1 |
| 19:02.00 | justinu | asterboy: instead of rotary, you might want to use the term line hunting, or hunt group |
| 19:02.04 | justinu | with your telco |
| 19:02.12 | justinu | sometimes they don't understand unless you speak their language |
| 19:02.13 | salviadud | perhaps i could teach you guys how to make tacos,that should make up for the info i need very nicely, i believe |
| 19:02.15 | mxmasster | kpettit: thanks |
| 19:02.27 | tzanger | mxmasster: just define echocancel=yes (or a number) and it should work |
| 19:02.41 | justinu | salviadud: hola, como estas? |
| 19:02.58 | salviadud | mucho muy angry |
| 19:03.07 | salviadud | hehe |
| 19:03.12 | tzafrir | tzanger, it seems to have no effect. And looking at the code, there seem to be a number of thests there for FXS signalling |
| 19:03.18 | justinu | tzager: (10:58:23) tzafrir: Is there any way to get zaptel to do echo cancelation to FXS extensions? |
| 19:03.21 | salviadud | not really, im just bewildered at why my mixmonitor application fails sometimes |
| 19:03.44 | salviadud | if ANYBODY, just anybody, has the same problem |
| 19:03.46 | salviadud | give me a call |
| 19:04.20 | justinu | how's it failing? |
| 19:04.30 | salviadud | basically |
| 19:04.33 | *** join/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 19:04.39 | tzanger | tzafrir: interesting |
| 19:04.43 | salviadud | it records the beginning of a call |
| 19:04.57 | salviadud | then by some weird reason it stops recording before i hang up |
| 19:05.02 | salviadud | incomplete recordings |
| 19:05.03 | justinu | sip? |
| 19:05.07 | salviadud | yeah, sip |
| 19:05.11 | salviadud | is that bad? |
| 19:05.14 | justinu | sip to sip? |
| 19:05.18 | justinu | no, i use mixmonitor on sip all the time |
| 19:05.21 | salviadud | not really |
| 19:05.25 | salviadud | sip to iax |
| 19:05.26 | justinu | sip to zap? |
| 19:05.27 | justinu | oh |
| 19:05.35 | justinu | ok, i've used SIP-SIP, and SIP-ZAP |
| 19:05.36 | salviadud | i was recording some toll free number conversations |
| 19:05.37 | justinu | no issues |
| 19:05.40 | justinu | never tried sip-iax |
| 19:05.56 | justinu | submit it as a bug on the bug tracker maybe, sounds like some kinda bug to me |
| 19:05.57 | salviadud | so, sip to sip should be fine then... |
| 19:06.14 | *** join/#asterisk outtolunc (n=me@adsl-66-218-53-172.dslextreme.com) |
| 19:06.27 | salviadud | i've never submitted a bug |
| 19:06.31 | salviadud | where do i go? |
| 19:06.40 | justinu | bugs.digium.com |
| 19:06.50 | justinu | have you looked thru your full log? |
| 19:06.57 | justinu | see if it prints out any debug/error message? |
| 19:07.07 | salviadud | mixmonitor claims it recorded the whole call |
| 19:07.29 | salviadud | but my output file is short, in comparison to the real call |
| 19:07.33 | justinu | ic |
| 19:07.40 | justinu | so no indication of failure at all |
| 19:07.45 | salviadud | exactly |
| 19:07.52 | justinu | you're not recording to an NFS share or anything, right? and you have plenty of disk space? |
| 19:07.58 | salviadud | you gotta take my word for it, i guess |
| 19:08.03 | salviadud | it is not an nfs share |
| 19:08.07 | salviadud | and yes, plenty of space |
| 19:08.13 | justinu | sounds like a bug then |
| 19:08.24 | justinu | try it with SIP-SIP see if you get any better esults |
| 19:08.26 | justinu | results |
| 19:08.30 | justinu | if so, then definitely submit that |
| 19:08.35 | salviadud | actually, i did |
| 19:08.43 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@c-71-228-12-47.hsd1.il.comcast.net) |
| 19:08.44 | salviadud | i tried sip-sip just now |
| 19:08.45 | salviadud | no problems |
| 19:08.57 | salviadud | sip to iax doesn't work that good |
| 19:09.05 | justinu | what asterisk ver? |
| 19:09.07 | salviadud | i can record 30 secs at max |
| 19:09.14 | salviadud | 1.24 |
| 19:09.21 | justinu | perhaps try down/upgradaing also |
| 19:09.27 | justinu | you might be able to point out to them where it broke ;) |
| 19:09.51 | SpaceBass | I have installed fax and pdf support, but I only get blank pages |
| 19:09.53 | salviadud | i read the changelog for 1.25 |
| 19:10.00 | SpaceBass | anyone know why that might be? |
| 19:10.03 | salviadud | i think this bug might still exist there too |
| 19:10.25 | *** part/#asterisk frenzy (n=frenzy@196.45.144.40) |
| 19:11.23 | *** join/#asterisk backblue (n=moo@87-196-66-128.net.novis.pt) |
| 19:11.51 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 19:12.08 | asteriskmonkey | digium changed there web site |
| 19:12.26 | asteriskmonkey | :( horrible for tech docs now |
| 19:12.28 | backblue | yes, and broke a couple of links too! :P |
| 19:12.31 | *** part/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 19:12.35 | *** join/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 19:12.56 | asteriskmonkey | cant seem to find a sample config for the 2400 anymore |
| 19:15.04 | *** join/#asterisk Curus (n=Curus@x1-6-00-12-17-df-1b-be.k182.webspeed.dk) |
| 19:16.00 | justinu | exonic: you still here? |
| 19:16.07 | *** join/#asterisk Aurs (n=Aurs@a217-118-41-157.bluecom.no) |
| 19:16.43 | asterboy | How do I get rid of haveing to dial say "9" before making a call? |
| 19:16.59 | Nugget | edit your dialplan so that you don't have to dial say "9" before making a call. |
| 19:17.14 | asteriskmonkey | lol your config man extensions.conf |
| 19:18.17 | tsume | what kind of monkey uses 9 before calls still? |
| 19:18.18 | salviadud | r u behind an oldchool pbx? |
| 19:18.35 | tsume | any system I setup can identify whether its internal or external |
| 19:19.22 | justinu | exonic: found the answer to my problem |
| 19:19.30 | justinu | by reading the manager.c source |
| 19:20.05 | Nugget | that's cheating! |
| 19:20.30 | justinu | lol |
| 19:20.51 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 19:21.43 | *** join/#asterisk SplasPood (n=jwb@gate.lga2.us.voxel.net) |
| 19:21.55 | *** join/#asterisk troyb1 (n=central@CPE00907f17e478-CM014300013422.cpe.net.cable.rogers.com) |
| 19:22.20 | salviadud | why is THAT cheating? |
| 19:22.32 | salviadud | cheatin is more like cheating on your GF |
| 19:22.35 | justinu | in commercial software development, we don't have source code |
| 19:22.55 | troyb1 | do you have munchies? |
| 19:22.56 | tsume | salviadud: yeah, a woman screwing the friendly pooch instead of the husband ;) |
| 19:23.16 | justinu | cheating on your GF isn't cheating... it's not like you're married |
| 19:23.23 | justinu | troyb1: not yet. |
| 19:23.30 | salviadud | justinu is right |
| 19:23.35 | troyb1 | justinu i couldnt do anything without munchies =P |
| 19:23.38 | tsume | right, you can have several gfs :) |
| 19:23.39 | salviadud | it can't be called cheating unless you are married |
| 19:23.47 | troyb1 | meh a girlfriend is like a taxi, you pay and move on. errr |
| 19:23.51 | justinu | you might lose your GF |
| 19:23.54 | justinu | but oh well :) |
| 19:23.59 | justinu | time to get a new one |
| 19:24.13 | tsume | bah, women need to accept, there might not always be just one love |
| 19:24.17 | tsume | its about having chilren ;) |
| 19:24.25 | justinu | troyb1: munchies come later... after 4:20 :P |
| 19:24.36 | troyb1 | justinu thats 2 hours away for me ;) |
| 19:24.40 | justinu | bah |
| 19:25.01 | tsume | I guess its more common for the little people can only get one girl friend to marry them and only have them because they can't pick up women ;) |
| 19:25.12 | *** join/#asterisk tahorg (i=tahorg@lutin.jard.in) |
| 19:25.16 | tsume | or impregnate ;) |
| 19:25.23 | troyb1 | doing one of those: "To find your party please the first 4 digits of their last name" |
| 19:25.35 | justinu | my bride-to-be is somewhat interested in polygamy |
| 19:25.46 | justinu | she thinks it would be cool for me to have another wife to cook for me, and stuff |
| 19:25.50 | salviadud | justinu, are we talking swinger-like here? |
| 19:25.52 | justinu | i'm thinking "right on" |
| 19:25.56 | tsume | justinu: would be intresting |
| 19:26.01 | justinu | it's not swinging if you're all married :) |
| 19:26.01 | tsume | living in the same house |
| 19:26.05 | troyb1 | shit justinu i want 3 wives, one to cook.. one to clean and one for :) |
| 19:26.05 | tsume | alpha male! :D |
| 19:26.13 | justinu | hell yeas |
| 19:26.13 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 19:26.13 | Abydos313 | sounds good until she wants another partner |
| 19:26.20 | tsume | group sex! :D |
| 19:26.27 | justinu | she can have the other wife whenever she wants |
| 19:26.28 | willt | sounds like 3 headaches to me |
| 19:26.29 | justinu | =) |
| 19:26.29 | vuud | crap, I have enough trouble dealing with one woman, never mind 3 nagging and pestering me |
| 19:26.32 | troyb1 | there should enver be more then one mail though :P |
| 19:26.34 | troyb1 | *male |
| 19:26.36 | justinu | it could definitely be more headaches |
| 19:26.39 | salviadud | maybe if you're mormon, it's socially acceptable to have lots of wives |
| 19:26.39 | justinu | troyb1: i agree |
| 19:26.43 | willt | vuud: EXACTLY! |
| 19:26.45 | justinu | i might become mormon |
| 19:26.50 | salviadud | and you could get them all SIP phones |
| 19:26.50 | troyb1 | vuud it cant be that bad :) |
| 19:27.01 | troyb1 | salviadud but there's only one 69 :P |
| 19:27.01 | tsume | women I see better be able to kick my ass, or be a bit muscular |
| 19:27.05 | salviadud | and in extensions.conf, they can't talk to each other |
| 19:27.09 | tsume | but not be totally more rip than me :) |
| 19:27.38 | justinu | sounds dangerous |
| 19:27.46 | troyb1 | salviadud thats so yesterday, you have to make it so that it rings but they cant speak :P |
| 19:27.49 | tsume | better sex, and better child delivery ;) |
| 19:27.52 | salviadud | i like girls that are not fat, yet, they got some meat on them |
| 19:28.01 | tsume | rip == rocks! :D |
| 19:28.03 | Abydos313 | i have an issue when another caller is called the msg just stops after like 2 sec. you don't hear what the attendant has to say |
| 19:28.04 | troyb1 | haha |
| 19:28.14 | tsume | they are tighter in their conchas too ;) |
| 19:28.20 | salviadud | tight conchas! |
| 19:28.24 | troyb1 | heh |
| 19:28.25 | salviadud | you mean pucha |
| 19:28.36 | salviadud | well, we got lots of names for pussy down here... |
| 19:28.45 | troyb1 | salviadud come to think of it we need one to digest the newspaper in the morning for us ;) |
| 19:28.52 | justinu | any girl can develop that skill |
| 19:28.58 | justinu | it's all about the muscle control |
| 19:28.59 | vuud | Three woman... god grant mercy on your poor misguided soul |
| 19:29.05 | justinu | just 2 |
| 19:29.11 | vuud | and mormons can't drink coffee |
| 19:29.13 | tsume | plus its nice when you have large equiptment and large testicles. They love the slapping ;) |
| 19:29.14 | troyb1 | hypa7ia :) |
| 19:29.26 | tsume | vuud: I want 3 women :D |
| 19:29.28 | salviadud | teabagging should be a sport |
| 19:29.32 | Abydos313 | haha |
| 19:29.33 | justinu | hah |
| 19:29.34 | troyb1 | haha thats what i was going to say! |
| 19:29.51 | Abydos313 | the movie 'waiting' had some teabagging going on |
| 19:29.53 | tsume | too bad humans can't hit hard as canines ;) |
| 19:29.56 | justinu | no - now is the time to talk about teabagging |
| 19:30.02 | vuud | * needs help getting inbound gizmo to work |
| 19:30.03 | justinu | not dialplans |
| 19:30.17 | troyb1 | this isnt the asterisk channel :P |
| 19:30.23 | Nivex | go join #teabagging then |
| 19:30.24 | tsume | asterisk is full of peopel who get laid often, unlike those nortel guys :P |
| 19:30.26 | Darwin35 | then lick mty teabag |
| 19:30.44 | salviadud | yeah, take it in the eye mofo! |
| 19:30.57 | tsume | we are alpha, they are only betas ;) |
| 19:31.07 | Darwin35 | I am Gama |
| 19:31.12 | tsume | Darwin35: heh :) |
| 19:31.39 | willt | yes switch to asterisk/voip not only will you save money you get helpfull pointers on teabagging! |
| 19:31.55 | troyb1 | willt heh. |
| 19:31.58 | Darwin35 | I want to learn real time |
| 19:31.59 | tsume | yes! :D |
| 19:32.07 | willt | real time teabagging? |
| 19:32.21 | Darwin35 | but I have alot of exten that us astdb for status checking |
| 19:32.23 | Abydos313 | which conf file is that in..heh |
| 19:32.23 | troyb1 | RT&T |
| 19:32.37 | troyb1 | Abydos313 asterisk.conf :P |
| 19:32.38 | asterboy | For some reason I can't make a call unless I have _9. in my dial plan |
