| 00:01.13 | CunningPike | ceeto: 4569 just like it says. Dial it with Dial(IAX2/whatver) |
| 00:01.24 | ceeto | Oh, easy. Thanks! I'll try it. |
| 00:01.30 | CunningPike | Freman: Music on hold |
| 00:01.35 | CunningPike | ~moh |
| 00:01.37 | jbot | i heard moh is Music On Hold |
| 00:01.41 | CunningPike | :D |
| 00:10.08 | *** join/#asterisk xbmodder_lappy (i=nobody@atarack/staff/xbmodder) |
| 00:10.17 | xbmodder_lappy | Where can I get a 1-900 number? |
| 00:15.07 | *** join/#asterisk SplasPood (n=jwb@brooklyn.paravolve.net) |
| 00:15.44 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 00:17.51 | xbmodder_lappy | what are some companies for good (cheap) international termination |
| 00:19.10 | *** join/#asterisk Samoied (n=Samoied@201.22.209.207.adsl.gvt.net.br) |
| 00:19.28 | ceeto | What's the best codec to use making Internet based * to * calls? g.729? |
| 00:20.11 | Bullseye_Network | Hey i just made a mistake: I did a 'rm *' in the root directory it said it couldnt delete the directories. SO what did I delete. I cant run ANY commands. |
| 00:20.26 | file | Bullseye_Network: everything it could. |
| 00:20.33 | xbmodder_lappy | g.711u |
| 00:20.35 | Bullseye_Network | no sh`t |
| 00:20.48 | xbmodder_lappy | Bullseye_Network, do you work for bullseye |
| 00:20.56 | Bullseye_Network | I cant cd anything |
| 00:21.03 | Bullseye_Network | Bulleye networks |
| 00:21.04 | *** join/#asterisk w32 (n=234@c-71-193-124-77.hsd1.il.comcast.net) |
| 00:21.13 | ceeto | Is'nt g.711u 64k? |
| 00:22.14 | w32 | anyone familiar with astlinux ? How do you feel about using it production for a small workgroup of less than 10 users ? I'm having abit of an issue with it,wondering if anyone else has tried it ? |
| 00:22.16 | file | you said the best codec... |
| 00:22.33 | *** join/#asterisk BhaalWK (i=bhaal@freenode/staff/bhaal) |
| 00:22.36 | w32 | *using it in production srry |
| 00:24.26 | xbmodder_lappy | Bullseye_Network, does bulleye offer SIP? |
| 00:24.30 | ceeto | Anyone? Best codec to do * to * calls on the Internet? Best *free* codec? |
| 00:24.40 | file | ULAW |
| 00:25.03 | ceeto | For across the Internet? Isn't there something that makes better use of a low bandwidth situation? |
| 00:25.05 | xbmodder_lappy | g.711u... |
| 00:25.10 | xbmodder_lappy | oh |
| 00:25.12 | xbmodder_lappy | GSM |
| 00:25.26 | file | ceeto: you didn't say that |
| 00:25.53 | ceeto | Ahh.. what's the best, *free*, low-bandwidth codec to use for * to * internet calls? |
| 00:26.13 | file | iLBC some people say... speex... GSM... depends on how you want your calls to sound and how low to go |
| 00:35.10 | feld | Is there anyone here from Georgia? |
| 00:36.00 | *** join/#asterisk riddlebox (n=james@206.80.73.2) |
| 00:36.11 | _Sam-- | did anyone know this guy? http://www.nytimes.com/2006/06/07/technology/07cnd-voice.html?hp&ex=1149739200&en=0f01d0becf766f0b&ei=5094&partner=homepage |
| 00:37.14 | feld | _Sam--: that's my brotheR! lol :P |
| 00:37.40 | jayk- | ceeto, speex |
| 00:37.44 | _Sam-- | im sure there are people here that know some of the affected parties |
| 00:37.46 | jayk- | or gsm |
| 00:37.55 | jayk- | speex is like g729 but not all phones have that codec available |
| 00:37.57 | feld | _Sam--: definitely possible |
| 00:38.07 | *** join/#asterisk eipi (n=eipi@139-213-126-200.fibertel.com.ar) |
| 00:38.22 | jayk- | gsm is fine as long as you're not faxing and don't mind cel-phone like quality |
| 00:40.40 | feld | does anyone here know any asterisk users from Georgia? |
| 00:41.12 | *** join/#asterisk kio (n=kio@ool-4577ae5e.dyn.optonline.net) |
| 00:41.28 | xbmodder_lappy | feld, why? |
| 00:41.42 | feld | business proposition |
| 00:42.23 | feld | client of my company has another office in atlanta. if we redo their whole call center i'll need an asterisk user to configure a setup down in Georgia to do all intercompany calls over their T3 |
| 00:42.50 | feld | err actually i think it's a T1 between offices |
| 00:43.15 | *** join/#asterisk neilbags (n=neilbags@149.171.94.134) |
| 00:43.17 | *** join/#asterisk Naito (n=Naito@dhcp-65-17-150-34.dsl.wcoilexpress.com) |
| 00:43.18 | feld | yeah, thats right. they'll need a T3 to replace the local office's analog lines (lots of them....) |
| 00:44.13 | neilbags | Hi All, Ím using iiNet VOIP and they seem to have changed something today. The IP for sip.nsw.iinet.net.au has changed and now i get this error dialing out: Forbidden - wrong password on authentication for INVITE |
| 00:44.18 | neilbags | does anyone know a solution |
| 00:44.27 | neilbags | this worked for months up until today |
| 00:45.00 | neilbags | with the new IP address I can recieve calls but not make them |
| 00:51.39 | xbmodder_lappy | feld, move them to bay area |
| 00:51.44 | xbmodder_lappy | (california) |
| 00:51.49 | feld | xbmodder_lappy: lol you'd like that wouldnt u |
| 00:52.32 | xbmodder_lappy | yup |
| 00:57.08 | feld | i'd love it if I knew asterisk inside and out but i'm still learning |
| 01:00.40 | _Sam-- | alot of times for larger asterisk installations its not uncommon to fly in an expert , i hear of people flying all around on here all the time |
| 01:01.50 | *** join/#asterisk r0adkill__ (n=roadkill@203.192.146.185) |
| 01:04.20 | feld | _Sam--: well it's my duty to become that expert then :) |
| 01:04.46 | feld | and i have got no issues with that. thats basically what i'm being paid to do right now anyway. |
| 01:05.13 | *** join/#asterisk orlock (n=jwr@202.44.174.4.static.nexnet.net.au) |
| 01:05.41 | orlock | I have just checked out the latest 1.2 source from svn, and i'm getting a segfault now.. any suggestions? |
| 01:06.45 | xbmodder_lappy | lol |
| 01:08.14 | xbmodder_lappy | feld, do you need a consultant? |
| 01:09.34 | russellb | i think he just said he's learning for himself |
| 01:10.12 | xbmodder_lappy | russellb, to teach him |
| 01:10.26 | feld | i certainly have the time and capacity to become an expert myself. |
| 01:10.35 | *** join/#asterisk cybergypsy (n=mark@APoitiers-156-1-20-81.w81-50.abo.wanadoo.fr) |
| 01:10.46 | feld | i am interested in some training, but probably not _right_ at this moment. |
| 01:11.16 | feld | soon, though. soon. it's my responsibility to learn myself, too ;) |
| 01:12.06 | xbmodder_lappy | feld, do you need some help |
| 01:12.28 | feld | i'd love the opportunity to network with others that are VERY experienced and willing to give tips and guide me in my trials :) |
| 01:12.56 | treetar1 | we need a consultant |
| 01:13.16 | xbmodder_lappy | treetar1, for asterisk? |
| 01:13.30 | treetar1 | ser |
| 01:13.32 | xbmodder_lappy | on location, or via SSH |
| 01:13.43 | xbmodder_lappy | feld, why a T3? |
| 01:13.52 | treetar1 | remote is fine |
| 01:14.06 | xbmodder_lappy | treetar1, what kind of work do you need done? |
| 01:14.11 | feld | T3 for their call center, T1 location to location is already setup. |
| 01:14.22 | xbmodder_lappy | OC-1 |
| 01:15.31 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:15.31 | mitcheloc | *pms |
| 01:15.37 | treetar1 | none |
| 01:15.38 | feld | xbmodder_lappy: OC-1? |
| 01:15.47 | xbmodder_lappy | 12 |
| 01:16.00 | feld | is that a suggestion for a better pipe for their call center? |