| 19:32.39 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 19:32.43 | salviadud | yeah, when IAX2 is able to send out video in a nice way |
| 19:32.46 | asterboy | 9 can be whatever digit. |
| 19:32.57 | asterboy | I thought you could just use a . |
| 19:33.01 | salviadud | i'm gonna teabag the camera, it would be a nice gesture from my country |
| 19:33.03 | asterboy | "." |
| 19:33.13 | troyb1 | salviadud you do that =P |
| 19:33.15 | salviadud | everybody will start doing it at the beginning of important meetings |
| 19:33.17 | asterboy | lol, teabag |
| 19:33.42 | salviadud | well, videoconferences |
| 19:33.51 | Abydos313 | start with the camera real close so people can't tell what it is, then slowly back the camera away to horrorify the viewer..haha |
| 19:34.18 | salviadud | horrify? no way man, this is love |
| 19:34.22 | salviadud | teabagging is love |
| 19:34.28 | Abydos313 | heh |
| 19:34.33 | salviadud | if you can't see it that way holmes |
| 19:34.41 | salviadud | you need a good spoon of teabagging |
| 19:34.48 | asterboy | Here is my teabag...err...line from extensions.conf: exten => _9.,1,Dial(Zap/2) |
| 19:34.52 | salviadud | or you might need to teabag your GF |
| 19:34.53 | Abydos313 | to the unsuspecting viewer i think it might be startling |
| 19:35.04 | troyb1 | asterboy your all wrong :P |
| 19:35.11 | asterboy | lol |
| 19:35.17 | troyb1 | teabagging cant be expressed in numbers.. |
| 19:35.36 | asterboy | it could be used in context though |
| 19:35.38 | salviadud | we might of gone too far with the teabagging, hehehe |
| 19:35.38 | asterboy | :P |
| 19:35.48 | kpettit | <PROTECTED> |
| 19:36.19 | justinu | wussies |
| 19:36.22 | asterboy | [][]==D |
| 19:36.42 | asterboy | brokeback asterisk |
| 19:37.42 | salviadud | hehe, that's funny |
| 19:37.56 | troyb1 | bbl |
| 19:38.03 | justinu | kodaachroome... gives us the nice brighter colors, gives us the greens of summers... makes you think all the world's a sunny day... o/~ |
| 19:38.03 | salviadud | if we could actually get some real asterisk work done here, that be amazing |
| 19:38.10 | justinu | you gotta love a guy who writes a song about film |
| 19:38.20 | salviadud | no wonder the guys at digium go "we don't control the channel... they're crazy over there" |
| 19:38.33 | justinu | a lot of people have had their problems solved |
| 19:38.36 | justinu | by me, and others |
| 19:38.37 | justinu | :P |
| 19:38.40 | salviadud | i agree |
| 19:38.49 | justinu | we deserve a teabagging break now and then |
| 19:38.56 | salviadud | yet, the occasional madness, gotta love it |
| 19:39.00 | justinu | heh |
| 19:39.08 | asterboy | I can call ou/in on my setup, but when I call in, no voice. What are some of the conditions that cause that? Polycom SIP --> * --->Digium FXO |
| 19:39.30 | justinu | nat settings, |
| 19:39.34 | justinu | canreinvite settings |
| 19:39.36 | justinu | codec settings |
| 19:39.38 | asterboy | ah |
| 19:39.40 | asterboy | NAT |
| 19:39.44 | rharfield | Anyone around today familiar with app_rpt? Having a keying problem. |
| 19:40.05 | asterboy | canreinvite should be "yes"? |
| 19:40.14 | justinu | it should be no, initially |
| 19:40.24 | asterboy | checking... |
| 19:40.35 | *** join/#asterisk Cyphon (n=Cyphon@ip68-225-173-236.om.om.cox.net) |
| 19:40.41 | Nivex | rharfield: sadly, no. I only recently discovered it myself. |
| 19:40.54 | Cyphon | How do I set the smtp server for voicemail? |
| 19:41.09 | hypa7ia | troyb1: whut :p |
| 19:41.32 | troyb1 | have you been following the conversation :) |
| 19:41.41 | rharfield | Nivex: S'okay, hard to find people to talk to because it isn't widespread yet. |
| 19:41.42 | kpettit | Cyphon, setup postfix, exim, sendmail, courier or whatever else you want to send the mail |
| 19:41.52 | troyb1 | actually hypa7ia im going to get on MSN i wanna talk to you :P |
| 19:42.11 | justinu | i know that if katty were here, she wouldn't like that topic ;) |
| 19:42.54 | *** join/#asterisk heison (n=heison@216.235.9.2) |
| 19:43.00 | *** join/#asterisk Andr3w_ (n=Andrew@stjhnf0122w-142162062153.pppoe-dynamic.nl.aliant.net) |
| 19:43.05 | Andr3w_ | hello |
| 19:43.07 | *** join/#asterisk chapeaurouge (n=chap@user-85-201-82-146.tvcablenet.be) |
| 19:44.20 | heison | hello |
| 19:44.41 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 19:45.45 | heison | Trying to checkout 1.2 via SVN and I get 400 Bad Request |
| 19:45.57 | heison | svn checkout http://svn.digium.com/svn/asterisk/branches/1.2 asterisk-1.2 |
| 19:45.57 | heison | svn: REPORT request failed on '/svn/asterisk/!svn/vcc/default' |
| 19:45.57 | heison | svn: REPORT of '/svn/asterisk/!svn/vcc/default': 400 Bad Request (http://svn.digium.com) |
| 19:46.21 | heison | any idea why? |
| 19:46.49 | ebag | asterisk is the coolest! |
| 19:47.53 | asterboy | justinu, what NAT parameters need to be setup to make sure inbound gets voice? I have NAT=yes and externalIP/LocalIP set, at the router the box is in DMZ |
| 19:48.09 | asterboy | canreinvite=no |
| 19:48.11 | Zodiacal | anyone know if asterisk has an extention to transfer directly to a users voicemail? |
| 19:48.28 | Zodiacal | i know #70 transfers |
| 19:48.32 | Lino` | hmmm |
| 19:48.32 | Zodiacal | but it rings first |
| 19:48.37 | justinu | asterboy: asterisk is in the DMZ? and the phone? |
| 19:48.38 | Lino` | asterisk@home has *NUMBER |
| 19:48.49 | asterboy | ah, no the phone is not. |
| 19:48.55 | [TK]D-Fender | asterboy : pastebin your [general] and phone contexts from sip.conf |
| 19:48.58 | justinu | phone is behind a different NAT? |
| 19:49.06 | Zodiacal | lino thank you! |
| 19:49.12 | asterboy | pasting.... |
| 19:49.15 | *** join/#asterisk Pr0nL0rd (n=kvirc@cnq47-59.cablevision.qc.ca) |
| 19:49.26 | Lino` | does ist work @ Zodiacal ? |
| 19:49.33 | Lino` | -s |
| 19:49.51 | Lino` | so now I have a problem with mISDN |
| 19:49.53 | Zodiacal | yeah it works |
| 19:49.58 | Lino` | very good |
| 19:49.59 | Pr0nL0rd | Hi Guy, I need some in help..I have 3 centrex line for inbound/outboud and I'm not able to configure more that one simultaneous outgoing call.. |
| 19:50.22 | Lino` | 2 HFC cards, 2 Fritz! cards, both seem to be installed but they dont work. when i do misdn show channel 1 or 2 or 3 or 4 |
| 19:50.24 | Lino` | it just goes |
| 19:50.33 | Lino` | * Stack Addr:40000401 Port 4 Type NT Prot. PMP L2Link DOWN L1Link:DOWN |
| 19:50.52 | justinu | sounds like no d-channel, or no 2b1q up |
| 19:51.03 | justinu | L1 link would be 2b1q, i think |
| 19:51.04 | *** part/#asterisk Pr0nL0rd (n=kvirc@cnq47-59.cablevision.qc.ca) |
| 19:51.05 | justinu | L2 would be HDLC |
| 19:51.05 | Lino` | hmmm |
| 19:51.09 | Lino` | sick |
| 19:51.15 | *** join/#asterisk Arkys (n=kvirc@cnq47-59.cablevision.qc.ca) |
| 19:51.17 | Lino` | the card is connected to a working s0 bus |
| 19:51.19 | Zodiacal | lino acctualy it doens't really work.. it calls a voicemail, it doesn't allow transfering to voicemail during a call |
| 19:51.25 | Lino` | hmmm |
| 19:51.28 | Lino` | crap |
| 19:51.30 | Arkys | Sorry about my nickname, this is not my computer :S |
| 19:51.33 | Zodiacal | lino close :) |
| 19:51.34 | justinu | bri is kinda funky for us u-laws |
| 19:51.40 | justinu | we don't know it well |
| 19:51.45 | Lino` | ;) |
| 19:51.53 | Arkys | anyone can help for multiple outbound call ? |
| 19:51.56 | Lino` | well this is germany, everybody has BRI |
| 19:51.59 | justinu | yeah |
| 19:52.03 | Lino` | almost everybody :) |
| 19:52.04 | Zodiacal | lino, ok if i transfer them to *X it works! |
| 19:52.05 | justinu | i've seen it, used it |
| 19:52.07 | justinu | but not in a while |
| 19:52.08 | Zodiacal | lino thanks again! |
| 19:52.09 | Lino` | yeah |
| 19:52.14 | Lino` | thats what its meant to be |
| 19:52.26 | Lino` | i dont get the transfer stuff anyway too stupid for it ;) |
| 19:53.25 | Lino` | np @ Zodiacal |
| 19:53.30 | Lino` | ~seen possible |
| 19:53.43 | jbot | possible <n=Babbel@23.255-136-217.adsl-fix.skynet.be> was last seen on IRC in channel #asterisk, 5d 7h 34m 22s ago, saying: 'I guess not'. |
| 19:53.43 | justinu | anyone using opera as a web browser? |
| 19:54.09 | *** join/#asterisk shido6 (n=shido6@d221-68-217.commercial.cgocable.net) |
| 19:54.52 | Arkys | :( |
| 19:54.57 | *** join/#asterisk DaveHope (n=dave@62.69.60.24) |
| 19:55.47 | *** join/#asterisk PakiPenguin (n=uppal@linuxpakistan/admin/pakipenguin) |
| 19:56.03 | *** join/#asterisk eldu (i=Duncan@37.231.99-84.rev.gaoland.net) |
| 19:56.04 | Zodiacal | lino one other issue :), it says "user is on the phone, Please leave a message", they arn't on the phone tho |
| 19:56.08 | eldu | hello |
| 19:56.17 | Zodiacal | know of a way to change that to the unavalable message? |
| 19:57.05 | Zodiacal | lino i think i have another issue thats causing this |
| 19:57.08 | Zodiacal | thanks i'll go play! |
| 19:58.05 | Darwin35 | boy this is a pain |
| 19:58.08 | *** join/#asterisk zotz (n=zotz@24.231.32.85) |
| 19:58.12 | *** join/#asterisk eric_hill (i=EricHill@204.94.175.11) |
| 19:58.15 | Darwin35 | thw wiki realtime page sucks |
| 19:59.12 | *** part/#asterisk mxmasster (n=mxmasste@ppp-71-138-117-215.dsl.irvnca.pacbell.net) |
| 19:59.55 | *** join/#asterisk X-Gen (n=x-gen@dsl-145-238-131.telkomadsl.co.za) |
| 20:00.13 | Octothorpe | Darwin35, so submit changes and help us all |
| 20:00.29 | Darwin35 | I am just learning realtime |
| 20:00.35 | Darwin35 | or else i would |
| 20:00.49 | *** join/#asterisk willt (i=willt@wifi-napanet-static-206-81-99-68.napanet.net) |
| 20:01.08 | Darwin35 | and it seems most people here dont run real time |
| 20:01.32 | asterboy | http://pastebin.ca/45839 |
| 20:01.44 | Arkys | Let me explain my problem; I have 3 centrex line hooked on 3 Zap port, I have configured 3 diff trunk and if I add one trunk to outbound routing it work but if I configure 2 trunks the second one receive a fast busy tone when the first one is in use |
| 20:02.03 | asterboy | no hacking my port 127.0.0.1 now. |
| 20:03.26 | *** part/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 20:03.48 | *** join/#asterisk Darwin35 (n=darwin35@sta-208-139-193-162.rockynet.com) |
| 20:03.56 | *** join/#asterisk jpablo (n=jpablo@200.94.130.194) |
| 20:04.48 | jpablo | hey people. how can i override the asterisk voicemail default unavailable messages, but with just one file i don't want it saying: the extension BLA BLA BLA BLA is unavaible. i want to remplace that with just one voice file. |
| 20:04.57 | Arkys | anyone know what is my problem ? |
| 20:05.34 | Darwin35 | arkys yes you touche asterisk |
| 20:05.39 | Arkys | ;) |
| 20:05.42 | Arkys | really funny |
| 20:05.50 | Abydos313 | heh |
| 20:06.14 | Darwin35 | asterisk is for hose who have extensive pbx knowledge and yes of pbx experiances |
| 20:06.39 | Darwin35 | those who have served inthe telecom world and survived |
| 20:06.43 | Arkys | Darwin35: I have it.. |
| 20:06.44 | Abydos313 | i guess this would be boring if it all worked |
| 20:06.55 | jpablo | Darwin35:that's so not true |
| 20:06.59 | Darwin35 | ark whaats the issu |
| 20:07.28 | jpablo | Darwin35, I would said asterisk is for people with a clue. no matter what background they have |
| 20:07.56 | eldu | is there a way to find the origin of the jitter ? eg: i have a constant jitter on my * output. |
| 20:08.10 | austinnichols101 | I'll agree that it's easier if you know how to think and talk like a telco guy |
| 20:08.48 | AlexCTI | Someone can recomend me a very good IAX2 softphone? |
| 20:08.54 | asterboy | [TK]D-Fender: Look ok? http://pastebin.ca/45839 |
| 20:09.03 | *** join/#asterisk ohad (n=ohad@19-231-13-72.cosmoweb.net) |
| 20:09.18 | ohad | hi, how do i get asterisk to send me an email each time i get a voicemail? |
| 20:09.58 | Darwin35 | to find the origion of jitter please referance the hitch hikers guide to the galaixy |
| 20:10.42 | Arkys | Darwin35: I'm able to outbound one 1 line but I'm not able to place more that 1 simultaneous call |
| 20:10.43 | Darwin35 | page number 4thousand 6hundered4ort8ght |
| 20:10.50 | Arkys | Darwin35: I have 3 centrex line |
| 20:10.56 | eldu | hehe Darwin35 |
| 20:11.10 | eldu | the answer should be 42 :) |
| 20:11.20 | jpablo | ohad, read the wiki in voicemail.conf |
| 20:11.21 | Darwin35 | heheh |
| 20:11.24 | Darwin35 | your right |
| 20:11.38 | eldu | but that doesnt match my f** jitter |
| 20:11.51 | *** join/#asterisk mcnobody (n=laaksola@laaksola.net) |
| 20:11.54 | eldu | im very disapointed at this point |
| 20:11.54 | Arkys | Darwin35: If I put 3 trunk in my trunk sequence the second line have a fast busy tone |
| 20:12.05 | ohad | jpablo, i have. my email is there.. but for some reason asterisk doesn't forward it to sendmail |
| 20:12.11 | eldu | i spent more than one day capturing and anlysing rtp |
| 20:12.21 | eldu | with no luck |
| 20:12.30 | Darwin35 | is this asterisk or asterisk@home ? |
| 20:12.31 | ohad | jpablo, + i left myself 5 vm's and i when i called 8500 and got only one.. |
| 20:12.34 | noky | hi |
| 20:12.38 | noky | i have a question |
| 20:13.29 | noky | i'm using ooh323 in my asterisk... i want that set "no register e164" like my gateway h323 voip... how can i set that ? |
| 20:13.47 | Darwin35 | for outbound I fine 90% of th calls fail due to incorrect dial string |
| 20:13.57 | noky | because in ooh323.conf sample appears something like e164=100 |
| 20:13.59 | Darwin35 | context/phrasing |
| 20:14.45 | Arkys | Darwin35: yes but If I just add 2 more trunk to the outbound routing if should work or not ? |
| 20:15.40 | jpablo | ohad, is your sendmail config working ? |
| 20:16.04 | Darwin35 | brb fone |
| 20:17.11 | ohad | jpablo, yes. now it seems that vm is not acutally working immidiatly .. it takes a really long time if at all the report that a user has a vm |
| 20:17.28 | jpablo | humm |
| 20:17.34 | ohad | i left myself 8 vm's and only one was reported |
| 20:17.45 | jpablo | no clue vm always had worked fine for me |
| 20:18.14 | eldu | Darwin35: do u have any hints to point me out ? |
| 20:19.05 | *** join/#asterisk backblue (n=moo@87-196-66-128.net.novis.pt) |
| 20:20.42 | jpablo | grrr |
| 20:21.30 | Skarmeth | what's a cost-effective ATA for connecting a Fax machine (send/receive)? |
| 20:21.32 | AlexCTI | Hi, I have a X-ten PRO, and g729 on my server, but it connect all the time with g711, how can i make that it choose g729? |
| 20:21.35 | justinu | austinnichols101: what's new in 8.2? |
| 20:21.50 | justinu | alexcti: disallow=all; allow=g729; |
| 20:22.29 | noky | i'm using ooh323 in my asterisk... i want that set "no register e164" like my gateway h323 voip... how can i set that ? |
| 20:22.54 | *** join/#asterisk jsharp (n=jsharp@65.88.255.245) |
| 20:23.13 | AlexCTI | Justinu, i did that, but it doesnt make the call, |
| 20:23.33 | justinu | then theres a problem somewhere |
| 20:23.52 | justinu | what happens when you type "show translation" at the CLI? |
| 20:24.01 | justinu | do you see numbers in the g729 column? |
| 20:24.32 | AlexCTI | let tell you.. hold on |
| 20:26.32 | *** join/#asterisk epablo (n=epablo@201.242.75.16) |
| 20:27.02 | AlexCTI | Justinu: The column g729 show me in almost all rows numbers |
| 20:27.11 | justinu | ok, good |
| 20:27.22 | Arkys | Darwin35: no clue ?? |
| 20:27.34 | AlexCTI | justinu: What that is means? |
| 20:27.45 | justinu | it means your g729 codec is loaded |
| 20:28.04 | AlexCTI | oh yeah,... I put 60 of them |
| 20:28.10 | justinu | alexcit: sip debug peer <phone> |
| 20:28.20 | justinu | make a call, paste the ENTIRE output from the INVITE to the BYE |
| 20:29.18 | Darwin35 | brb server room calls |
| 20:32.20 | epablo | Hi guys.. I was looking at DaveHope's asterisk on debian.. and the info comming out on the CLI is weird.. or incomplete.. I can't see what the calls are trying to do.. like Dial... anyone know what i have to do to set this up? |
| 20:32.38 | SpaceBass | my dring problem is quite complex... 2 zap lines: zap/1 is personal zap/2 is business and has a dring.... everything works fine in terms of routing until someone calls zap/2 then ALL calls come in as if from zap/2....if I comment out the dring, it works fine...personal calls to person extens, work calls get a cID prefix and go to work extensions |
| 20:32.40 | epablo | Normally i use sample scripts as base and it's enought |
| 20:33.31 | AlexCTI | justinu, I got this: chan_sip.c:3588 process_sdp: No compatible codecs! |
| 20:34.25 | justinu | yeah, that's an issue |
| 20:34.30 | *** join/#asterisk bails (n=bails@bailsyatton.plus.com) |
| 20:34.39 | justinu | we'll need to see the INVITE/200 OK exchange to figure out why that happens |
| 20:34.54 | justinu | perhaps g729 isn't enabled on Xlite-PRO? |
| 20:35.00 | *** part/#asterisk epablo (n=epablo@201.242.75.16) |
| 20:36.57 | AlexCTI | On the X-PRO it shows that option, but if i enable the ulaw, it always takes g711, but in the X-PRO screen it show g279. |
| 20:37.24 | bails | hi all i have a problem and a question, whats the correct sync output for an e1 card with span=1,0,0 because i gat intrnally clocked whatever setting i specify? |
| 20:38.38 | bails | sorry thats from zttool |
| 20:38.38 | *** join/#asterisk heison (n=heison@216.235.9.2) |
| 20:40.07 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 20:43.06 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:44.39 | AlexCTI | justinu: If I have g729 licenses on my server, Do I need g729 licenses con Clients? |
| 20:45.33 | justinu | alexcti: the clients should be licensed already... that's part of what you paid for when you bought x-pro |
| 20:46.37 | AlexCTI | justinu: So in that order of ideas I need lic on both sides, right? |
| 20:47.07 | justinu | no |
| 20:47.26 | justinu | the client is fine |
| 20:48.02 | AlexCTI | in my case, with the lic on the server, which is the best way to handle? |
| 20:48.16 | austinnichols101 | k - 7960 8.2 firmware in. Who needs something tested? |
| 20:48.40 | justinu | alexcti: send me the sip debug I requested |
| 20:48.46 | *** join/#asterisk backblue (n=moo@87-196-66-128.net.novis.pt) |
| 20:48.49 | AlexCTI | oki |
| 20:49.32 | royk_train | evening |
| 20:53.27 | AlexCTI | Justinu: http://pastebin.com/604160 |
| 20:53.57 | *** join/#asterisk sharp (n=sharp@c-68-45-160-72.hsd1.pa.comcast.net) |
| 20:54.09 | justinu | AlexCTI: right off the bat, your X-Pro client is not wanting to talk g729 |
| 20:54.17 | sharp | i want to use linphone (SIP) to listen to the demo |
| 20:54.19 | justinu | so the issue is with the client right now. |
| 20:54.46 | AlexCTI | ok.. |
| 20:55.17 | sharp | i set the default context to demo in sip.conf |
| 20:55.22 | sharp | it doesn't work |
| 20:55.45 | sharp | linphone says user cannot be found at given address |
| 20:55.53 | sharp | (sip:192.168.1.5) |
| 20:56.08 | AlexCTI | so, if I already have the Lic on server, what client should I use for not pay doble lic? |
| 20:56.43 | sharp | http://pastebin.com/604166 |
| 20:56.44 | *** join/#asterisk [hC] (n=hardcore@mail.metrobridge.com) |
| 20:58.12 | justinu | alexCTI: sorry, we're having a language barrier here... afaik X-Pro supports G729, with no extra cost. |
| 20:58.23 | justinu | AlexCTI: it sounds like a configuration issue with X-Pro |
| 21:02.59 | shido6 | so what if the client supports g729 if you do any fiddling with the audio other than passthrough you WILL need a license |
| 21:03.10 | shido6 | on the ast box |
| 21:03.22 | justinu | he's got licenses |
| 21:03.27 | jarrod | dude why is it when i make sip calls from polycom thru asterisk i hear this '*psss* .... *psss*' |
| 21:03.29 | shido6 | then he's good to go. |
| 21:03.37 | justinu | it's still not working |
| 21:03.39 | shido6 | your ast box has gas :) |
| 21:03.43 | jarrod | ha |
| 21:03.47 | jarrod | seriously... |
| 21:03.58 | shido6 | unload your codec and reload it what happens then? |
| 21:04.09 | jarrod | all g711 |
| 21:04.09 | shido6 | hell |
| 21:04.13 | jarrod | its just a pssh |
| 21:04.14 | shido6 | unload all the modules you DONT need |
| 21:04.16 | jarrod | pssh |
| 21:04.22 | jarrod | PSSSSSSSSSH |
| 21:04.39 | shido6 | corrosion on the rj11 ? :) |
| 21:04.48 | AlexCTI | is not.. i switch the order of codecs but nothing, looks like is not enable internally |
| 21:05.00 | shido6 | disallow=all (screw you!) |
| 21:05.09 | shido6 | allow=g729 ( I only want this, dangit) |
| 21:07.08 | justinu | alexcti: well... i can assure you that eyebeam supports G729. but I have never run X-Pro |
| 21:07.09 | jarrod | there is no rj11 |
| 21:07.30 | fu3 | I had a bunch of 66 blocks go bad once.. |
| 21:07.37 | fu3 | no idea why, I figured they must have corroded |
| 21:07.38 | fu3 | somehow. |
| 21:08.12 | *** join/#asterisk Octothorpe (n=octothor@unaffiliated/octothorpe) |
| 21:08.41 | *** join/#asterisk ST-3 (i=ser@dipsy.tch.org) |
| 21:09.26 | justinu | hard to imagine a 66 block going bad |
| 21:09.35 | justinu | it's basically an inanimate object |
| 21:09.42 | ST-3 | the pins spread out |
| 21:09.46 | ST-3 | and stop making contact |
| 21:09.54 | ST-3 | (sorry i know im jumping into the middle of something) |
| 21:10.03 | justinu | that's what IRC is for ;) |
| 21:10.06 | ST-3 | lol |
| 21:10.07 | ST-3 | true |
| 21:10.08 | justinu | you can rebend them together |
| 21:10.17 | ST-3 | needle nose plyers |
| 21:10.17 | ST-3 | word |
| 21:10.41 | ST-3 | I can't say i've never done that.... |
| 21:11.18 | *** join/#asterisk asterboy (n=kevin@S010600485480f4be.ed.shawcable.net) |
| 21:11.35 | asterboy | irssi rules! |
| 21:11.41 | russellb | indeed |
| 21:11.49 | asterboy | Ok, so I changed my config. |
| 21:11.59 | asterboy | No router, just a hub. |
| 21:12.01 | asterboy | No NAT |
| 21:12.12 | asterboy | can phone out no problem. |
| 21:12.21 | asterboy | Can phone in, BUT... |
| 21:12.28 | asterboy | no voice! |
| 21:12.35 | asterboy | what gives? |
| 21:12.51 | asterboy | Need some serious help on this ASAP. |
| 21:13.02 | asterboy | I'm presenting this box as a demo. |
| 21:13.25 | asterboy | Won't be selling * if you can't call in. |
| 21:13.34 | Hmmhesays | asterboy: how much you gonna pay me |
| 21:13.45 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 21:13.48 | asterboy | pizza? |
| 21:13.58 | Hmmhesays | $75/hour |
| 21:14.03 | Hmmhesays | i'll fix whatever you broke |
| 21:14.06 | justinu | that's a good deal, you should take it :) |
| 21:14.07 | glazzier | hello all. |
| 21:14.12 | russellb | i'll do it for $74.50 |
| 21:14.14 | asterboy | at least spit down the crack |
| 21:14.17 | justinu | lol, you whore |
| 21:14.27 | Hmmhesays | throw a dead hooker in there too |
| 21:14.31 | asterboy | lol |
| 21:14.46 | russellb | jk, I won't do it at all :-p |
| 21:14.55 | Hmmhesays | asterboy whats your prolem mang |
| 21:15.11 | glazzier | I have RTFM untill I am blue in the face... any one understand the AgentCallbackLogin() ? |
| 21:15.15 | asterboy | not sure, taken NAT out of the equation |
| 21:15.15 | russellb | have you verified that the voice traffic is actually flowing from the phone? |
| 21:15.24 | asterboy | ztmonitor? |
| 21:15.28 | Hmmhesays | glazzier i've used it a few times |
| 21:15.36 | russellb | and that the server is actually getting it? |
| 21:16.11 | asterboy | when I call in, the Polycom lights up and I can answer. |
| 21:16.19 | asterboy | CLI shows the call |
| 21:16.27 | jarrod | dang this *pssst* |
| 21:16.29 | jarrod | *pssst* |
| 21:16.31 | asterboy | checking with ztmonitor |
| 21:16.40 | glazzier | Hmmhesays, It keeps saying "Extension ... is not vaild for automatic login" |
| 21:16.51 | glazzier | any guesses? |
| 21:17.31 | Hmmhesays | pastebin that extensions.conf |
| 21:17.37 | Hmmhesays | that part of it |
| 21:18.24 | glazzier | Hmmhesays, ok just a sec. |
| 21:18.35 | asterboy | Interesting, ztmonitor shows TX from the SIP phone voice, but no RX or voice from the calling phone. |
| 21:19.23 | *** join/#asterisk mujjoo (n=murtazaj@h94s217a102n47.user.nortelnetworks.com) |
| 21:19.25 | mujjoo | hello all |
| 21:19.31 | justinu | asterboy: you're receiving your call on a zap interface |
| 21:19.32 | justinu | ? |
| 21:19.41 | mujjoo | i have a timeout question |
| 21:20.00 | asterboy | correct |
| 21:20.19 | mujjoo | when I dial out over my SIP trunk the call hangs up after about 4 rings |
| 21:20.37 | asterboy | just keyed in on this: Spawn extension (clone, s, 3) exited non-zero 'Zap/1-1' |
| 21:20.50 | asterboy | don't think I have something setup right in extensions.conf |
| 21:20.58 | mujjoo | what timeout value do i need to adjust |
| 21:21.26 | russellb | mujjoo: the timeout argument to Dial() |
| 21:21.31 | russellb | mujjoo: show application dial |
| 21:21.41 | *** join/#asterisk backblue (n=moo@87-196-66-128.net.novis.pt) |
| 21:22.38 | justinu | asterboy: if you're not seeing RX audio... talk to the telco! |