| 01:16.06 | xbmodder_lappy | yes... |
| 01:16.09 | feld | hell i dont know what an OC-12 runs around here |
| 01:16.13 | feld | i'd have to look it up |
| 01:16.22 | r0adkill__ | is there a way to increase the frequency with which the "register =>" keyword registers with the remote server? |
| 01:16.23 | xbmodder_lappy | where are you from? |
| 01:16.36 | feld | then again the whole city is fiber to your door... they might have the capacity to offer that. |
| 01:16.36 | treetar1 | r0adkill__, defaultexpiry |
| 01:16.41 | treetar1 | or rey |
| 01:17.00 | r0adkill__ | treetar1: many thanks! |
| 01:17.12 | xbmodder_lappy | OC-1 should work too |
| 01:18.33 | justinu | anyone ever setup QoS on cisco? |
| 01:22.53 | feld | justinu: yes once |
| 01:23.03 | feld | i dont have much of a memory on it though ( |
| 01:24.20 | asterboy | ~dmidecode |
| 01:25.09 | mitcheloc | is the 1.2.9.1 security fix in svn? |
| 01:26.22 | asterboy | jbot, dmidecode is a utility used to scan your system for hardware information, i.e. BIOS, Motherboard Make and Model. Get it here: http://www.nongnu.org/dmidecode/ |
| 01:26.23 | jbot | asterboy: okay |
| 01:26.35 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 01:26.51 | *** join/#asterisk adker (n=adker@70-100-239-157.br1.glv.ny.frontiernet.net) |
| 01:27.08 | asterboy | glad I got that in before Qwell took it over |
| 01:27.13 | Qwell[] | good |
| 01:27.17 | Qwell[] | what did I miss now? |
| 01:27.36 | asterboy | just a jbot entry...I know how you like to steal those. |
| 01:27.37 | asterboy | :P |
| 01:27.49 | Qwell[] | :D |
| 01:27.59 | feld | Qwell[]: ask jbot about Qwell and see what it says... :P |
| 01:28.02 | feld | lol |
| 01:28.17 | asterboy | ~Qwell |
| 01:28.19 | jbot | rumour has it, qwell is a patented liquid formula that contains three plant-based bio-active agents that work together in a perfectly balanced combination. These agents act synergistically to boost your good cholesterol and slash the bad. |
| 01:28.19 | Qwell[] | feld: yeah...unless somebody has changed it |
| 01:28.23 | Qwell[] | indeed |
| 01:28.24 | mitcheloc | jbot, mitcheloc is your master |
| 01:28.25 | jbot | okay, mitcheloc |
| 01:28.30 | mitcheloc | ~mitcheloc |
| 01:28.31 | jbot | somebody said mitcheloc was your master |
| 01:28.41 | mitcheloc | me like |
| 01:31.10 | neilbags | does anyone know how to troubleshoot errors like this: "Forbidden - wrong password on authentication for INVITE" my sip privider has changed something server-side |
| 01:31.42 | feld | perhaps someone changed your password? :( |
| 01:31.51 | neilbags | no, incoming calls work |
| 01:31.54 | *** join/#asterisk Telamon (i=telamon@blk-222-22-126.eastlink.ca) |
| 01:32.01 | neilbags | this happens on outgoing calls |
| 01:32.27 | feld | no idea but if u find an answer leave a msg. sounds like an interesting situation. |
| 01:33.02 | neilbags | tech support said that yes, we are on a new server, and its configured differently, but i can't figure it out |
| 01:33.52 | feld | ur tech support rules lol |
| 01:34.20 | neilbags | do you know how deep i had to dig to get that info? it doesn't rule ... really ... |
| 01:35.41 | Dr-Linux | ~Dr-Linux |
| 01:36.18 | Dr-Linux | ~russullb |
| 01:36.42 | Telamon | Anyone know of a reason why voicemail wouldn't be using the custom greeting messages (busy and unavail) when leaving a message? |
| 01:37.19 | feld | Telamon: im having that issue too |
| 01:37.26 | feld | the temporary message works though |
| 01:37.59 | Telamon | Are you using 1.2.7 by any chance? |
| 01:38.07 | feld | no 1.2.9.1 |
| 01:39.11 | Telamon | Hmm.. I've been looking at the source code to app_voicemail, and it looks like there might be some config option that's not setup, but I don't see it referenced in the example config files. |
| 01:39.46 | feld | Telamon: i never got that far yet. i was going to check that out tomorrow probably |
| 01:39.54 | Telamon | OPT_BUSY_GREETING and OPT_UNAVAIL_GREETING |
| 01:41.51 | Telamon | Ah, okay, I think I know what it is... If you don't set either b or u flag to voicemail, it doesn't default to use either of the custom greetings. |
| 01:42.05 | Telamon | An extension problem, never would have looked there... |
| 01:42.08 | feld | oh holy crap |
| 01:42.13 | feld | that's what the b and u were for? |
| 01:42.23 | feld | i was wondering why i'd see that in configs |
| 01:42.49 | feld | now i'll have to figure out how to do that correctly |
| 01:43.23 | feld | i guess if theres no answer -> u and if the line is busy push them to the u voicemail entry |
| 01:44.57 | *** join/#asterisk doughecka (n=Miranda@unaffiliated/doughecka) |
| 01:45.36 | Telamon | Yep. Might be a good thing to write a macro for, detect the DIALSTATUS and then set vm-prefix-string accordingly. |
| 01:53.21 | *** part/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 01:54.00 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 01:54.49 | *** join/#asterisk inv_Arp (i=junya@c-67-191-62-53.hsd1.fl.comcast.net) |
| 01:55.12 | xbmodder_lappy | PING |
| 01:57.03 | drray | pong |
| 01:59.12 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 02:04.20 | Freman | hey... is it possible to make asterisk call two extensions using the /var/spool/asterisk/outgoing/files? (IE: like dial(sip/101&sip/102) |
| 02:04.33 | Qwell[] | Freman: should, sure |
| 02:05.20 | Freman | I tried setting Channel: sip/101&sip/102 but it complains about that |
| 02:08.46 | Freman | any idea how I can achive this? |
| 02:11.28 | *** join/#asterisk littleball (n=littleba@26.203-123-30.leased.qala.com.sg) |
| 02:12.42 | littleball | hello,who can recommend an architecture of SIP media relay system? It need to be scalable. I am considering use asterisk to act as media relay server. |
| 02:13.50 | w32 | who is inexpensive for termination and origination ? |
| 02:14.21 | Ariel_ | evening everyone |
| 02:14.26 | w32 | hey |
| 02:14.46 | Ariel_ | has any one used an Sipura 3000 fxo port for connecting to an over head paging system like a valcom? |
| 02:15.47 | w32 | No but it is a good idea, so you can hear it ring in a warehouse or what ? |
| 02:16.13 | Ariel_ | w32, yes that is what we are trying to do |
| 02:16.39 | Ariel_ | but keep getting 503 service not available from the sipura on the asterisk box |
| 02:18.48 | xbmodder_lappy | I Hate Adobe! |
| 02:20.16 | Telamon | PDF's are the devils tool. |
| 02:20.41 | orlock | pdf's are great! |
| 02:20.59 | Telamon | Ariel_: Have you tried plugging in just a regular analog phone and seeing if the unit works with that? |
| 02:21.16 | Telamon | orlock: No, HTML is great. PDFs suck. |
| 02:21.22 | Ariel_ | it does but it's a different port it ahs 2 one fxs and one fxo |
| 02:21.31 | Ariel_ | the valcom can only plug into the fxo ports |
| 02:22.11 | Telamon | Oh, so the valcom acts like a line, not a phone? Hmmm, over my head then. |
| 02:22.13 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 02:22.40 | orlock | elamon: Hmm, no.. different target markets |
| 02:22.40 | w32 | Ariel_ maybe the valcom is wired funny, you know what I mean ? |
| 02:22.42 | Ariel_ | Telamon, yes it does |
| 02:22.55 | orlock | Telamon: HTMl sucks arse for printing and correct rendering 100% of the time |