| 21:22.45 | justinu | because the problem isn't in your PBX |
| 21:22.45 | glazzier | exten => 8212,1,AgentCallbackLogin(212,s,SIP/jgladen) |
| 21:22.45 | glazzier | exten => 8212,n,Playback(beep) |
| 21:22.45 | glazzier | exten => 8212,n,Hangup() |
| 21:22.46 | glazzier | exten => 8212,1,AgentCallbackLogin(212,s,SIP/jgladen) |
| 21:22.46 | glazzier | exten => 8212,n,Playback(beep) |
| 21:22.46 | glazzier | exten => 8212,n,Hangup() |
| 21:22.48 | glazzier | exten => 8212,1,AgentCallbackLogin(212,s,SIP/jgladen) |
| 21:22.49 | Hmmhesays | whoa |
| 21:22.50 | glazzier | exten => 8212,n,Playback(beep) |
| 21:22.52 | glazzier | exten => 8212,n,Hangup() |
| 21:22.52 | Hmmhesays | ~pb |
| 21:22.53 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:22.54 | glazzier | exten => 8212,1,AgentCallbackLogin(212,s,SIP/jgladen) |
| 21:22.56 | glazzier | exten => 8212,n,Playback(beep) |
| 21:22.58 | glazzier | exten => 8212,n,Hangup() |
| 21:23.00 | glazzier | exten => 8212,1,AgentCallbackLogin(212,s,SIP/jgladen) |
| 21:23.00 | *** kick/#asterisk [glazzier!n=russell@asterisk/developer-and-stable-maintainer/drumkilla] by russellb (russellb) |
| 21:23.04 | *** join/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 21:23.04 | Hmmhesays | haha |
| 21:23.12 | Hmmhesays | i wouldn't try that again |
| 21:23.15 | Hmmhesays | ~pb |
| 21:23.16 | jbot | pb is probably a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:23.20 | mujjoo | do i set this in the extension.conf |
| 21:23.24 | russellb | mujjoo: yes |
| 21:23.25 | glazzier | my bad |
| 21:23.26 | justinu | (13:20:43) justinu: asterboy: if you're not seeing RX audio... talk to the telco! |
| 21:23.26 | justinu | (13:20:51) justinu: because the problem isn't in your PBX |
| 21:23.31 | mujjoo | ok |
| 21:24.38 | Hmmhesays | glazzier: that won't work out too hot for you |
| 21:25.26 | Hmmhesays | paypal me a $50 and i'll fix it for you |
| 21:26.27 | justinu | down to 50? |
| 21:26.30 | justinu | damn whores |
| 21:27.25 | *** join/#asterisk sivana_ (n=sivana@mixdown.ca) |
| 21:27.32 | Hmmhesays | hell i'd probably do it for $30 |
| 21:27.36 | Hmmhesays | i need some beer money |
| 21:27.41 | Hmmhesays | and batteries for my monitor |
| 21:27.48 | sivana_ | if I have slin audio files and I playback through a PRI.. is there transcoding? |
| 21:27.54 | asterboy | justinu, I can call the line fine outside of * |
| 21:28.05 | justinu | then your zap card is messed up |
| 21:28.07 | justinu | somehow |
| 21:28.21 | justinu | what is it? |
| 21:28.22 | asterboy | hmmm |
| 21:29.43 | asterboy | I've been trying the calls on a clone and on a Wildcard X101P |
| 21:29.44 | mujjoo | russellb : so if I am using exten => _NXXNXXXXXX,1,Macro(dialout-trunk,2,${EXTEN},) |
| 21:29.51 | mujjoo | I need to modify the 2? |
| 21:29.55 | asterboy | both have same issue. |
| 21:29.58 | justinu | ok |
| 21:30.03 | justinu | well... that's just weird. |
| 21:30.11 | justinu | not sure what else to tell you |
| 21:30.17 | asterboy | thats what throwing a wrench into things. |
| 21:30.18 | justinu | problem with zaptel drivers? |
| 21:30.28 | justinu | try recompiling and re-installing, perhaps |
| 21:30.57 | asterboy | Could be I'm using VOIP ---> Analog ---> Zap |
| 21:30.57 | russellb | mujjoo: I have no idea. I'd have to see the macro ... |
| 21:31.06 | mujjoo | where can i post the macro |
| 21:31.06 | justinu | voip->analog->zap? |
| 21:31.10 | russellb | ~pb |
| 21:31.11 | jbot | i guess pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 21:31.11 | asterboy | maybe that won't be the case when I connect to real pots |
| 21:31.12 | mujjoo | i dont want to flood here |
| 21:32.36 | mujjoo | russellb : pasted it there |
| 21:32.42 | justinu | you need to paste the link here now |
| 21:32.53 | russellb | i have to go now, but surely someone else here can help you. |
| 21:32.59 | *** join/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 21:33.35 | mujjoo | http://pastebin.com/604270 |
| 21:34.29 | glazzier | Thanks Hmmhesays. your low bid with no forethought told me enough.. |
| 21:35.04 | markit | hi, I've iax that has as context "incoming-prova", but when I receive the call with iax, I have this message |
| 21:35.12 | markit | *CLI> Mar 15 22:30:07 NOTICE[29265]: chan_iax2.c:6811 socket_read: Rejected connect attempt from 81.174.34.132, request '723@incoming-prova' does not exist |
| 21:35.40 | markit | [incoming-prova] |
| 21:35.40 | markit | exten => s,1,Dial(${I_UFFICIO_IN_RING},15,t) |
| 21:35.51 | markit | any idea? |
| 21:36.04 | markit | (the latter 2 lines are from extensions.conf, of course) |
| 21:36.10 | mujjoo | justinu: do you mind taking a look at the macro |
| 21:37.07 | *** join/#asterisk Dr-Linux (n=nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 21:37.19 | Dr-Linux | hi all |
| 21:37.26 | justinu | hello Dr. |
| 21:37.33 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 21:37.50 | Dr-Linux | hey, Justin how are you friend? :) |
| 21:38.01 | justinu | ok, you? |
| 21:38.41 | lemmy | has anybody managed to use the callthrough function of a fritz!box 7170 from within *? |
| 21:39.04 | [TK]D-Fender | markit : "s" isn't what its looking for. its looking for *723* |
| 21:39.10 | lemmy | i can't figure it out. %) |
| 21:40.31 | glazzier | exit |
| 21:40.34 | *** part/#asterisk glazzier (n=glazzier@c-67-181-136-109.hsd1.ca.comcast.net) |
| 21:40.59 | markit | [TK]D-Fender: but shouldn't 's' be the starting poing without the need of a match? |
| 21:42.23 | [TK]D-Fender | markit : "s" is only used by analog interfaces for which a target number is not known. |
| 21:42.54 | [TK]D-Fender | markit : if you want to match all *NUMBERS*, you'd do something like exten =>_x.,1,Whatever() |
| 21:43.07 | markit | [TK]D-Fender: mmm is it a recent change in asterisk? |
| 21:43.12 | shmaltz | markit, i would be in your case, or you could use a wildcard like _X. |
| 21:43.33 | shmaltz | markit, not it's always been that way |
| 21:43.39 | markit | shmaltz: well, I had the feeling that it worked this way in the past, but I could be wrong |
| 21:43.48 | markit | ok, I surrender ;) |
| 21:44.04 | justinu | any java coders around? |
| 21:44.07 | markit | thanks a lot, maybe I've misunderstood some doc or sample |
| 21:44.18 | *** join/#asterisk Prival (i=user75@Sherbrooke-HSE-ppp3610823.sympatico.ca) |
| 21:44.19 | markit | [TK]D-Fender: , shmaltz thanks a lot |
| 21:44.28 | shmaltz | np |
| 21:44.36 | shmaltz | justinu, I'm sure |
| 21:44.41 | shmaltz | but not me |
| 21:44.42 | shmaltz | :( |
| 21:44.47 | sivana_ | markit: in macros you can use 's' |
| 21:44.55 | Prival | Hi, I'm running * 1.0.10 and we have very low volume on voicemail. Any hints to increase the voicemail volume? |
| 21:44.58 | justinu | i'm trying to figure out why ArrayList.add() would block |
| 21:45.01 | justinu | driving me nuts |
| 21:45.31 | shmaltz | Prival, get more people to call you and leave messages |
| 21:45.33 | shmaltz | :) |
| 21:45.58 | Prival | shmaltz :-P I'm talking about the audio volume.... |
| 21:46.09 | shmaltz | Prival, oh, thanks for explaining |
| 21:46.13 | shmaltz | ;) |
| 21:46.18 | justinu | 2400 Euro for asterisk boot camp? |
| 21:46.22 | justinu | 2500, i mean |
| 21:46.23 | justinu | what a rip |
| 21:46.25 | lemmy | justinu: what do you mean with "block". by default an arraylist isn't synchronized. |
| 21:46.27 | Hmmhesays | sounds like a rip off |
| 21:46.33 | justinu | lemmy: i mean that it doesn't return! |
| 21:46.35 | *** join/#asterisk Dr-Linux (n=nothing@host202-147-168-130.lhr.dancom.net.pk) |
| 21:46.36 | shmaltz | justinu, you can do what I did |
| 21:46.50 | justinu | i could probably teach asterisk boot camp myself |
| 21:46.50 | lemmy | justinu: multithreaded enviroment? |
| 21:46.55 | Hmmhesays | you cannot substitute a wad of cash for good old fashioned experience |
| 21:46.55 | Dr-Linux | my fuckin PC gets hanged :@ |
| 21:46.56 | justinu | lemmy: yeah, threads happening |
| 21:46.59 | shmaltz | justinu, exactly |
| 21:47.18 | lemmy | justinu: and this arraylist is synchronized? |
| 21:47.28 | justinu | no |
| 21:47.29 | Hmmhesays | the first time someone asked how to install it on windows i'd probably piss on their boots |
| 21:47.30 | shmaltz | Hmmhesays, all the money in the world won't buy anyone any will or expeience |
| 21:47.34 | justinu | but nothing else should be be touching it |
| 21:47.37 | Hmmhesays | or intelligence |
| 21:47.55 | shmaltz | Hmmhesays, I just wont trust him to setup asterisk |
| 21:47.57 | Dr-Linux | T100P is nomore in use? |
| 21:48.06 | fu3 | Hey guys. I just got off the phone with my telco, and they were so shocked at the quality of the phone call,they called other people into the room just to hear it. They were doubting Asterisk's capabilities, but wow.. we sure showed them! |
| 21:48.16 | fu3 | I thought it was funny to hear the telco rave about line quality :) |
| 21:48.21 | jsharp | You finally got your stuff working? |
| 21:48.21 | jsharp | Sweet |
| 21:48.24 | fu3 | hell yeah man |
| 21:48.27 | fu3 | it works 100% ! |
| 21:48.28 | shmaltz | fu3, you mean the techie from the telco |
| 21:48.29 | lemmy | justinu: did you debug through the code? |
| 21:48.31 | jsharp | Righteous. |
| 21:48.40 | *** join/#asterisk x86_ (n=x86@p3m/member/x86) |
| 21:48.44 | fu3 | shmaltz.. the tech's and the service reps. |
| 21:48.49 | Hmmhesays | there is no reason voip shouldn't be clear as a bell |
| 21:48.50 | shmaltz | fu3, which telco was it anyhow? |
| 21:48.52 | fu3 | Qwest |
| 21:48.53 | justinu | lemmy: yeah, and the debugger is acting oddly too |
| 21:49.09 | justinu | something must be fucked up in my dev environment |
| 21:49.16 | [av]bani | arg, this laptop hd is so slow |
| 21:49.20 | shmaltz | Hmmhesays, only when it's all digital, and your ISP isn't playing games with your data |
| 21:49.25 | Hmmhesays | true |
| 21:49.30 | fu3 | So, to all who doubt FreeBSD as an OS for Asterisk.. IT WORKS FINE! |
| 21:49.39 | Hmmhesays | wow george lucas has agreed to write a sucktastic star wars tv show |
| 21:49.47 | jsharp | And when you're not expecting toll quality 100% availability over the intardnet. |
| 21:49.49 | justinu | fu3: that's pretty funny |
| 21:49.55 | shmaltz | fu3, you running Asterisk on freeBSD? |
| 21:49.59 | fu3 | For the record, i'm running FreeBSD 6.1-prerelease, Asterisk 1.2.5 and the 29th release of the zaptel-bsd drivers. |
| 21:50.00 | fu3 | yes.. |
| 21:50.03 | [TK]D-Fender | fu3 : * compiles on most every *nix platform, its ZAPTEL that's tricky |
| 21:50.15 | Hmmhesays | yeah but my milliwatt test over a crossover cable should be pristine |
| 21:50.16 | fu3 | yeah, well there were comments about weather or not FreeBSD could do it |
| 21:50.18 | fu3 | but it does |
| 21:50.20 | fu3 | for now! |
| 21:50.21 | fu3 | :) |
| 21:50.30 | shmaltz | fu3, and whatabout MOH? does it work? |
| 21:50.35 | fu3 | whats MOH? |
| 21:50.40 | [hC] | oh dear. |
| 21:50.40 | shmaltz | ~moh |
| 21:50.42 | jbot | extra, extra, read all about it, moh is Music On Hold |
| 21:50.46 | fu3 | ohhh |
| 21:50.46 | fu3 | sorry |
| 21:50.53 | fu3 | I havent tested that yet. |
| 21:50.58 | fu3 | But I will report once I do. |
| 21:51.01 | shmaltz | hmmmmmmm let me know |
| 21:51.03 | Hmmhesays | how are you going to play skidrow to your waiting customers without moh? |
| 21:51.04 | fu3 | I shall. |
| 21:51.14 | Skid | :o |
| 21:51.14 | fu3 | for what it's worth, I can record and playback music just fine. |
| 21:51.15 | jsharp | lords of acid. |
| 21:51.16 | Skid | highlight |
| 21:51.16 | Skid | :) |
| 21:51.22 | shmaltz | I like to play I'm sorry MaMa |
| 21:51.25 | shmaltz | ............ |
| 21:51.32 | fu3 | Im gonna play cop killah |
| 21:51.34 | fu3 | over and over :) |
| 21:51.37 | Dr-Linux | [TK]D-Fender: T100P is no more they sell TE10P for T1 single port, |
| 21:51.50 | justinu | i guess i'll update the JDK for the hell of it |
| 21:51.56 | Dr-Linux | so this port is FXO or FXS , they didn't mentioned :S |
| 21:52.02 | Dr-Linux | or works for both? :S |
| 21:52.02 | *** join/#asterisk jijgeh (i=jijgeh@0-1pool139-55.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 21:52.08 | fu3 | I'm flying to Houston and will be back on the 21st.. i'll let you know how the MOH works then. |
| 21:52.16 | [TK]D-Fender | T100P = T1 only (no E1) |
| 21:52.17 | shmaltz | fu3, you doing any transcoding? like g711 to g729? |
| 21:52.25 | fu3 | nope |
| 21:52.43 | shmaltz | fu3, are those moduels loaded though? |
| 21:52.46 | asteriskmonkey | where can i find a 2400p sample config |
| 21:52.56 | fu3 | i dont know. let me check. |
| 21:53.02 | shmaltz | asteriskmonkey, you tried the wiki? |
| 21:53.16 | *** join/#asterisk asterisk99 (n=astguy@modemcable169.194-130-66.mc.videotron.ca) |
| 21:53.18 | *** join/#asterisk nagl (n=nagl@vie-086-059-104-148.dsl.sil.at) |
| 21:53.23 | justinu | KMFDM |
| 21:53.24 | fu3 | yes, they appear to be loaded via 'show modules' |
| 21:53.33 | [TK]D-Fender | Hmmhesays : "Get a 3-piece Wall-steet smile and son you'll look just like me" :) |
| 21:53.55 | shmaltz | fu3, So I guess you did it |
| 21:53.55 | asteriskmonkey | yes couldnt find anything |
| 21:53.58 | shmaltz | congrats |
| 21:53.59 | Dr-Linux | [TK]D-Fender: but T100P is no more available, if you ask for, they will give you TE10P :S |
| 21:54.02 | fu3 | thanks. |
| 21:54.06 | Hmmhesays | nice |
| 21:54.12 | fu3 | I thank the people at FreeBSD and the ones who made the zaptel-bsd port |
| 21:54.13 | asteriskmonkey | i have a te110p for sale |
| 21:54.14 | [TK]D-Fender | Dr-Linux : Well aware of that... |
| 21:54.21 | fu3 | and of course |
| 21:54.25 | fu3 | everyone at #asterisk |
| 21:54.36 | fu3 | because I would be absoultly fucking lost up my own asshole if I couldnt come here. |
| 21:54.46 | justinu | there's an image |
| 21:54.50 | fu3 | :) |
| 21:55.05 | justinu | how'd you resolve the E&M outpulsing issues? |
| 21:55.05 | asterisk99 | Anyone have problems with a FXS module failing to startup after a boot? (with supposedly no changes to zapata/zaptel.conf) |
| 21:55.07 | [TK]D-Fender | Hmmhesays : I mastered "Youth Gone Wild", "18 and Life", and "I'll Remember You" was the first song I ever performed :) |
| 21:55.13 | Dr-Linux | [TK]D-Fender: yes, but that TE110P is has a FXO port or FXS ? :S |
| 21:55.24 | justinu | TE110P is a digital card |
| 21:55.26 | fu3 | i called my qwest service rep.. he said they werent pushing shit to me, so we switched to 7 digit outpulsing and she works! |
| 21:55.32 | justinu | it can support whatever signalling you might need |
| 21:55.41 | [TK]D-Fender | Dr-Linux : NO, they are DIGITAL cards only. |
| 21:55.42 | jsharp | So qwest was the problem all along? |
| 21:55.44 | justinu | fu3: cool, that's what I figured |
| 21:55.47 | fu3 | as usual |
| 21:55.50 | fu3 | yeah, you were right. |
| 21:55.50 | Dr-Linux | asteriskmonkey: why you wanna sell te110p ? |
| 21:55.57 | jsharp | :O |
| 21:56.07 | fu3 | Every time I suspect something, I have to jump through a million hoops to convince qwest that im right.. |
| 21:56.11 | fu3 | oh well.. it's been fun :) |
| 21:56.27 | fu3 | haha |
| 21:56.33 | asteriskmonkey | Dr-Linux: im moved to a te406.. now im moving to a 104d :) |
| 21:56.34 | justinu | yep, telcos are assholes |
| 21:56.37 | fu3 | I'm looking forward to the next four or five days off anyway.. |
| 21:56.43 | fu3 | too bad it's going to rain in Houston for my entire visit :| |
| 21:56.44 | asterisk99 | fu3: Thast cuz you are dealing with Level 1 Tech Support which is heavily scripted |
| 21:56.51 | fu3 | no, im way past them asterisk99. |
| 21:56.58 | fu3 | the guys I talk to, talk to the guys at the CO direct. |
| 21:57.05 | jsharp | We all have special places in our hearts for screwing with the phone company. |
| 21:57.09 | shmaltz | fu3, you got their number? |
| 21:57.12 | fu3 | yes |
| 21:57.14 | fu3 | i kept it :) |
| 21:57.17 | justinu | fu3: i see you figured that out too |
| 21:57.18 | *** join/#asterisk Darwin35 (n=Darwin@c-24-9-75-234.hsd1.co.comcast.net) |
| 21:57.23 | shmaltz | jsharp, you a 2600 subscriber? |
| 21:57.23 | justinu | taking my advice and talking to the switch techs direct. |
| 21:57.34 | Dr-Linux | hhm.. |
| 21:57.45 | fu3 | yep |
| 21:57.47 | shmaltz | fu3, good for you where you located? maybe I can share some NOC numbers with you? |
| 21:57.47 | fu3 | it's way better |
| 21:57.51 | jsharp | No. Got better things to do with my time. |
| 21:57.52 | asterisk99 | anyone here up on their Zaptel installation for FXO/FXS??? |
| 21:57.53 | justinu | :) |
| 21:58.02 | fu3 | shmaltz.. maybe.. msg me. |
| 21:58.19 | Dr-Linux | one of our client wants asterisk solution for his transaction IVR |
| 21:58.34 | fu3 | Yeah, its great being able to ask about specific questions and not hear "I'll have to check" but rather "yep" or "nope" direct from the switchroom :) |
| 21:58.38 | asteriskmonkey | smaltz i cant find anything for the 2400 you know where i can find some smaple confs |
| 21:58.52 | justinu | yeah, the switch techs and translations people are the ones who do all the real work |
| 21:59.06 | justinu | i usually call them, ask them to change something, and then follow it up with a change order to the reps |
| 21:59.11 | fu3 | yeah, although I can certainly understand them not wanting to be bothered by the bullshit. |
| 21:59.17 | Zodiacal | how do i transfer to an fxs channel? do i create a trunk for it? |
| 21:59.56 | *** join/#asterisk Prival2 (i=user65@Sherbrooke-HSE-ppp3610823.sympatico.ca) |
| 22:00.22 | Prival2 | Hi all, any hints to raise the voicemail audio level on 10.0.10? |
| 22:00.24 | justinu | if you're not a total fuckup, they'll be cool to you |
| 22:00.34 | shmaltz | asteriskmonkey, I'm checking |
| 22:00.37 | asterisk99 | fy3: I used to work for a subsid of Verizon... I found very few people who really understood telecom --- even engineers wearing pinkie rings |
| 22:00.47 | justinu | oh yeah... most people have no fucking clue |
| 22:00.49 | fu3 | yeah.. i dont get bitchy with them or anything, plus it's nice to actually be able to talk the talk now. |
| 22:01.14 | fu3 | yeah your right.. the engineers they sent here when I was having voltage issues were asking ME about the specs.. |
| 22:01.26 | shmaltz | asteriskmonkey, try this: |
| 22:01.28 | shmaltz | http://www.digium.com/en/supportcenter/documentation/viewdocs/TDM2400P |
| 22:01.28 | Dr-Linux | justinu: ;) |
| 22:01.52 | asteriskmonkey | yes there nothing in those about the 2400 |
| 22:02.10 | asterisk99 | I couldn't find one engineer who could tell me how a 2 telephone are able to connect duplex with only 2 wires and not 4 |
| 22:02.25 | justinu | ah |
| 22:02.31 | justinu | the hybrid network is what makes that possible |
| 22:02.52 | shmaltz | notbad: |
| 22:02.53 | shmaltz | http://news.yahoo.com/s/nm/20060315/wr_nm/congress_financial_gambling_dc;_ylt=AsqRxQhrVjwYpwOxX8.O9s_6VbIF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA-- |
| 22:03.16 | fu3 | all they will do is make better games and better illusions of gambling. |
| 22:03.26 | asterisk99 | Actually, it's simpler than that ... it's only 1 circuit... both earpieces and both mouthpieces are on one circuit |
| 22:03.34 | fu3 | like in Japan, you cant gamble for money, but you can gamble for a prize ticket, which you claim, and then sell to collect your winnings. |
| 22:04.13 | asterisk99 | Like attending a function where you have to buy tickets and use those in turn to buy beer |
| 22:04.22 | fu3 | yeah, kind of. I hate those :) |
| 22:04.33 | *** join/#asterisk AlexCTI (n=alex@pembrkfl-bellsouth-24-53-200-134.miamfl.adelphia.net) |
| 22:04.50 | justinu | asterisk99: not true... there's a hybrid network in a POTS phone that converts the 2 wire signal to 4 wire (2 to the speaker, 2 to the mic) |
| 22:04.51 | shmaltz | asteriskmonkey, it does, just make sure that you have the latest stable release of zaptel and complile it then load it, then configure zaptel.conf and zapata.conf, as you would with TDM400 just change channels to fxs/fxo as needed |
| 22:07.03 | Dr-Linux | Libpri package is neccesary for TE110P ? |
| 22:07.07 | *** join/#asterisk florz (n=florz@2001:1a50:503c:0:0:0:0:1) |
| 22:07.08 | asterisk99 | justinu: hmmmm. I was told that none of that existed back in 1871 when Dr. Bell invented (er stole) the telephone --- he used only 2 wires to connect everything |
| 22:07.22 | *** join/#asterisk RoyK (n=roy@g-001.osl255.netcom.no) |
| 22:07.27 | asteriskmonkey | shmaltz : k ) |
| 22:07.39 | justinu | asterisk99: that may be true, i'm no expert on signal propagation/transmission |
| 22:07.39 | asterboy | Is there a way to turn down the mic gain on a Polycom phone? |
| 22:07.58 | justinu | it's in the sip.cfg file somewhere |
| 22:08.04 | justinu | under the settings that says "DO NOT CHANGE THIS" |
| 22:08.12 | shmaltz | I like this one: |
| 22:08.13 | shmaltz | http://news.yahoo.com/s/nm/20060315/tc_nm/media_films_cellphones_dc;_ylt=AqS8pPXlGm4gKu413WIFKST67rEF;_ylu=X3oDMTBjMHVqMTQ4BHNlYwN5bnN1YmNhdA-- |
| 22:08.28 | shmaltz | asterboy, it's in sip.cfg just do a grep on gain |
| 22:08.43 | asterisk99 | Can anyone tell me why Asterisk would claim to be unable to register a zap channel when I can see it defined with /sbin/ztcfg ??? |
| 22:08.50 | asteriskmonkey | whats usually the error when we get /dev/zap/ctl not found |
| 22:09.08 | [TK]D-Fender | asterboy : don't think so |
| 22:09.27 | shmaltz | asterisk99, yes it it's not defined in /etc/asterisk/zapata.conf |
| 22:09.56 | asterisk99 | shmaltz: I checked that too ... it's there |
| 22:10.11 | shmaltz | asterisk99, can you pb your configs and the ztcfg -vvv output? |
| 22:10.17 | *** join/#asterisk tzafrir_laptop (n=tzafrir@bzq-88-153-166-193.red.bezeqint.net) |
| 22:10.26 | asterisk99 | shmaltz: K - gimme 2 mins |
| 22:11.10 | tzafrir_laptop | hi |
| 22:11.25 | RoyK | hi |
| 22:11.33 | shmaltz | tzafrir_laptop, any updates on FXO for ..... ? |
| 22:12.50 | powerchip | how i can loop ? |
| 22:13.36 | RoyK_train-se | i loop can rollercoster an |
| 22:13.43 | *** join/#asterisk heison (n=heison@CPE00023fd763e2-CM000a73667501.cpe.net.cable.rogers.com) |
| 22:13.51 | fu3 | everyone go here, and download all the videos, and then watch them. |
| 22:13.52 | fu3 | http://www.question911.com/links.php |
| 22:13.55 | fu3 | :) |
| 22:13.57 | Octothorpe[away] | asteriskmonkey: it is a udev problem |
| 22:13.59 | fu3 | cya next tuesday |
| 22:14.01 | justinu | roky: are you nerding out on the train? |
| 22:14.06 | justinu | royk |
| 22:14.19 | tzafrir_laptop | shmaltz, not ready yet. Generally the address is info@xorcom.com. Alternatively mail me (tzafrir.cohen@xorcom.com) if you want me to properly forward it... |
| 22:14.19 | justinu | grps sucks ass here in the US |
| 22:14.28 | asterisk99 | shmaltz: http://pastebin.com/604349 |
| 22:14.28 | fu3|gone | yes, it does. |
| 22:14.35 | justinu | gprs, that is |
| 22:14.47 | shmaltz | tzafrir_laptop, thank you |
| 22:15.18 | justinu | fu3: agreed. |
| 22:15.18 | Nivex | fu3|gone: hear hear! |
| 22:15.27 | shmaltz | fu3|gone, I'll get you a ticket with amtrack if you want |
| 22:15.27 | justinu | i need a high speed train from here to SF and Vegas |
| 22:15.29 | justinu | and pheonix |
| 22:15.38 | justinu | amtrak is a lame joke |
| 22:15.43 | fu3|gone | shmaltz.. that would be cool. |
| 22:15.56 | fu3|gone | but my flight is on friday, and everything is paid for. :) |
| 22:16.08 | shmaltz | fu3|gone, but I take a 150% markup |
| 22:16.11 | fu3|gone | hahaha |
| 22:17.03 | fu3|gone | not to go way off topic here, but has anyone else watched the CPAN reaction to the 9/11 commission report? |
| 22:17.03 | fu3|gone | wow.. |
| 22:17.28 | shmaltz | asterisk99, take out the fourth channel from zapata.conf and try it now |
| 22:17.33 | asterisk99 | shmaltz: http://pastebin.com/604360 (the error messages) |
| 22:17.38 | Nivex | Why is the Comprehensive Perl Archive Network commenting on politics? |
| 22:17.51 | *** part/#asterisk asteriskmonkey (n=phil@69.156.197.242) |
| 22:18.00 | asterisk99 | shmaltz: it works find if I comment out the channel => 4 line |
| 22:18.00 | fu3|gone | :) |
| 22:18.05 | fu3|gone | ok |
| 22:18.20 | fu3|gone | i meant CSPAN :) |
| 22:18.20 | shmaltz | asterisk99, I know it does, thats why I asked you to do it |
| 22:18.25 | asterisk99 | shmaltz: done |
| 22:18.53 | shmaltz | asterisk99, now we have to figure out a way to get it working anyhow |
| 22:19.34 | asterisk99 | shmaltz: crazy thing is... It was working fine up until I rebooted (and unless I've gone crazy, I didn't chnage anything) |
| 22:19.45 | shmaltz | asterisk99, lets try this |
| 22:19.54 | shmaltz | asterisk99, do a demsg -c |
| 22:19.59 | *** join/#asterisk sandra78 (n=sdgfs@200.106.108.241) |
| 22:20.01 | shmaltz | unload zap and then reload it |