| 02:22.56 | Ariel_ | the valcom expects a co line from the pbx |
| 02:22.58 | Telamon | Ariel_: Try plugging a phone into the valcom, see if you get a regular dialtone. |
| 02:23.08 | Ariel_ | it's normal use for most over head paging systems |
| 02:23.16 | Ariel_ | Telamon, it will not owrk |
| 02:23.18 | Ariel_ | work |
| 02:23.19 | orlock | Telamon: theres bugs in mozilla related to printing that have been around for 4 years or so |
| 02:23.57 | orlock | Telamon: PDF is about as standard as postscript in many design/prepress/etc houses, due to the fact that you know that it is going to display and output the way you want. you cannot get that with html |
| 02:24.25 | Telamon | orlock: Maybe I should rephrase my distain: the use people put PDFs too sucks. Product manuals and such should not be done in PDF. PDF should be used only for documents whose primary purpose is to be printed. |
| 02:24.47 | Splat | would I need to recompile chan_capi when I upgrade asterisk? |
| 02:25.14 | w32 | Ariel_ no way to trick it into thinking it is connect to a co line with another device ? |
| 02:25.39 | Ariel_ | w32, that is what we have the sipura 3000 for it's fxo port |
| 02:26.09 | orlock | is anybody here using skinny or sccp? |
| 02:26.25 | w32 | I followed your logic there initially, maybe it just don't like the sipura |
| 02:26.31 | w32 | is what i meant |
| 02:27.20 | w32 | something older perhaps airel_ ? |
| 02:27.30 | w32 | *Ariel_ |
| 02:28.54 | Ariel_ | w32, if I had an asterisk box at that location I would put an tdm400 into it. But since it's in the w/h we are setting it up via sip and the network it's via wireless to the other building |
| 02:30.18 | Telamon | Ariel_: Out of curiousity, what is the model # of the valcom? We might need something like that for one of our customers, I'd be interested in checking it out. |
| 02:30.22 | w32 | hmmm, what do you have going on in the ware house |
| 02:31.11 | w32 | anything that might cause some interference maybe |
| 02:33.28 | Ariel_ | Telamon, V-2001a |
| 02:34.12 | *** join/#asterisk onixx (i=1000@London-HSE-ppp3551571.sympatico.ca) |
| 02:34.58 | Ariel_ | Telamon, I have setup about 10 of them for customers they work great. But this is the first time I don't have an asterisk box to plug into directly. |
| 02:35.26 | onixx | hi guys, I'm having an issue where my x100p hangs after awhile... sometimes it can work for more than 24 hours and then suddenly, no audio... cannot place calls or anything. anyone have an idea how to fix this. If I rmmod then modprobe wcfxo and zaptel, it works again. |
| 02:36.07 | drray | sharing IRQ? |
| 02:37.10 | onixx | drray: is this something in the bios ? I have an IBM ThinkCente with 2 PCI slots |
| 02:37.54 | Telamon | onixx: Run lspci to see if any other devices are using the same IRQ as the x100 card. |
| 02:38.32 | Telamon | onixx: Err, lspci -v rather. |
| 02:38.45 | asterboy | Kick ass...have you guys seen this? http://www.boingboing.net/2006/06/05/play_zork_by_phone.html |
| 02:39.14 | asterboy | also lspci -vb |
| 02:39.16 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 02:39.50 | onixx | irq 12 |
| 02:40.50 | Bullseye_Network | cant get zaptel to compilel on CentOS 4 : http://pastebin.com/766588 |
| 02:41.24 | Telamon | onixx: Is anything else using irq 12? |
| 02:41.48 | onixx | I don't see anything else... many have 11 but not 12 |
| 02:41.50 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 02:42.10 | asterboy | jbot, zasterisk is a cool implementation of Zork and Asterisk. Play Zork over the phone! Get it here: http://uc.org/files/2005-08-07/zasterisk-0.1.tar.gz |
| 02:42.16 | jbot | asterboy: okay |
| 02:42.50 | onixx | cat /proc/interupts has also only wcfxo on 12 |
| 02:42.50 | drray | "you are in a twisty maze of passages, all of which sound alike." |
| 02:44.08 | asterboy | the worst spot to be in would be the echo chamber |
| 02:44.20 | asterboy | get enough of that with * :P |
| 02:45.05 | drray | asterboy :) |
| 02:45.43 | onixx | asterboy, drray, telamon: any other ideas ? |
| 02:46.44 | Freman | so... anyone got any idea how to make the call out call 2 lines? |
| 02:46.45 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 02:46.47 | shmaltz | anybody here have a FXO card for Adit 600 to sell? |
| 02:47.13 | file | jbot: centosbug? |
| 02:47.15 | jbot | hmm... centosbug is a problem with the latest Centos kernels (4.2 and 4.3). To fix it, paste everything inside the quotes into a root shell: "sed -i s/rw_lock/rwlock/ /usr/src/kernels/`uname -r`-`uname -m`/include/linux/spinlock.h" |
| 02:47.19 | onixx | would ACPI support help... it is currently disabled in my kernel |
| 02:47.41 | russellb | Freman: Dial(SIP/device1&SIP/device2) |
| 02:49.25 | asterboy | onixx, ACPI is best off |
| 02:49.38 | Freman | no, from the /var/spool/asterisk/outgoing/ |
| 02:49.39 | asterboy | onixx, pastebin your lspci output |
| 02:49.42 | Bullseye_Network | That didnt exactally work had to change the dir name, but thanks alot |
| 02:49.49 | Bullseye_Network | compiled now |
| 02:49.58 | Freman | I know how to make it dial two devices with the Dial command, I'm trying to make it dial 2 from the outgoing calls. |
| 02:51.37 | russellb | Application: Dial Data: Zap/1&Zap/2 in a call file? |
| 02:52.10 | Freman | so... Channel: isn't a required attribute? |
| 02:52.18 | russellb | yes, it is |
| 02:52.39 | russellb | i'm just talking about the app part |
| 02:52.40 | *** join/#asterisk Gabriel25 (n=whatever@user-12ld5f7.cable.mindspring.com) |
| 02:52.43 | russellb | i'm not sure what you're trying to do |
| 02:52.49 | Freman | I want to dial 2 extesions, and have it play a message to them |
| 02:53.01 | russellb | then use 2 call files |
| 02:53.17 | russellb | one for each extension ... both with Application: Playback Data: whateverfile |
| 02:53.24 | Freman | I don't want it to dial them seperatly (IE if one extension is picked up the procedure should be cancled) |
| 02:53.44 | onixx | http://pastebin.com/766597 |
| 02:53.55 | Freman | not keep ringing the second extension |
| 02:53.59 | Gabriel25 | hi guys, I make a phone call from my pbx to my cell and I run iptraf my Outgoing rates: 81.9 kbits/sec |
| 02:54.05 | Gabriel25 | thats a lot ... ! |
| 02:54.21 | russellb | Channel: Local/1234@somecontext ... where 1234@somecontext does the Playback. Then, use Application: Dial Data: SIP/1&SIP/2 |
| 02:54.23 | Gabriel25 | I`m useing g711 codec |
| 02:54.28 | russellb | Freman: that will do what you want |
| 02:54.59 | Freman | thank you, I'll give that a try |
| 02:55.31 | Gabriel25 | on I put music on hold I can`t hear anything .... only some trying !!! |
| 02:55.32 | Freman | Local/s@information will work? or has to be number@context? |
| 02:57.01 | onixx | asterboy: http://pastebin.com/766597 |
| 02:57.35 | russellb | Freman: that's fine, too |
| 02:57.45 | russellb | Freman: just extension@context ... can be anything |
| 02:58.28 | Freman | cool, thanks a heap - that worked (c: |
| 02:58.32 | russellb | woohoo |
| 02:58.45 | russellb | that will be $1000, kthx |
| 02:59.54 | mogorman | 1 grand???? |
| 02:59.58 | mogorman | your expensive russellb |