| 22:20.09 | shmaltz | and pb dmesg for me |
| 22:20.13 | sandra78 | pls help!!! |
| 22:20.23 | sandra78 | :( |
| 22:20.39 | shmaltz | sandra78, run to the closest POTS line and dial 911 |
| 22:20.52 | sandra78 | :S |
| 22:21.29 | Dr-Linux | sandra78: what makes you cry ? |
| 22:21.30 | RoyK_train-se | sandra78: ask a clear question, and someone might give you a good answer :P |
| 22:21.34 | sandra78 | i got problems with my xp100 doesn't detect when somebody answer from the another side |
| 22:21.43 | asterisk99 | shmaltz: Command not found |
| 22:22.00 | sevard | I'm getting Call not approved when I try dialing an extension, can anyone help me out? I wish this error was more specific. |
| 22:22.02 | sandra78 | i have callprogress= yes bussydetect=yes etc etc |
| 22:22.02 | shmaltz | asterisk99, sorry it's dmesg -c |
| 22:22.32 | asterisk99 | shmaltz: done |
| 22:22.36 | Dr-Linux | RoyK_train-se: lol |
| 22:23.13 | shmaltz | asterisk99, reload zaptel, then run ztcfg and pb the output of dmesg |
| 22:23.34 | asterisk99 | shmaltz: reload as in modprobe zaptel ? |
| 22:23.38 | Dr-Linux | ztcfg -v |
| 22:23.46 | shmaltz | asterisk99, yep |
| 22:24.01 | shmaltz | asterisk99, rmmod for each zaptel module loaded |
| 22:24.05 | shmaltz | then modprobe again |
| 22:24.10 | shmaltz | then ztcfg |
| 22:24.13 | shmaltz | then dmesg |
| 22:24.25 | fu3|gone | ok, for real this time, cya later! |
| 22:25.38 | sevard | Can anyone give me a hand? :/ :( |
| 22:26.20 | Darwin35 | left or right |
| 22:26.30 | justinu | <rimshot/> |
| 22:27.00 | asterisk99 | shmaltz: have a look at http://pastebin.com/604369 |
| 22:27.07 | justinu | ~seen exonic |
| 22:27.11 | jbot | exonic is currently on #asterisk. Has said a total of 4 messages. Is idling for 3h 33m 18s, last said: 'mxmasster, I'm not aware of one, Although you could originate a call in whatever context/exten and watch ;)'. |
| 22:27.17 | justinu | wake up! |
| 22:28.12 | *** join/#asterisk ToTo (n=ToTo@host33-161.pool870.interbusiness.it) |
| 22:28.36 | Darwin35 | but reads that voicemail realtime no longer works with pgsql nativly |
| 22:28.40 | shmaltz | asterisk99, you did something wrong here, redo it as follows: |
| 22:28.42 | shmaltz | rmmod wctdm, rmmod zaptel, dmesg -c, modprobe zaptel, modprobe wctdm, ztcfg, dmesg |
| 22:28.43 | shmaltz | and patebin the output |
| 22:28.54 | RoyK_train-se | methinks fu2 is far better than fu3 |
| 22:29.51 | markit | [TK]D-Fender, shmaltz http://www.voip-info.org/wiki/index.php?page=Asterisk+s+extension |
| 22:30.05 | markit | "If no other match exist for the call within the context, the s extension is activated." |
| 22:30.14 | markit | so should work, why not? |
| 22:30.21 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 22:30.39 | sandra78 | asterisk doesn't detect when somebody answer the call from the pstn side |
| 22:30.49 | r_evolution | that's creepy. |
| 22:31.02 | shmaltz | markit, because you didn't edit that page and changed it? you should, because you tested it and it didn't work |
| 22:31.06 | [TK]D-Fender | markit : There IS a known called number! *723* <------ |
| 22:31.19 | asterisk99 | shmaltz: http://pastebin.com/604378 |
| 22:31.27 | [TK]D-Fender | markit : 723 was CALLED. Getting the idea? |
| 22:31.31 | shmaltz | sandra78, can you explain why you decided such a claim against asterisk? |
| 22:32.11 | r_evolution | l( |
| 22:32.14 | r_evolution | ;) even |
| 22:32.34 | sandra78 | <shmaltz> i got a xp100 when i make a call to the pstn and somebody hangup the call asterisk doesn't detect the answering |
| 22:33.12 | RoyK_train-se | nite |
| 22:33.13 | shmaltz | asterisk99, I got no clue why it works for you when you comment out the 4th channel, because according to your configs it shouldn't work at all |
| 22:33.14 | *** join/#asterisk DaPrivateer (i=Privatee@CRIMSON.OFF-HOURS.COM) |
| 22:33.37 | shmaltz | sandra78, if they hangup asterisk shouldn't detect answering |
| 22:34.01 | r_evolution | that's got to have been meant another way... seriously. |
| 22:34.06 | sevard | [sev-69] |
| 22:34.06 | sevard | type=friend |
| 22:34.06 | sevard | regexten=69 |
| 22:34.06 | sevard | callerid="Severed Seot" <69> |
| 22:34.06 | sevard | host=dynamic |
| 22:34.07 | sevard | nat=no |
| 22:34.09 | sevard | canreinvite=yes |
| 22:34.10 | justinu | ~pb |
| 22:34.12 | jbot | it has been said that pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca |
| 22:34.12 | asterisk99 | shmaltz: what about the config that shouldn't work, but does? |
| 22:34.12 | sevard | disallow=all |
| 22:34.13 | sevard | allow=speex |
| 22:34.15 | sevard | allow=gsm |
| 22:34.17 | sevard | allow=ulaw |
| 22:34.19 | sevard | allow=alaw |
| 22:34.21 | sevard | mailbox=69@default |
| 22:34.21 | shmaltz | asterisk99, what digium card do you have in there? |
| 22:34.23 | sevard | SHIT |
| 22:34.25 | sevard | SORRY |
| 22:34.28 | sevard | justinu: bad paste, i leaned on the right mouse button |
| 22:34.31 | sevard | in putty :( |
| 22:34.41 | *** join/#asterisk Raszh (n=Spoon@66.253.253.201) |
| 22:34.43 | sevard | i r losar. |
| 22:34.46 | justinu | heh |
| 22:34.53 | Raszh | greetings |
| 22:34.55 | r_evolution | yer gross justin :-\ |
| 22:34.56 | sevard | justinu: while you're slapping, want to help me? :) |
| 22:35.03 | sevard | i'm lucky it wasn't my entire config heh |
| 22:35.19 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-49-128.cybersurf.com) |
| 22:35.23 | justinu | i'm like coding and stuff |
| 22:35.24 | Raszh | How do run zaptel on 2.4 without CRC_CCITT? |
| 22:35.25 | justinu | but it's mad boring |
| 22:35.30 | sevard | justinu: I'm getting call not approved on every call :( |
| 22:35.32 | Raszh | er, s/do/do you/ |
| 22:35.43 | justinu | call not approved? what responds with that message? |
| 22:36.00 | sevard | justinu: I'm using the x-lite sip client |
| 22:36.25 | justinu | so xlite responds with call not approved, or asterisk? |
| 22:36.27 | sevard | I thought call not approved was a codec error but i'm allowing all codecs |
| 22:36.36 | sevard | justinu: Sorry, x-lite responds with call not approved. |
| 22:36.45 | justinu | um, yeah. call not approved is generally a codec issue |
| 22:36.50 | sevard | sip debug and the client debugs aren't giving me much information. |
| 22:36.56 | justinu | sevard if you want to pastebin the SIP debug, i'll take a look |
| 22:37.11 | asterisk99 | shmaltz: TDM400P with 2 FXO modules (red) in slots 1 and 2; and 1 FXS (green) in slot 4 |
| 22:37.18 | sevard | justinu: I'm really not getting any debugs outside of register information |
| 22:37.30 | shmaltz | asterisk99, because dmesg is not reporting any digium hardware |
| 22:37.32 | justinu | sevard you should see an invite going from your asterisk machine to Xlite |
| 22:37.59 | sevard | I got nothing mate. Not unless i shutdown my x-lite client and fire it back up again. |
| 22:38.00 | [TK]D-Fender | sevard, what user are you reging as in X-Lite? |
| 22:38.13 | justinu | so how are you making your call to xlite? |
| 22:38.17 | sevard | [TK]D-Fender: 'friend' |
| 22:38.31 | sevard | justinu: what?! |
| 22:38.39 | asterisk99 | shmaltz: aye carumba!!!! I used to have green LEDs on in slots 1 and 2 ... now both dark |
| 22:38.41 | *** part/#asterisk Utah_Dave (n=boucha@0-1pool139-55.nas28.salt-lake-city1.ut.us.da.qwest.net) |
| 22:38.45 | [TK]D-Fender | sevard : I mean what username/account/whatever. not the TYPE in sip.conf. |
| 22:38.49 | justinu | sevard: you said that xlite is responding with a call not approved message |
| 22:38.50 | sevard | justinu: :) i have two x-lite clients registered to the asterisk server. Neither can dial. |
| 22:38.59 | sevard | justinu: correct. |
| 22:39.00 | justinu | sevard: what is causing that responce. |
| 22:39.01 | shmaltz | asterisk99, you didn't follow instructions |
| 22:39.03 | justinu | response? |
| 22:39.14 | shmaltz | asterisk99 I'll give them to you again |
| 22:39.38 | sevard | justinu: when I try to dial anything. Right now I have extensions for two clients, so when I try to dial the other client. It also comes up when I to dial a number that doesn't exist |
| 22:39.39 | justinu | sevard: are you telling me that when you try to make a call from xlite, it says "Call Not Approved"? |
| 22:39.47 | sevard | justinu: Yes. |
| 22:39.49 | justinu | oh. |
| 22:39.56 | justinu | that's simple... put a #1 in front of the number, or something |
| 22:40.00 | asterisk99 | shmaltz: Didn't I? See pastebin from b4 |
| 22:40.01 | justinu | xlite is being a dildo |
| 22:40.20 | *** join/#asterisk Peaceful (n=Peaceful@70.98.162.62) |
| 22:40.38 | shmaltz | asterisk99, do the following: |
| 22:40.40 | shmaltz | rmmod wctdm |
| 22:40.42 | shmaltz | rmmod zaptel |
| 22:40.43 | shmaltz | dmesg -c |
| 22:40.45 | shmaltz | modprobe zaptel |
| 22:40.46 | shmaltz | modprobe wctdm |
| 22:40.48 | shmaltz | ztcfg |
| 22:40.49 | shmaltz | dmegs |
| 22:40.51 | shmaltz | and pastebin it all |
| 22:40.52 | shmaltz | asterisk99, yes according to your pb you didn't |
| 22:41.05 | asterisk99 | shmaltz: sorry |
| 22:41.11 | shmaltz | you missing modprobe wctdm |
| 22:41.26 | Peaceful | Are there any VOIP fax machines that can handle multiple simultaneous inbound faxes coming through asterisk? |
| 22:41.40 | tzafrir_laptop | shmaltz, why modprobe wctdm doesn't load zaptel? |
| 22:41.40 | sevard | justinu: that doesn't work :) |
| 22:41.55 | justinu | it's something like that |
| 22:42.15 | shmaltz | tzafrir_laptop, I'm not sure I'm trying to figure that out, meanwhile he didn't load it at all, meaning he didn't run modprobe wctdm |
| 22:42.18 | justinu | since xlite isn't sending out invites, but responding with that error, it's some kind of xlite problem. |
| 22:42.30 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 22:42.56 | shmaltz | Peaceful, what are you trying to accomplish? |
| 22:43.15 | sevard | justinu: So, if I try to dial #89... call not approved, what the heck are you saying guy? |
| 22:43.27 | justinu | i'm saying dial #1 then your number |
| 22:43.28 | sevard | WHAT THE CRAP |
| 22:43.30 | Raszh | Is it still possible to use zaptel under 2.4? |
| 22:43.33 | sevard | why does that work? |
| 22:43.35 | justinu | lol |
| 22:43.38 | sevard | that shouldn't work |
| 22:43.39 | sevard | wtf! |
| 22:43.43 | justinu | have fun with xlite, bud |
| 22:43.47 | justinu | :P |
| 22:43.52 | sevard | justinu: what the hell man. |
| 22:44.02 | justinu | buy a real phone |
| 22:44.19 | sevard | why does that even work |
| 22:44.25 | sevard | that makes _no sense_ :/ |
| 22:44.32 | justinu | it's something within xlite... a way to select a specific SIP account |
| 22:44.35 | *** join/#asterisk lunaphyte_ (n=lunaphye@70.90.148.1) |
| 22:44.50 | asterisk99 | shmaltz: http://pastebin.com/604400 |
| 22:44.54 | justinu | #1 selects account1, #2 account 2, etc. |
| 22:45.14 | sevard | justinu: is there a way to um.. set that? |
| 22:45.21 | justinu | probably |
| 22:45.27 | justinu | i'm no expert on xlite |
| 22:45.28 | asterisk99 | shmaltz: and 2 greed LEDs for slots 1 and 2 are lit |
| 22:45.32 | justinu | i use eyebeam for softphone |
| 22:45.33 | sevard | justinu: p.s. this is super gay. |
| 22:45.41 | justinu | xlite sucks |
| 22:45.44 | sevard | no shit. |
| 22:46.07 | shmaltz | asterisk99, zaptel driver is not finding any moduels for 3 and 4 |
| 22:46.19 | [TK]D-Fender | x-lite works just fine.... |
| 22:46.25 | *** join/#asterisk Skarmeth (n=Skarmeth@200.165.81.130) |
| 22:46.29 | justinu | it still sucks |
| 22:46.32 | justinu | something can work, and suck |
| 22:46.42 | [TK]D-Fender | sevard : I already asked once : What username are you entering into X-Lite for your registration info? |
| 22:46.43 | shmaltz | asterisk99, this is the problem: |
| 22:46.44 | shmaltz | # |
| 22:46.46 | shmaltz | [4303497.243000] Found a Wildcard TDM: Wildcard TDM400P REV I (2 modules) |
| 22:47.04 | *** join/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 22:47.08 | asterisk99 | shmaltz: slot 3 is empty; slot 4 has the FXS daughtercard |
| 22:47.17 | shmaltz | asterisk99, why? |
| 22:47.26 | sevard | [TK]D-Fender: Sorry dude. |
| 22:47.35 | shmaltz | asterisk99, why not put the daughterboard onto slot 3? |
| 22:47.36 | sevard | [TK]D-Fender: sev-69 |
| 22:48.00 | asterisk99 | shmaltz: keep FXS separated from FXO |
| 22:48.13 | asterisk99 | shmaltz: it's worked that way for 2 weeks |
| 22:48.14 | sevard | [TK]D-Fender: justinu was right, it was an x-lite problem, sometimes it works sometimes it doesn't, if you file it under proxy default and dial it works fine, but if you fial it under proxy 1 even though default is disabled and proxy 1 is enabled you have to dial #1<number> |
| 22:48.18 | sevard | I've never had to do that before. |
| 22:48.24 | shmaltz | asterisk99, [4303496.485000] Timeout waiting for calibration of module 3 |
| 22:48.25 | shmaltz | This is the real problem |
| 22:48.39 | *** join/#asterisk oracle^ (n=cam@unaffiliated/cameleons) |
| 22:48.45 | oracle^ | hello |
| 22:48.47 | Raszh | When was CRC_CCITT added as a requirement for Zaptel? |
| 22:48.47 | asterisk99 | shmaltz: ok - I'll move the daughtercard and see if it makes a diff |
| 22:48.50 | shmaltz | asterisk99, the power cord is connected to the card? |
| 22:49.06 | asterisk99 | shmaltz: yup - It even rang the phone |
| 22:49.24 | asterisk99 | shmaltz: it's like the daightercard has died |
| 22:49.27 | shmaltz | asterisk99, why did you reboot? |
| 22:49.43 | shmaltz | asterisk99, thats what I'm trying to figure out |
| 22:49.48 | asterisk99 | shmaltz: dumbass here put Apache into a loop |
| 22:49.56 | shmaltz | asterisk99, you using an ADSI capable phone with this? |
| 22:50.01 | shmaltz | lol |
| 22:50.04 | tzafrir_laptop | Raszh, for some CRC functions, IIRC |
| 22:50.08 | asterisk99 | shmaltz: beats me |
| 22:50.16 | oracle^ | I'm trying to do something like that : when someone calls, it has 10sec to input a pass code, if pass is not correct, then it goes to the main menu, if it's correct, it goes to a submenu ; how can i get the return value of Authenticate and combine it with a Gotoif ? |
| 22:50.33 | asterisk99 | shmaltz: it's a cheap GE with call display |
| 22:50.38 | tzafrir_laptop | hi oracle^ |
| 22:50.41 | Raszh | tzafrir_la: and how does that work if you're using Linux 2.4, which doesn't have CRC_CCITT? |
| 22:50.55 | shmaltz | oracle^, http://www.voip-info.org/wiki-asterisk+cmd+authenticate |
| 22:51.16 | Raszh | even in as late as 2.4.31 |
| 22:51.21 | *** join/#asterisk _asr (i=asr@pimpbox.latency.net) |
| 22:52.34 | tzafrir_laptop | Raszh, I believe you better state what doesn't work right ahead... |
| 22:53.08 | shmaltz | oracle^, if you get to the priority after the authenticate command then it was correct, otherwise I think the call is hungup |
| 22:53.08 | Raszh | nothing, I haven't reached that point. The README indicates that CRC_CCITT be included in the kernel |
| 22:54.01 | Raszh | in particular, I'm using Gentoo, and the ebuild for zaptel is enforcing that requirement |
| 22:54.08 | tzafrir_laptop | well, it works with the Debian 2.4 kernel |
| 22:54.19 | tzafrir_laptop | (a modified 2.4.27) |
| 22:54.33 | Raszh | at first I thought "sha, right, Portage is lying to me, what's the real story". as it turns out, it's right |
| 22:54.48 | asterisk99 | shmaltz: I'll power it down and move the card... maybe reseating it will help |
| 22:54.58 | Raszh | so do you have /lib/modules/2.4*/kernel/lib/crc-ccitt.ko ? |
| 22:55.03 | shmaltz | asterisk99, what motherboard you using? |
| 22:55.21 | Raszh | (or did you build it into the kernel?) |
| 22:55.28 | tzafrir_laptop | Raszh, there are no .ko-s for 2.4 |
| 22:55.31 | Raszh | yes, my bad |
| 22:55.39 | Raszh | do you have crc-ccitt either as a module or in kernel? |
| 22:55.41 | asterisk99 | shmaltz: shuttle xpm sb86i |
| 22:56.02 | Raszh | or are you using zaptel 1.0 (because it didn't require it then, hence my original question of "When") |
| 22:56.17 | Darwin35 | ok everyone must pay me 30 bucks |
| 22:56.33 | Darwin35 | pay up or no more support for you |
| 22:56.37 | shmaltz | asterisk99, I have seen MB that wont reset the Digium cards (only on the TE405) unless the power is unpluged for like 1 minute |
| 22:56.47 | tzafrir_laptop | Raszh, check the files list at http://packages.debian.org/stable/base/kernel-image-2.4.27-2-386 |
| 22:57.29 | umay | shmaltz: ive seen this on several boards, too |
| 22:57.34 | asterisk99 | shmaltz: hmmmmmm wierd but that's life in the fast lane .... I'll move the module and hope that that fixes it |
| 22:57.37 | tzafrir_laptop | But then again, I never tested zaptel 1.2 with kernel 2.4 . |
| 22:57.43 | oracle^ | shmaltz, j - jump to priority n+101 if the authentication fails and that priority exists , i'm not sure of how to use it, supposing the authentificate is something like that : exten s,1,Authenticate(1111|???) what should i put instead of ??? to go to s,3 for ex ? |
| 22:57.46 | sevard | Question: My extensions can dial eachother, they ring, but when one picks up the x-lite client says 403: Forbidden |
| 22:57.48 | justinu | if anyone should be paid 30 bucks by all, it should be me |
| 22:57.51 | asterisk99 | shmaltz: wird thing... this was working fine before my reboot |
| 22:58.05 | Raszh | tzafrir_la: ok, so you're using 1.0.* still? |
| 22:58.27 | Raszh | because yeah, so am I, but want to upgrade Asterisk, et al |
| 22:58.29 | tzafrir_laptop | yea, we still use it. But kernel 2.6 |
| 22:58.44 | shmaltz | oracle^, use j like this Authenticate(1111|j) |
| 22:58.55 | sevard | Question: My extensions can dial eachother, they ring, but when one picks up the x-lite client says 403: Forbidden --- PasteBin http://pastebin.com/604439 |
| 22:59.10 | sevard | justinu: would you happen to have a magical answer to that one? :) |
| 22:59.12 | shmaltz | oracle^, but be aware that if priorityjumping is set to yes, j will have the oposite effect |
| 22:59.16 | Raszh | I'd be using 2.6 but am using some multi-port modem cards (for dialin/out, not voice) that don't work in 2.6 |
| 22:59.25 | russellb | shmaltz: are you sure? that should not be the case |
| 22:59.36 | Prival2 | Hi all, any hints to raise the voicemail audio level on 10.0.10? |
| 22:59.39 | AlexCTI | Justinu: There are free softphones which comes with the g729 already? |
| 22:59.46 | justinu | nothing free |
| 22:59.51 | shmaltz | russellb, IIRC yes |
| 22:59.53 | justinu | sevard: the clue is on line 19 |
| 23:00.03 | justinu | sevard: codec mismatch |
| 23:00.04 | russellb | shmaltz: lies! |
| 23:00.16 | sevard | justinu: I see. |
| 23:00.20 | sevard | justinu: Hmmmmmmm |
| 23:00.28 | russellb | shmaltz: if you try that and it happens, please tell me so I can fix it ... |
| 23:00.54 | Raszh | and it's clear that it's still supported for 2.4, just not how |
| 23:01.08 | justinu | SIP/pete is trying to use codec 512 (speex) |
| 23:01.20 | [TK]D-Fender | AlexCTI : G729 is a patented coded which means it costs the developer for eaach copy to include it. Who do you think would make a soft-phone and not pass on the cost to yuo be CHARGING you for it? |
| 23:01.21 | sevard | justinu: All of the codecs are turned on in xlite though, I couldn't see why they should be mis matching. |
| 23:01.23 | justinu | SIP/sev is trying to use g711u |
| 23:01.32 | justinu | disallow=all |
| 23:01.34 | justinu | allow=ulaw |
| 23:01.41 | justinu | thank you, that'll be 30 dollars, please :P |
| 23:02.31 | sevard | justinu: I'm inclined to agree but, http://pastebin.com/604449 |
| 23:03.01 | sevard | justinu: both x-lite clients are allowing all codecs and they're also all allowed in the sip.conf |
| 23:03.08 | shmaltz | russellb, I remeber reading it somewhere I didn't make that up, but just to make sure we are talking about +101 and the j option? right? |
| 23:03.37 | russellb | yeah |
| 23:03.41 | justinu | sevard: the way codec negotiating works in SIP is kinda funky. |
| 23:03.44 | russellb | i believe you |
| 23:03.49 | russellb | I just hope that's not actually the case |
| 23:03.50 | justinu | sevard: trust me, take out the allow=speex |
| 23:03.50 | sevard | justinu: ? |
| 23:03.53 | shmaltz | from <asterisksource>/configs/extension.conf.sample: |
| 23:03.55 | shmaltz | ; If priorityjumping is set to 'yes', then applications that support |
| 23:03.57 | shmaltz | ; 'jumping' to a different priority based on the result of their operations |
| 23:03.58 | shmaltz | ; will do so (this is backwards compatible behavior with pre-1.2 releases |
| 23:04.00 | shmaltz | ; of Asterisk). Individual applications can also be requested to do this |
| 23:04.02 | shmaltz | ; by passing a 'j' option in their arguments. |
| 23:04.05 | *** join/#asterisk ManxPower (n=ewieling@dpc6745150107.direcpc.com) |
| 23:04.23 | russellb | if that happens, it's likely only in a certain application, due to a typo |
| 23:04.37 | shmaltz | russellb, but I remember that at one point a notice was put to the userlist that j will reverse the priorityjumping setting |
| 23:04.52 | sevard | justinu: So, that was messed up. YOu were right on the money. |
| 23:05.01 | sevard | justinu: Can you explain codec matching? |
| 23:05.29 | sevard | in sip that is. |
| 23:05.31 | mujjoo | hey again |
| 23:05.48 | justinu | it's how the two UAs decide which codec to use. |
| 23:05.53 | mujjoo | can anyone please help me figure out why the call hangs up after just 4 rings |
| 23:05.58 | justinu | it's pretty stupid |
| 23:06.02 | sandra78 | <shmaltz> i mean if somebodu answer from the pstn side after 10 sec the call is cut off |
| 23:06.16 | justinu | sevard: basically, one of your xlite clients is saying it prefers to use speex over ulaw |
| 23:06.21 | justinu | and it won't fallback |
| 23:06.24 | shmaltz | sandra78, what does the CLI show? |
| 23:07.06 | Zodiacal | anyone know what could cause severe static on one of my fxo modules? its not the line cuz i switched lines and the port still does this... when i try to get a dial tone on the line it just scraches constantly. it was working fine yesterday.... |
| 23:07.16 | sevard | justinu: horrible. |
| 23:07.25 | justinu | yep, it's annoying |
| 23:07.50 | sevard | justinu: well at least now I know which errors to look for. |
| 23:07.51 | *** join/#asterisk diclophis (n=diclophi@65.203.37.58) |
| 23:07.54 | diclophis | howdy all |
| 23:07.56 | justinu | hve fun |
| 23:08.03 | sevard | Although, I don't see where in the file it says one picked speex and one picked ulaw |
| 23:08.03 | diclophis | what does a red alarm on a PRI mean? |
| 23:08.08 | justinu | it means its down |
| 23:08.19 | justinu | sevard: you didn't paste enough of the calls for me to point it out |
| 23:08.38 | diclophis | ... so if i have 2 machines, both with the same zaptel config and one shows a red alarm on span 1 |
| 23:08.45 | diclophis | then i plug back into the other and the alarm is clear |
| 23:08.49 | justinu | sevard: line 19 tells you, you just have to know how to read it |
| 23:08.51 | shmaltz | russellb, google deserted me :( |
| 23:08.56 | diclophis | then the chanes are something is wrong phuscally with the connection |
| 23:09.09 | sevard | justinu: Ahh, I see. So 512== speex and 4 == ulaw |
| 23:09.11 | justinu | diclophis: card is dead, or broken |
| 23:09.14 | justinu | sevard: correct. |
| 23:09.21 | justinu | sevard: show codec from CLI to get the numbers |
| 23:09.33 | justinu | diclophis: or cable problem |
| 23:09.34 | sevard | Excellent. |
| 23:09.47 | Peaceful | shmaltz: I am trying to answer lots of simultaneous faxes without buying lots of fax machines and SPA-2002's. |
| 23:09.52 | ManxPower | ~docs |
| 23:09.53 | jbot | i heard docs is 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 |
| 23:09.53 | ManxPower | ~mailinglist |
| 23:09.55 | jbot | 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, or and there is also the Macintosh Asterisk mailing list at http://www.astmasters.net/maml.htmm |
| 23:10.06 | Raszh | Peaceful: what about Hylafax? |
| 23:10.12 | shmaltz | Peaceful, and exactly do you plan on having the lines for that? |
| 23:10.25 | Peaceful | shmaltz: Well, I've already got 45. |
| 23:10.37 | russellb | shmaltz: there you go. :) |
| 23:10.38 | Raszh | that's 3 too many |
| 23:10.49 | Peaceful | Raszh: I'll check out Hylafax... |
| 23:10.58 | shmaltz | Peaceful, how are they connected to your asterisk box? |
| 23:11.01 | ManxPower | diclophis, red alarm normally means "I don't detect anything" |
| 23:11.11 | Raszh | I'd be surprised if someone hasn't already done some Asterisk/Hylafax integration |
| 23:11.21 | Peaceful | shmaltz: the lines? Two T1/PRI lines. |
| 23:11.36 | shmaltz | Peaceful, then don't use *any* spa products |
| 23:11.51 | shmaltz | Peaceful, how many simultaneous lines you trying to support? |
| 23:11.59 | *** part/#asterisk mujjoo (n=murtazaj@h94s217a102n47.user.nortelnetworks.com) |