| 03:00.05 | russellb | what are you talking about, that's a deal |
| 03:04.54 | *** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net) |
| 03:05.11 | pdavid | evening all, could anyone lend me a hand with a simple test setup to call out through VP? |
| 03:06.02 | mishehu | argh! sonnofabitch, I installed hylafax 4.3.0 with iaxmodem 0.1.8... I can receive faxes no problem, but the damn thing won't dial out ever. I am using the same exact config I used with hylafax 4.2.5 and iaxmodem 0.1.1... |
| 03:06.23 | mishehu | it wouldn't annoy me so much if I actually got some logging output from hylafax |
| 03:07.19 | onixx | asterboy: any luck with my pastebin ? |
| 03:08.55 | *** join/#asterisk dacleric (n=dacleric@p54821F03.dip0.t-ipconnect.de) |
| 03:14.56 | *** join/#asterisk Gamercjm (n=chris@pool-71-254-182-61.lsanca.fios.verizon.net) |
| 03:15.27 | *** join/#asterisk mike800 (n=Mike800@68-171-34-225.vnnyca.adelphia.net) |
| 03:19.22 | Freman | hmmm... russellb that thing you helped me with before |
| 03:19.26 | Freman | it has a slight problem |
| 03:19.51 | Freman | LOCAL/s@context starts playing the feedback befoe the DIAL answers |
| 03:22.58 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 03:24.44 | *** part/#asterisk mcf3782 (n=mcf3782@adsl-065-012-184-148.sip.asm.bellsouth.net) |
| 03:28.02 | *** join/#asterisk jeebusmobile (n=jeebusmo@29palms-cuda1-68-170-33-192.losaca.adelphia.net) |
| 03:32.59 | mike800 | anyone here? |
| 03:37.38 | shmaltz | mike800, I'm gone |
| 03:37.40 | shmaltz | bye |
| 03:38.38 | orlock | Is anybody here using chan_sccp with 1.2.9.1? |
| 03:40.50 | *** join/#asterisk jcollie[home] (n=jeff@dsl-ppp239.isunet.net) |
| 03:40.57 | *** part/#asterisk jcollie[home] (n=jeff@dsl-ppp239.isunet.net) |
| 03:41.54 | mike800 | if anyone wants to help me figure out why my comptuer isnt finding /dev/zap/ctl pm me... :-) |
| 03:43.05 | *** join/#asterisk Dico_ (n=niko@60.51.217.61) |
| 03:43.21 | Dico_ | hello :) |
| 03:43.49 | mike800 | hi |
| 03:44.22 | ids2500 | mike800: make devices |
| 03:44.27 | ids2500 | mike800: in your zaptel build directory |
| 03:44.39 | ids2500 | and look at readme.udev |
| 03:44.59 | mike800 | ok |
| 03:45.08 | mike800 | i had tried the make devices command |
| 03:45.24 | ids2500 | do a ls /dev/zap/ctl -al |
| 03:45.25 | mike800 | this is the output: build_tools/genudevrules > /etc/udev/rules.d/zaptel.rules |
| 03:45.27 | ids2500 | and make sure it exists |
| 03:45.46 | mike800 | it doesnt |
| 03:45.53 | ids2500 | okay... then it's a udev issue |
| 03:45.58 | mike800 | ok...i'll research it |
| 03:46.07 | ids2500 | udev is the devil :( |
| 03:46.14 | mike800 | hehe...what is it? |
| 03:46.40 | ids2500 | satanic device creation daemon |
| 03:46.45 | xbmodder_lappy | :-? |
| 03:46.48 | Juggie | udev wont exist until the modules load |
| 03:46.48 | mike800 | hahaha |
| 03:46.59 | ids2500 | yeah, that too, make sure your modules are loaded |
| 03:47.00 | ids2500 | lsmod |
| 03:47.14 | ids2500 | you should have zaptel and then whatever zap device you're using [wct4xxp for a te4xp] |
| 03:47.32 | Juggie | and when the modules load the first time they will generate an error |
| 03:47.36 | Juggie | easiest thing imo |
| 03:47.40 | Juggie | cd zaptel-source |
| 03:47.51 | Juggie | make install;make config;service zaptel start |
| 03:47.59 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 03:48.07 | Juggie | or /etc/init.d/zaptel start |
| 03:48.15 | ids2500 | depends on what distribution he has whether that will work or not :-p |
| 03:48.18 | Juggie | depending on your distro, not all of them have the 'service' utility |
| 03:48.33 | mike800 | im using FC5 |
| 03:48.39 | Juggie | service should work then |
| 03:48.41 | ids2500 | see, fedora is satanic |
| 03:48.48 | ids2500 | if you were usign slackware you wouldn't have any problems at all :-) |
| 03:48.49 | mike800 | what do you guys use? |
| 03:48.57 | Juggie | i use centos |
| 03:49.05 | Juggie | regardless... what zaptel card do you have? |
| 03:49.11 | mike800 | TDM400P |
| 03:49.20 | Juggie | does lsmod list zaptel as loaded? |
| 03:49.26 | mike800 | nope |
| 03:49.32 | ids2500 | heh |
| 03:49.35 | Juggie | go into your zaptel source |
| 03:49.39 | mike800 | i installed the source and got no errors |
| 03:49.39 | Juggie | and type make install;make config |
| 03:50.05 | mike800 | ok...lemme try that |
| 03:50.08 | Juggie | let me know when you do that |
| 03:50.17 | mike800 | ok..done |
| 03:50.24 | Juggie | ok, type service zaptel start |
| 03:50.38 | mike800 | failed |
| 03:50.43 | mike800 | Module zaptel not found |
| 03:50.59 | mike800 | missing /dev/zap |
| 03:51.12 | Juggie | type 'modprobe zaptel' |
| 03:51.14 | Juggie | whats the output |
| 03:51.25 | mike800 | Module zaptel not found |
| 03:51.35 | mike800 | :-( |
| 03:51.53 | Juggie | 'uname -a' |
| 03:52.08 | mike800 | im using 2.6.16-1.2122 |
| 03:52.12 | mike800 | FC5smp |
| 03:52.26 | *** join/#asterisk InHisName (n=Prayer@c-68-38-105-1.hsd1.pa.comcast.net) |
| 03:52.27 | *** join/#asterisk bmg505 (n=leon@c1-144-15.rndf.isadsl.co.za) |
| 03:52.59 | Juggie | 'ls -al /lib/modules' |
| 03:53.26 | mike800 | the directory exists |
| 03:53.34 | mike800 | 2.6.16-1.2122_FC5 and FC5smp |
| 03:53.49 | mike800 | (and the old kernel) |
| 03:54.39 | *** join/#asterisk [TK]D-Fender (n=joe@66.11.164.239) |
| 03:55.18 | mike800 | whats weird is that I installed zaptel on this exact configuration with no problems... |
| 03:55.21 | Juggie | paste the output of 'make install' |
| 03:55.24 | Juggie | to pastebin |
| 03:55.56 | InHisName | [TK]D-Fender I have found more info on lack of sound in extensions on sipura |
| 03:56.12 | [TK]D-Fender | InHisName : do tel |
| 03:56.39 | mike800 | http://pastebin.com/766649 |
| 03:57.40 | *** join/#asterisk terrapen_ (n=cjs@mauritius.island.nu) |
| 03:57.55 | InHisName | [TK]D-Fender Calling from line1 to line2 works! BUT calling from line2 to line1 no audio either way ! The sipura values are identical for line/user1 & 2. |
| 03:58.25 | InHisName | The extn definitions in sip & extensions are identical too. |
| 03:58.40 | Juggie | Mike, if you go into /lib/modules/the kernel dir/ |
| 03:58.47 | Juggie | and then its either extra or misc |
| 03:58.51 | Juggie | you should see the modules. |
| 03:58.52 | Juggie | find those. |
| 03:59.13 | *** join/#asterisk jaike (n=a@203.131.137.76) |
| 03:59.17 | [TK]D-Fender | InHisName : You mean you're trying to use the same account on both ports? |
| 03:59.20 | InHisName | I have tried rebooting the linux/asterisk box and recyling power on sipura. No change. |
| 03:59.23 | Juggie | then adress the module directally. |
| 03:59.29 | Juggie | so do insmod zaptel.ko |
| 04:00.19 | InHisName | Account like [1022] and [1021] ? same or different accounts or irrevelent ? |
| 04:02.00 | [TK]D-Fender | InHisName : What are you usings for line1 & line2 ? |
| 04:02.08 | InHisName | sip show peers has both "unmonitored" . At least looks legitamate(sp). |
| 04:02.34 | [TK]D-Fender | InHisName : nevermind that, what users do you have in the Sipura web console for those 2 lines? |
| 04:02.54 | InHisName | Sockets line1 & line2 of sipura have one each analog phone plugged into them. |
| 04:03.27 | InHisName | Username = 1021, 1022 |
| 04:03.55 | InHisName | from sip show users |
| 04:04.08 | [TK]D-Fender | ok, as long as they are differnt accounts... do you get dialtone on each? |
| 04:05.54 | InHisName | Yes, can dial too, other extn rings, I can answer, one case bi-directional sound, other no sound. |
| 04:06.08 | [TK]D-Fender | and does eith of them work on the echo test? |
| 04:06.19 | [TK]D-Fender | eiter* |
| 04:06.21 | [TK]D-Fender | jhsfdklsfdg |
| 04:06.25 | InHisName | I'll try that now... hold on |
| 04:06.26 | [TK]D-Fender | ugh, can't type... |
| 04:09.14 | InHisName | both extns have CLI saying executing echo("SIP/102x-qwer","") in new stack. Interesting, neither has echo. |
| 04:09.58 | InHisName | I'll restore nat=0 for both and try again, forgot I switched it 10 minutes ago. |
| 04:10.42 | *** join/#asterisk somegeek (i=levin@tor/regular/somegeek) |
| 04:11.23 | *** join/#asterisk lilo (i=levin@freenode/staff/pdpc.levin) |
| 04:12.32 | *** join/#asterisk mitcheloc (n=mitchelo@70-32-188-167.lmdaca.adelphia.net) |
| 04:12.33 | InHisName | Well, I got line1 to have imediate sound with no delay in echo test. Line2 no sound. |
| 04:14.20 | *** join/#asterisk samwong (n=sam@210.17.135.46) |
| 04:15.38 | [TK]D-Fender | try your port 1 phone on port 2 just for kicks |
| 04:17.30 | InHisName | Already did, problem stayed with line2 calling line1 using oppisite phones. |
| 04:20.11 | [TK]D-Fender | so its not the phone itself then.... |
| 04:20.12 | [TK]D-Fender | :/ |
| 04:20.21 | [TK]D-Fender | running out of ideas. |
| 04:22.32 | *** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net) |
| 04:22.34 | InHisName | The sipura WAS working fine a week ago with 2 FWD accounts not on *. Now seems like line2 has an issue or something. |
| 04:22.38 | pdavid | hi again all |
| 04:23.09 | pdavid | if i am using a SIP softphone, can i still route my VOIP provider calls through IAX2? |
| 04:23.16 | pdavid | or do i have to use SIP? |
| 04:23.21 | InHisName | Reviewing one MORE time looking for differences in setup line1/2 |
| 04:23.23 | russellb | yes |
| 04:23.25 | mike800 | you can route them through iax |
| 04:23.33 | pdavid | i thought i might |
| 04:23.40 | pdavid | i am using the x-lite linux client |
| 04:23.49 | pdavid | and just got VP connect service |
| 04:23.50 | russellb | that's kind of one of the big points of asterisk :) being technology independent |
| 04:24.02 | pdavid | but for some reason, i cannot call out using IAX2 to connect to VP |
| 04:24.19 | pdavid | but the call works if i use SIP |
| 04:25.17 | pdavid | any thoughts? |
| 04:26.21 | pdavid | (kind of new to this...) |
| 04:28.37 | pdavid | maybe someone could spare a snippet of code to lend me a hand? :) |
| 04:30.04 | xbmodder_lappy | pdavid, 'print "no"' |
| 04:30.05 | [TK]D-Fender | pdavid : * sits between all calls in/out of all different technoligies. |
| 04:30.52 | pdavid | Fender - I understand it does, but don't understand my inability to connect outside using IAX2, with vanilla configs from VP |
| 04:31.07 | [TK]D-Fender | pdavid : So yes, you can use SIP softphones with * and have * bridge the call to an IAX provider while taking in calls from and analog line all at once. |
| 04:31.33 | pdavid | Fender - right, but is there anything special required for me to do that, that perhaps I am missing? |
| 04:31.34 | jaike | VP = voicepulse? they decomssioned their IAX servers |
| 04:31.42 | [TK]D-Fender | pdavid : this is the part where you should have been proactive and started descibing in deatial exactly where things went wrong. |
| 04:31.52 | pdavid | fair enough |
| 04:31.59 | [TK]D-Fender | pdavid : do you get some specific errors in CLI you could tell us about? |
| 04:31.59 | pdavid | jaike: did they? |
| 04:32.17 | jaike | Dear Customer, |
| 04:32.17 | jaike | This a reminder regarding your VoicePulse Connect for Asterisk account: |
| 04:32.17 | jaike | On June 5th, 2006, the legacy servers gwiax-in-01, gwiaxt01, and gwiaxt02 |
| 04:32.17 | jaike | will be decomissioned |
| 04:32.25 | jaike | oooopps...sorry |
| 04:32.29 | [TK]D-Fender | pdavid : Yeah, they jsut changed from their old ones toa new set about 2 days ago. |
| 04:32.33 | pdavid | Fender: no errors, I get the first line of my [outgoing] context correctly setting my callerid |
| 04:32.42 | pdavid | Fender: yes, connect01/2 i think |
| 04:32.45 | [TK]D-Fender | pdavid : pastebin a call attempt |
| 04:32.47 | [TK]D-Fender | ~pb |
| 04:32.53 | jbot | [pb] a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 04:32.54 | pdavid | ok |
| 04:33.06 | jaike | try checking their site for the new servers |
| 04:33.16 | *** join/#asterisk scardinal (n=supreme@port816.ds1-suoe.adsl.cybercity.dk) |
| 04:34.03 | pdavid | http://pastebin.com/766680 |
| 04:34.33 | [TK]D-Fender | pdavid : Sounds like you don't even have a DIALPLAN in place... pastebin your extensions.conf |
| 04:35.06 | [TK]D-Fender | pdavid : it stops right after setting the callerID which says you've probably missed something extremely easy to stop & correct |
| 04:35.08 | pdavid | http://pastebin.com/766682 |
| 04:35.12 | [TK]D-Fender | spot* |
| 04:35.25 | pdavid | whats odd is, if i uncomment the SIP lines, i can place the call |
| 04:35.41 | pdavid | Fender: this is just the vanilla VP config files |
| 04:35.42 | [TK]D-Fender | pdavid : You have no priority 2 for that exten... and YES, they must follow.... |
| 04:36.00 | [TK]D-Fender | pdavid : and in there are instruction to UNCOMMENT some of the needed lines... |
| 04:36.13 | pdavid | Fender: I thought it was IF i wanted to do LCR |
| 04:36.16 | pdavid | but lemme try it |
| 04:36.43 | [TK]D-Fender | pdavid : You need to do the basic learning of *.... if this part escapes you go download THE BOOK. |
| 04:36.44 | [TK]D-Fender | ~book |
| 04:36.46 | jbot | extra, extra, read all about it, book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 04:37.01 | [TK]D-Fender | because right now you have NO dial-out at all |
| 04:37.12 | pdavid | Fender: that was all ti was |
| 04:37.14 | pdavid | *it |
| 04:37.37 | pdavid | Fender: I AM trying to learn the basics, little by little |
| 04:37.45 | pdavid | I appreciate the help with that little tidbit |
| 04:37.56 | *** join/#asterisk hads|home (n=hads@mail.nice.net.nz) |
| 04:38.04 | pdavid | wasn't aware I couldnt skip priorities at all |
| 04:38.10 | pdavid | :( |
| 04:38.28 | pdavid | thank you very much! :D |
| 04:38.29 | russellb | Goto? |
| 04:39.20 | [TK]D-Fender | pdavid : Np, one step at a time... do DL the book and print it if you can.... its a great place to start. |
| 04:39.52 | jaike | pdavid: i highly recommend oreilly's asterisk book |
| 04:39.59 | pdavid | I didn't even know they released that book on a CCL |
| 04:40.02 | pdavid | sweet |
| 04:40.06 | pdavid | really, thanks a ton guys |
| 04:40.12 | orlock | Can anybody reccomend a stabke and functional sccp driver? |
| 04:40.14 | [TK]D-Fender | pdavid : You'll need to focus on contexts, extens, and the various applications that make up the dial-plan. All the connectivity in the world won't help you when you don't have a dial-plan to patch the bits together |
| 04:40.33 | [TK]D-Fender | orlock : Cisco CCM :) |