| 23:12.16 | Raszh | https://sourceforge.net/projects/iaxmodem |
| 23:12.20 | Peaceful | I've got our office building using all Cisco 7960's. I've got one SPA-3000 connected to our fax machine. |
| 23:12.24 | shmaltz | russellb, thank you for correcting me :) |
| 23:12.49 | Raszh | http://www.voip-info.org/wiki/view/Asterisk+IAXmodem |
| 23:12.52 | shmaltz | Peaceful, how many simultaneous incoming faxes you want to support? |
| 23:12.56 | Raszh | there ya go |
| 23:12.57 | Peaceful | shmaltz: I'd like a fax solution that could answer ~15 simultaneous faxes coming through asterisk |
| 23:13.19 | shmaltz | Peaceful, but you don't want to have 15 fax machines? |
| 23:13.26 | Raszh | ^^ |
| 23:13.30 | Peaceful | shmaltz: exactly |
| 23:13.47 | Peaceful | shmaltz: and I'd only be running that kind of capacity during peak hours |
| 23:14.25 | shmaltz | Peaceful, then Raszh had the right answer for you |
| 23:14.35 | Peaceful | ok. |
| 23:14.35 | Raszh | and http://www.hylafax.org/archive/2005-09/msg00375.html |
| 23:14.48 | *** join/#asterisk juanjoc (n=juanjoc@222-32-235-201.fibertel.com.ar) |
| 23:15.02 | ManxPower | spandsp and rxfax have gotten much better over the past year or so, you should look into that. |
| 23:15.11 | ManxPower | Then you can avoid the hylafax and modems |
| 23:16.28 | Raszh | yeah, IAXmodem uses spandsp |
| 23:17.03 | jarrod | why in the world do I get these 'psss' noises sometimes during quiet times |
| 23:17.13 | jarrod | especially when its ringing... *pssss* |
| 23:17.21 | justinu | how are you talking to PSTN? |
| 23:17.33 | *** join/#asterisk synthetiq (n=family@c-67-163-169-5.hsd1.ct.comcast.net) |
| 23:17.39 | jarrod | mainly through cisco media gateway with pri |
| 23:17.43 | *** part/#asterisk Raszh (n=Spoon@66.253.253.201) |
| 23:17.44 | justinu | it's the gateway then |
| 23:17.47 | jarrod | we also have some backup digium cards |
| 23:17.50 | justinu | try turning down the rxgains |
| 23:17.56 | jarrod | happens thru them too |
| 23:17.58 | justinu | sounds like your RX audio is way too hot |
| 23:18.01 | justinu | on everything |
| 23:18.25 | synthetiq | anyone here use spandsp ? ive compiled is and everythiung and when i use asterisk -vvvvvv it stops loading at app_rxfax.so ...any ideas why |
| 23:18.39 | justinu | use asterisk -vvvvc |
| 23:18.50 | AlexCTI | [TK]D-fender: I undestand that part, thanks |
| 23:19.11 | Darwin35 | ok who here is good with real tiime that is willing for a payment help me move or help explain real time better ot me |
| 23:19.29 | Darwin35 | I have a whole dial plan I want to mve to real time |
| 23:19.41 | jarrod | how extensive is the dial plan |
| 23:20.12 | Darwin35 | let me grab it |
| 23:21.26 | AlexCTI | [TK]D-fender: So I have to buy softphone even with the g729 lic that i already have on my server? So my question is do I need to have both sides licensed? X-PRO g729 and Asterisk server g729 or can I work just with the lic on just one site, and if there is a way, can you tell me how? |
| 23:23.47 | Dr-Linux | justinu: what's difference between PRI and ISDN? :S |
| 23:25.54 | synthetiq | pri is part of isdn |
| 23:26.08 | justinu | PRI is primary rate interface, it's a type of ISDN connection |
| 23:26.37 | Dr-Linux | yeah, but what's differnce :S |
| 23:26.40 | Dr-Linux | hhmm.. :S |
| 23:27.29 | justinu | it's not one or the other |
| 23:27.32 | justinu | ISDN is a generic term |
| 23:27.38 | justinu | PRI is a specific type of circuit |
| 23:27.51 | synthetiq | isdn is either pri or bri |
| 23:28.08 | Darwin35 | http://pastebin.ca/45860 that extensive |
| 23:28.22 | Dr-Linux | ic |
| 23:28.57 | *** join/#asterisk hfb (n=hfb@pool-71-116-252-188.lsanca.dsl-w.verizon.net) |
| 23:30.58 | Darwin35 | I want to learn but not getting much input |
| 23:34.13 | [TK]D-Fender | AlexCTI : When you pay for X-Pro it COMES with G729. |
| 23:34.27 | [TK]D-Fender | AlexCTI : You only need 1 license for * |
| 23:36.11 | *** join/#asterisk mtgh (n=chatzill@dsl093-001-038.det1.dsl.speakeasy.net) |
| 23:37.08 | *** part/#asterisk markit (n=konversa@host119-245.pool8172.interbusiness.it) |
| 23:38.30 | xevo | Is there a phone (other than Cisco) that has an on-phone directory (via XML or whatever)? |
| 23:38.47 | Aurs | polycom |
| 23:38.59 | Aurs | linksys sipura spa-941 |
| 23:39.17 | xevo | The spa941 does? Good! |
| 23:39.25 | Aurs | yes, it does |
| 23:39.48 | justinu | polycom 601 |
| 23:40.02 | justinu | the entire polycom line has a phone directory |
| 23:40.07 | justinu | the 601 has an XML minibrowser |
| 23:40.17 | xevo | That's very cool |
| 23:40.22 | xevo | How does the SPA941 do it? |
| 23:40.44 | justinu | no idea |
| 23:40.48 | justinu | that phone isn't worth the money |
| 23:41.01 | Aurs | justinu: why? me likes |
| 23:41.10 | justinu | 1 ethernet port |
| 23:41.13 | Aurs | true |
| 23:41.19 | justinu | polycom 501 makes a better phone for a similar price |
| 23:41.21 | xevo | 942s have two, right? |
| 23:41.29 | justinu | 942 == 2 10mbit ports |
| 23:41.30 | justinu | 10mbit! |
| 23:41.34 | justinu | what fucking century is this? |
| 23:41.48 | ManxPower | justinu, you have to wonder what they were thinking, eh? |
| 23:41.53 | justinu | so idiotic |
| 23:42.05 | [hC] | yeah thats pretty stupid |
| 23:42.09 | [hC] | im surprised they even released the 941. |
| 23:42.15 | [hC] | with no poe/switch port |
| 23:42.31 | [hC] | You're really only shooting yourself in the foot with it, to save what.. $30? |
| 23:42.35 | xevo | the 501 can view XML over what, http? |
| 23:42.44 | FuriousGeorge | i know you guys are gonna tell me im crazy, but im losing my iax provider peer (timeout) from time to time, and the only way i have been able to get him back is by restarting the server |
| 23:42.47 | justinu | 501's phonebook isn't xml driven |
| 23:42.51 | FuriousGeorge | restarting networking probably would have worked |
| 23:42.56 | justinu | it's downloaded from the provisioning server at phone boot |
| 23:43.03 | justinu | and saved back to the prov server when the user modifies it |
| 23:43.04 | xevo | ahh ok |
| 23:43.09 | *** join/#asterisk buu (i=buu@ded430-deb-175-30.netsonic.net) |
| 23:43.16 | xevo | does dhcp tell it where to look for it? |
| 23:43.17 | justinu | 601 has the real microbrowser, which uses http |
| 23:43.20 | justinu | dhcp does |
| 23:43.26 | buu | So. Is there some kind of brilliant prebuilt solution for dynamically controlling hold music? |
| 23:43.26 | justinu | along with all it's other config files |
| 23:43.29 | justinu | and software |
| 23:43.44 | xevo | polycom sounds like it's the way to go |
| 23:43.59 | *** join/#asterisk dja_ (n=alden@cpe-24-95-62-160.columbus.res.rr.com) |
| 23:44.31 | justinu | xevo: i don't think you'll be disapointed.. at least demo one |
| 23:44.54 | Aurs | xevo: polycom is not a bad choice at all |
| 23:44.59 | xevo | will do |
| 23:45.33 | xevo | we were going to use the 601 for the receptionist phone anyway |
| 23:45.47 | justinu | 601 rawks |
| 23:45.53 | justinu | it's what sits on my desk, and i've tried them all |
| 23:46.01 | xevo | and it looks like it's the one for my desk too |
| 23:46.02 | xevo | hehe |
| 23:46.06 | FuriousGeorge | anyone ever notice their iax providers becomes unreachable, even when he shouldnt be? (i.e. its some mysterious problem on your end) |
| 23:46.21 | justinu | there is some problem with qualify, apparently |
| 23:46.42 | justinu | i don't have too many issues with qualify and sip |
| 23:46.43 | xevo | ordering a couple now |
| 23:46.49 | justinu | but i've heard people bitch about that on IAX before, FuriousGeorge |
| 23:47.22 | *** join/#asterisk giggles (n=chatzill@ool-18bb0d86.dyn.optonline.net) |
| 23:47.48 | willt | should call-limit=1 make it so my user can only use one line at a time? |
| 23:48.23 | *** part/#asterisk angom_w (n=angom@red-corp-200.38.16.10.telnor.net) |
| 23:48.24 | willt | seems to not work :( |
| 23:48.43 | ManxPower | most of that call limit stuff never worked well for me. |
| 23:48.53 | *** join/#asterisk tdonahue (n=tdonahue@208.51.101.201) |
| 23:49.05 | *** join/#asterisk zaf (n=tfournet@ip68-226-162-240.lf.br.cox.net) |
| 23:49.29 | willt | It seems there should be some way to doit |
| 23:49.39 | willt | maybe it's a broken feature? :( |
| 23:49.43 | justinu | probably |
| 23:49.56 | ManxPower | willt, since I don't bill for calls it's not an issue for me. |
| 23:50.32 | willt | is the forums at digium a good place to inquire about it or is there a better place? |
| 23:50.51 | ManxPower | forums? |
| 23:50.57 | ManxPower | I just check the mailing list archives |
| 23:51.08 | *** join/#asterisk sofh (i=realvir@30-150-254-84.skylogicnet.it) |
| 23:51.47 | sofh | hello all..is there anything like CRTP in asterisk ? like cisco had |
| 23:51.59 | ManxPower | sofh, no. |
| 23:52.10 | ManxPower | But since CRTP is mostly a router to router thing..... |
| 23:52.14 | FuriousGeorge | justinu: i got qualify set to 750 right now, but i kinda like that catch to avoid crappy sounding calls |
| 23:52.22 | sofh | hmm.. |
| 23:52.35 | FuriousGeorge | think i should take it out? |
| 23:52.36 | ManxPower | FuriousGeorge, remember the qualify number is NOT the ping times. |
| 23:52.40 | FuriousGeorge | id rather change to sip, tbh |
| 23:52.45 | hfb | Is anyone familiar with (old) ekt65xx phones? |
| 23:52.54 | FuriousGeorge | ManxPower: remember? i never knew to begin with :) |
| 23:53.00 | sofh | as i just notice that if we are using ciso or quintum gateway then a call with g729 is not taking more then 30kbit/s for both incoming and outgoing |
| 23:53.06 | ManxPower | it's the time to takes to receive a response to a SIP options packet (or IAX2 ping) |
| 23:53.09 | FuriousGeorge | is that how often it sends keep alives |
| 23:53.13 | hfb | Toshiba ekt65xx that is. |
| 23:53.16 | sofh | whereas in asterisk g729 and one call is consuming almost 62kbit/s :( |
| 23:53.23 | justinu | FuriousGeorge: you should be using the jitter reports for that kinda stuff! |
| 23:53.34 | justinu | doesn't iax do a qos dump at the end of a call? |
| 23:53.36 | ManxPower | sofh, your measurements are off. |
| 23:53.46 | sofh | like ? |
| 23:53.59 | FuriousGeorge | justinu: isnt it better to check before the call to see how reachable the provider is or isnt |
| 23:54.05 | FuriousGeorge | and will SIP have this same problem? |
| 23:54.08 | sofh | i told what i realised.. |
| 23:54.08 | ManxPower | G729 is about 8kbps + about 24kbps of overhead. |
| 23:54.20 | sofh | u mean 24 + 8 ? |
| 23:54.37 | justinu | Furious: reachability has nothing to do with QoS usually |
| 23:54.41 | justinu | reachability is up/down |
| 23:55.20 | sofh | if its up/down => 31kbit/s then its great ..but i found its consuming 30kbit/s for one side |
| 23:56.05 | FuriousGeorge | justinu: well im not sure if im 100% with you but i get the idea. let me ask this way: assuming taking out qualify solves the problem, isn't there now a chance that * will try to use a provider that is terribly lagged? |
| 23:56.08 | sofh | Manxpower! or is there any setting in linux FW or asterisk itself to tune up the bw consumption ? |
| 23:56.39 | justinu | FuriousGeorge: probably true |
| 23:57.10 | justinu | FuriousGeorge: if I were serious about it... i'd write a patch to asterisk that would use the QoS stats from the calls to modify the select order of the IAX prodivers. |
| 23:57.25 | justinu | but it doesn't do that out of the box |
| 23:57.26 | justinu | ;) |
| 23:57.40 | ManWithYellowBat | :-D |
| 23:58.10 | *** join/#asterisk fugitivo (n=ajf@201.216.246.181) |
| 23:58.24 | sofh | ?"| |
| 23:58.24 | fugitivo | ~seen coppice |
| 23:58.27 | jbot | coppice <n=chatzill@211.155.168.153> was last seen on IRC in channel #asterisk, 1d 13h 57m 25s ago, saying: '"Nun"'. |
| 23:59.17 | *** join/#asterisk dec (n=tom@ppp151-30.lns3.adl2.internode.on.net) |
| 23:59.32 | ManWithYellowBat | ~seen fugitivo |
| 23:59.34 | jbot | fugitivo is currently on #asterisk (1m 24s). Has said a total of 1 messages. Is idling for 1m 10s, last said: '~seen coppice'. |
| 23:59.34 | ManWithYellowBat | :( |
| 23:59.38 | ManWithYellowBat | ;) |
| 23:59.49 | hfb | Hmmm. I just found an old log for this channel. |
| 23:59.53 | ManWithYellowBat | i was hoping for a loop |
| 23:59.55 | dec | Hello... I'm receiving warnings in my asterisk log when an iax call comes in: socket_read: Received trunked frame before first full voice frame |