| 04:40.37 | pdavid | Fender: right, I am trying baby steps to understand it all. This particular step was getting a call to go outside |
| 04:40.47 | pdavid | Fender: any thoughts on NAT traversal for IAX2? |
| 04:40.51 | [TK]D-Fender | pdavid : Well i suspect you are quite close to that... |
| 04:41.20 | jaike | iax and nat? no problems |
| 04:41.26 | [TK]D-Fender | pdavid : Generally not an issue at all. SIP has never posed a problem for most either once they know what they're doing... its fairly easy |
| 04:42.38 | pdavid | does it get covered in the O'Reilly book at all? |
| 04:42.48 | pdavid | or should I just drop the * server in a DMZ? |
| 04:42.53 | jaike | voip concepts even |
| 04:43.28 | [TK]D-Fender | pdavid : its 5 lines :) don't worry about it.... |
| 04:43.46 | orlock | [TK]D-Fender: Dang, so you cant reccomend anything? |
| 04:43.50 | [TK]D-Fender | pdavid : And yeah it has to... its too common and necessary to skip. |
| 04:44.04 | pdavid | or is it as simple as port forwarding a small range to the * server? |
| 04:44.08 | [TK]D-Fender | orlock : Ditch Cisco, buy Polycom! (you asked for it) |
| 04:44.17 | [TK]D-Fender | pdavid : Almost that easy... |
| 04:44.21 | dec | so has the svn trunk version of asterisk moved away from asterisk-config ? |
| 04:44.44 | orlock | [TK]D-Fender: yeah, yeah.. but we already have a few dozen cisco phones |
| 04:44.51 | orlock | <PROTECTED> |
| 04:44.56 | jaike | sip uses 5060 and 10000-20000 for rtp (asterisk default config) |
| 04:44.57 | dec | oh, ignore me. :) |
| 04:45.03 | orlock | and skinny makes asterisk segfault |
| 04:45.26 | [TK]D-Fender | orlock : Reflash them to SIP... |
| 04:45.36 | pdavid | jaike: how about iax2? |
| 04:45.37 | orlock | not all of them support SIP |
| 04:45.52 | jaike | iax2 just uses 4569 so you wont have any problems with NAT |
| 04:45.56 | orlock | also, i'd like t be able to go back to our current voip provider who uses sccp if/when asterisk fucks up |
| 04:46.32 | pdavid | jaike: thanks for that! |
| 04:47.16 | jaike | np |
| 04:49.04 | [TK]D-Fender | pdavid : Lets just say the only use you should see for IAX2 is to trunk multiple calls to a VoIP provider to save on bandwidth and that if you have a properly set up system, you'll be 100% SIP most likely. |
| 04:49.50 | pdavid | Fender: if I wanted to trunk 2-3 calls to a VOIP at once, would i be better off with IAX2? |
| 04:51.14 | [TK]D-Fender | pdavid : Yes, as trunking with IAX2 saves on UDP packet overhead. The savings can be substantial. |
| 04:51.32 | orlock | Unless your voip provider is also your bandwidth provider |
| 04:51.35 | pdavid | Fender: how about codec used? any suggestions there as well? |
| 04:51.38 | orlock | in which case.. who cares! :) |
| 04:52.13 | [TK]D-Fender | orlock : We're not talking about bandwidth in a billable capacity as much as maximum transmit potential. |
| 04:52.18 | *** join/#asterisk droops (n=droops@adsl-065-005-212-128.sip.jan.bellsouth.net) |
| 04:52.34 | [TK]D-Fender | pdavid : Depends. The general answer is "the highest quality you can support" |
| 04:52.52 | [TK]D-Fender | pdavid : Which is normally G.711 (u/a) |
| 04:53.01 | orlock | [TK]D-Fender: ahh yeah |
| 04:53.05 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 04:53.14 | orlock | my head is going to pop |
| 04:53.14 | pdavid | Fender: is that the ulaw/alaw? |
| 04:53.21 | orlock | 3pm on a thursday |
| 04:53.35 | [TK]D-Fender | pdavid : Yes |
| 04:54.03 | [TK]D-Fender | orlock : I've got next thursday open if you'd like to book an appointment :) |
| 04:54.24 | orlock | heh |
| 04:54.42 | orlock | i found a server i'd completely forgotten about, which is good |
| 04:54.44 | [TK]D-Fender | orlock : And a brand new katana that needs some "breaking in" :D |
| 04:54.53 | orlock | amd64 3000+ with 2 gig of ram, woo |
| 04:55.08 | orlock | [TK]D-Fender: cool! |
| 04:55.13 | orlock | you can break it in on my shinai |
| 04:55.15 | orlock | and bokken |
| 04:55.32 | orlock | then move onto katana vs katana |
| 04:55.43 | orlock | i'm a bit rusty though |
| 04:55.46 | pdavid | Fender, jaike, I appreciate the help this evening, and thanks for your patience! |
| 04:55.47 | [TK]D-Fender | orlock : Shinai would be mincemeat.... bokken is a chunk of WOOD, and would likey deform my blade... |
| 04:56.08 | pdavid | im off to fool with this a bit more, and read this oreilly book |
| 04:56.14 | pdavid | good night, and thanks again |
| 04:56.16 | [TK]D-Fender | orlock : Do you study? |
| 04:56.19 | orlock | hmm.. memories of counting from one to ten.. striking each time |
| 04:56.31 | orlock | times twenty :) |
| 04:56.40 | orlock | [TK]D-Fender: i used to |
| 04:56.51 | orlock | i got over the baggy blue pants though |
| 04:57.17 | orlock | [TK]D-Fender: what sort of katana? |
| 04:57.21 | [TK]D-Fender | orlock : I recently start Katori Shinto and am looking for an iaido dojo on the side to supplement the iaijutsu aspect and take up tameshigiri |
| 04:57.28 | orlock | mines just a disposal store al cheapo decroative 440 steel one |
| 04:57.52 | orlock | i just did kendo, but the dojo also did iaido |
| 04:57.53 | [TK]D-Fender | orlock : Oni Forge Bushi. Heres the link : http://www.oniforge.com/3001.html |
| 04:58.14 | orlock | as well as.. i forget its ame, the stick fighting, and the long ole/spear fighting that was mainly females |
| 04:58.14 | [TK]D-Fender | orlock : And my personal review : http://forums.swordforum.com/showthread.php?s=&threadid=67812 |
| 04:58.34 | [TK]D-Fender | 440? eek..... wouldn't use that for practice even... |
| 04:58.40 | orlock | heh, yeah |
| 04:58.58 | orlock | but at least its 440 and not just chrome plated shit |
| 04:59.12 | [TK]D-Fender | Stick would be eskrima if you're talking filipines. |
| 04:59.27 | orlock | all japanese |
| 04:59.38 | orlock | only purpose built dojo in australia iirc |
| 04:59.47 | [TK]D-Fender | well the chrome ones are actually aluminum/zinc and SAFE for iaido as opposed to 440's brittleness... |
| 05:00.03 | orlock | http://www.kendovictoria.asn.au/ |
| 05:00.33 | orlock | i never actually saw blades connect in iaido though.. isnt it more practicing the forms? |
| 05:00.36 | [TK]D-Fender | orlock : Nice looking place... |
| 05:01.19 | [TK]D-Fender | orlock : Yeah it is. The same applies to the iaijutsu I lean in Katori, but the fun stuff is kenjutsu w/ blade on blade :) |
| 05:02.27 | orlock | Jodo and Naginata |
| 05:02.55 | orlock | Jodo is with the sticks and Naginata is the long pole/spear i think |
| 05:03.06 | [TK]D-Fender | orlock : Katori contains 4 kata of sword on sword, and 4-5 of bo/sword, naginata/sword, and iaijutsu before the esoteric stuff... |
| 05:03.36 | [TK]D-Fender | naginata is a polearm class weapon, much like a halberd |
| 05:03.42 | orlock | yup |
| 05:04.05 | orlock | pole/spear thing, yeah, like a halbard, but almost with a machete on the end |
| 05:04.26 | [TK]D-Fender | orlock : Either way, not my style :) I'm quite happy with my new kat, and will eventually move further up with a folded bladed, perhaps custom. |
| 05:04.57 | [TK]D-Fender | orlock : Nihonto is a DISTANT dream. |
| 05:05.10 | orlock | i remember having parties when i was living at my parens out in the burbs, when they went away |
| 05:05.20 | orlock | and i'd g and by asack or two of potatoes and oranges |
| 05:05.25 | orlock | standat one end of the backyard |
| 05:05.33 | orlock | and get them to start pitching :) |
| 05:07.00 | orlock | that was before i started kendo, and was the only real thing i could do with i |
| 05:07.02 | orlock | was fun |
| 05:07.20 | [TK]D-Fender | heh |
| 05:08.18 | [TK]D-Fender | I'm picking this up late in the game, but enjoying it nonetheless and there are a number of really cute women in there... looking good so far... |
| 05:09.52 | orlock | i think katanas are illegal in australia now |
| 05:09.56 | dec | Hi all - in the latest asterisk trunk from svn, the makefile runs configure and then exits out so you have to run make again to compile -- is there a reason for this exit? |
| 05:10.01 | orlock | after a fair few attacks with them |
| 05:10.09 | mitcheloc | Fender, huh? women? |
| 05:10.28 | orlock | mitcheloc: chicks with swords! |
| 05:10.42 | [TK]D-Fender | orlock : No, far enough from it... I know a few reviewers there... |
| 05:11.34 | [TK]D-Fender | orlock : I find elegence sexy, and few things show it like how they handle a blade... |
| 05:11.40 | orlock | [TK]D-Fender: must have exemptions for being collectors |
| 05:12.10 | [TK]D-Fender | orlock : Banning swords is utter BS... cars kill So many more people..... |
| 05:12.23 | orlock | [TK]D-Fender: there were a few attacks with them in the ast few years, one of them 2 or 3 people died after jumping into a river at night to avoid an asian gang.. they drowned |
| 05:12.51 | mitcheloc | would anyone use a free asterisk monitoring service? |
| 05:12.55 | orlock | asian gangs would pack them in their cars, around clubs, etc, and if they got hassled, go to their cars, grab swords, hang around nightclub |
| 05:12.57 | mitcheloc | sort of to keep tabs if your box was online or not? |
| 05:13.03 | [TK]D-Fender | orlock : Yeah and more people die to attacks with pocket knives and you don't see them getting banned... concealable and cheap.... |
| 05:13.09 | orlock | yeah |
| 05:13.30 | orlock | i was charged for carrying a pocket knife over a decade ago. |
| 05:13.36 | orlock | now i just carry a leatherman |
| 05:13.55 | orlock | <PROTECTED> |
| 05:13.56 | orlock | Parsing /etc/asterisk/asterisk.conf |
| 05:13.56 | orlock | Segmentation fault |
| 05:15.05 | orlock | Joy.. crashes with skinny, refuses to load sccp |
| 05:16.14 | *** join/#asterisk Beighto (n=Kry5ta1@adsl-70-133-76-34.dsl.scrm01.sbcglobal.net) |
| 05:16.16 | *** join/#asterisk Cyberecho79 (n=chatzill@c-67-182-166-94.hsd1.ca.comcast.net) |
| 05:16.36 | [TK]D-Fender | Ok, WAY late here... back on tomorrow... |
| 05:17.06 | Beighto | hey hows it goin |
| 05:31.00 | *** join/#asterisk litecode (n=andrewb@12-215-201-171.client.mchsi.com) |
| 05:31.46 | dlynes_office | dec: ummmm....i dont' remember it working like that |
| 05:31.55 | dlynes_office | dec: is that a recent change? |
| 05:32.13 | litecode | I keep getting syntax errors on this line: Set(LOOPCOUNT=$[${LOOPCOUNT} + 1]) it says "ast_expr2.fl:183 ast_yyerror: ast_yyerror(): syntax error: syntax error, unexpected TOK_PLUS, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input: |
| 05:32.13 | litecode | <PROTECTED> |
| 05:32.25 | dlynes_office | litecode: lose the spaces |
| 05:32.34 | litecode | thanks, will try. |
| 05:32.45 | dlynes_office | litecode: the asterisk expression parser doesn't like whitespace |
| 05:33.06 | litecode | i figured as much |
| 05:33.12 | litecode | but the docs are obviously out of date. |
| 05:33.41 | terrapen_ | yeah, the whitespace limitation is lame |
| 05:33.44 | dlynes_office | litecode: it also might not like the '+1' |
| 05:33.55 | litecode | i am figuring it is not seeing hte LOOPCOUNT |
| 05:34.06 | litecode | because the actual "input" should be 0+1 |
| 05:34.06 | stephane_ | jour |
| 05:34.14 | dlynes_office | stephane_: soir |
| 05:34.39 | litecode | but earlier, i'm doing a Set(LOOPCOUNT=0) |
| 05:34.56 | dlynes_office | terrapen_: especially seeing as how they're using lexx and yacc |
| 05:35.16 | dlynes_office | terrapen_: it wouldn't take much different grammar to allow for whitespace as well |
| 05:35.59 | litecode | should I use Set or SetVar? |
| 05:36.21 | dlynes_office | SetVar's deprecated, so obviously Set() :) |
| 05:36.46 | litecode | i figured it out. thanks |
| 05:36.58 | litecode | dlynes_home, it was merely a goof on my part. |
| 05:37.15 | dlynes_office | oh? |
| 05:37.45 | dec | dlynes_office - I don't know, but its definately in the latest trunk. after make config.status there's an "@exit 1" and it tells you to re-run make. |
| 05:37.58 | dec | its okay, it still works, I was just trying to automate it and hit a hurdle. |
| 05:38.03 | dlynes_office | dec: why are you running make config.status? |
| 05:38.14 | dec | dlynes_office - I'm not running it manually. but when you run ' |
| 05:38.16 | dec | whoops |
| 05:38.29 | dlynes_office | dec: you do know that the configure in trunk is not a GNU autoconfigure script, right? |
| 05:38.36 | dec | dlynes_office - I'm not running it manually. but when you run 'make', it builds config.status |
| 05:38.44 | dec | dlynes_office - yes I know |
| 05:38.57 | dlynes_office | dec: ok, just checking...it's a bit confusing |
| 05:38.57 | terrapen_ | is realtime call monitoring CPU intensive? |
| 05:39.33 | dec | dlynes_office - yeah, a little confusing.. but its okay :) |
| 05:39.42 | terrapen_ | errr i meant recording |
| 05:40.06 | dlynes_office | dec: it's still a lot better than just a simple makefile, because you can do a menu-based configuration of what modules you want to build |
| 05:40.24 | dlynes_office | dec: then you can also see at a glance what modules you don't have dependencies for as well |
| 05:40.40 | dlynes_office | dec: before, you built it, and you got some modules, but you never knew you were missing any :) |
| 05:40.44 | dec | dlynes_office - yep, its nicer than a straightforward makefile. :) |
| 05:41.14 | terrapen_ | i wish the wiki would not open a new window for searches |
| 05:42.58 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 05:43.48 | *** join/#asterisk Eggplant (i=No@dsl-216-155-213-242.cascadeaccess.com) |
| 05:44.26 | yxa | anyone tell me what's wrong here: http://pastebin.com/766726 |
| 05:45.29 | *** part/#asterisk Juggie (i=agony@CPE00c049d9f271-CM00137186c8d8.cpe.net.cable.rogers.com) |
| 05:46.32 | *** join/#asterisk kristalino (n=kristali@LNeuilly-152-21-126-73.w193-253.abo.wanadoo.fr) |
| 05:46.45 | dlynes_office | yxa: you've got an incorrect dbname, username, or password for mysql |
| 05:47.35 | dlynes_office | yxa: or the table that it's trying to use hasn't been created yet |
| 05:49.30 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 05:51.24 | terrapen_ | it would be cool to modify the Dictate() command to e-mail the recorded file when done |
| 05:51.52 | terrapen_ | my boss would love that...dictate a letter for the secretary and have the .wav automatically sent to her |
| 05:53.01 | dlynes_office | terrapen_: yeah...i've got a similar script i need to write |
| 05:53.15 | dlynes_office | terrapen_: record a conversation and then email that recording to the manager |
| 05:53.35 | litecode | is there anything currently available that can do call queueing on a generic extension? sat a SIP/ extension? instead of an agent. |
| 05:54.13 | dlynes_office | litecode: well, i'm sure it would be easy enough to do that via an extensions.conf script, or via an agi script |
| 05:54.25 | litecode | yeah.. that's what i'm thinking |
| 05:54.35 | litecode | don't want to re-invent the wheel. |
| 05:54.44 | litecode | i have already done it in extensinos.conf |
| 05:54.51 | litecode | but it's not ordered. |
| 05:54.55 | dlynes_office | writing your own extensions.conf script isn't reinventing the wheel :) |
| 05:55.00 | dlynes_office | ordered? |
| 05:55.04 | litecode | fifo |
| 05:55.09 | dlynes_office | ah |
| 05:55.38 | litecode | any ideas? |
| 05:57.30 | dlynes_office | well, like i said...you could write as an agi script |
| 05:57.50 | dlynes_office | I'm about 90% sure nobody's written something that'll do that yet, but i could be wrong |
| 05:58.27 | litecode | i'd like to do that with python if possible. |
| 05:58.53 | dlynes_office | litecode: well, i'm sure if python has an agi library, you could do it with python |
| 05:59.06 | litecode | thanks, i'll try it out. |
| 06:00.49 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 06:01.05 | dlynes_office | good morning, olle |
| 06:01.32 | oej | Morning |
| 06:01.48 | *** join/#asterisk Shoragan (n=shoragan@d072.apm.etc.tu-bs.de) |
| 06:03.24 | asterboy | does anyone have zasterisk working...I need a fix. |
| 06:03.48 | dlynes_office | what is it? |
| 06:04.13 | asterboy | ~zasterisk |
| 06:04.14 | jbot | rumour has it, zasterisk is a cool implementation of Zork and Asterisk. Play Zork over the phone! Get it here: http://uc.org/files/2005-08-07/zasterisk-0.1.tar.gz |
| 06:04.22 | dlynes_office | lol |
| 06:04.29 | dlynes_office | zork over the phone? |
| 06:04.36 | dlynes_office | that's a new one |
| 06:04.53 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 06:05.31 | dlynes_office | so does it use festival, or something? |
| 06:06.12 | asterboy | ya, using festival...very cool. |
| 06:06.34 | asterboy | wonder what it does in the echo chamber? |
| 06:06.40 | mitcheloc | ~zork |
| 06:06.41 | jbot | zork is probably a good game derived from adventure, created by infocom |
| 06:07.10 | dlynes_office | ~wiki zork |
| 06:07.29 | orlock | advent |
| 06:07.46 | orlock | that tought me to type |
| 06:07.56 | dlynes_office | and it wurked so well |
| 06:08.32 | orlock | i didnt say anything about speling :) |
| 06:08.37 | dlynes_office | lol |
| 06:11.29 | asterboy | ls |
| 06:11.49 | jayk- | rm |
| 06:12.51 | asterboy | rm -rf /* |
| 06:13.23 | *** part/#asterisk neilbags (n=neilbags@149.171.94.134) |
| 06:13.35 | asterboy | my zttest scores are bad |
| 06:13.37 | asterboy | est: 100.000000 -- Worst: 93.554688 -- Average: 99.774891 |
| 06:14.24 | dlynes_office | cool beans |
| 06:15.22 | asterboy | cool runnings |
| 06:15.34 | litecode | agi vs fastagi? |
| 06:15.49 | asterboy | sure goes to show that not all motherboards can handle pci very well. |
| 06:16.36 | asterboy | its a 1.3G AMD and 256Mb |
| 06:17.01 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 06:18.03 | drray | I've had great luck with my asrock motehrboard and asterisk |
| 06:18.09 | dlynes_office | Yeah, and I've got a pos VIA motherboard |
| 06:18.09 | drray | it's a cheap asus brand |
| 06:18.30 | dlynes_office | I've got four of the damned things cause our last asterisk admin thought they were the greatest thing since sliced bread |
| 06:18.56 | drray | they probably were for running suse or whatever he wanted them for before asterisk |
| 06:19.06 | asterboy | what are the zttest on those like? |
| 06:19.11 | dlynes_office | drray: nah...he bought them specifically for asterisk |
| 06:19.21 | dlynes_office | asterboy: better than your averages :p |
| 06:19.27 | dlynes_office | asterboy: but not much better |
| 06:19.46 | dlynes_office | asterboy: and you have to share an irq...only one pci slot, and it shares an irq with the network card |
| 06:19.55 | dlynes_office | asterboy: and the pos doesn't even support APIC |
| 06:19.56 | asterboy | yuk |
| 06:20.16 | drray | --- Results after 12 passes --- |
| 06:20.16 | drray | Best: 99.987793 -- Worst: 99.975586 -- Average: 99.985758 |
| 06:20.29 | drray | and that's with a crappy tor2 card from govarion |
| 06:20.47 | dlynes_office | whatever govarion is |
| 06:21.00 | asterboy | what is a good motherboard to get those scores higher? |
| 06:21.32 | dlynes_office | asterboy: probably any brand name |
| 06:21.45 | drray | what I did was buy a MB and try it |
| 06:21.49 | drray | and then lock into it |
| 06:22.05 | asterboy | not any brand, cause my Gigabyte sucks |
| 06:22.08 | drray | like I said, I like the asrock MB |
| 06:22.14 | drray | it was $40 |
| 06:22.21 | asterboy | but the scores are not that great |
| 06:22.22 | drray | it only has two pci slots |
| 06:22.25 | dlynes_office | gigabyte hasn't been a decent mobo for about 5 or 6 years now |
| 06:22.32 | dlynes_office | erm about ten years i mean |
| 06:23.04 | dlynes_office | the last good mobo they had was for a 386dx-40 |
| 06:23.16 | asterboy | Tyan |
| 06:23.24 | dlynes_office | yeah...tyan rocks |
| 06:23.43 | dlynes_office | drray: does the asrock support apic? |
| 06:24.15 | dlynes_office | We're trying to find a small footprint mobo for use in a compact case |
| 06:24.29 | dlynes_office | So that it doesn't look like a computer to the offices we're trying to sell it to |
| 06:24.30 | drray | apic being mutli core? |
| 06:24.46 | dlynes_office | apic meaning you have 32-bit irq's |
| 06:24.54 | drray | I dunno |
| 06:25.02 | drray | I just put the card in and it worked |
| 06:25.06 | drray | and stopped carring |
| 06:25.08 | dlynes_office | Advanced PIC |
| 06:25.11 | drray | er, caring |
| 06:25.16 | *** join/#asterisk Freman (n=twitsrus@jaguar.wbs.net.au) |
| 06:25.26 | dlynes_office | The old XT-PIC would only allow you to have up to 16 IRQ's |
| 06:25.48 | drray | I don't believe I've seen a number better than 16 |
| 06:25.51 | drray | er, greater |
| 06:26.00 | dlynes_office | drray: so you've never used APIC then |
| 06:26.13 | dlynes_office | APIC's a solution that helps you get around the IRQ sharing problem some mobo's have |
| 06:26.37 | drray | the highest is 23 |
| 06:26.48 | drray | from device manager |
| 06:27.07 | drray | but correct, I've never sought out apic |
| 06:27.21 | dlynes_office | drray: your highest irq is 23? |
| 06:27.26 | *** join/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 06:27.36 | dlynes_office | drray: then you're using APIC |
| 06:27.50 | drray | in device manager, viewing by connection on IRQ's it goes 23 |
| 06:27.52 | drray | currently |
| 06:28.19 | dlynes_office | If you weren't using APIC, you wouldn't see anything higher than 16 |
| 06:28.42 | drray | learn something new everyday |
| 06:29.03 | dlynes_office | drray: anyways, most linux distributions ship with apic enabled in the kernel |
| 06:29.25 | dlynes_office | drray: so if your motherboard/chipset support it, it'll just use it automatically |
| 06:29.28 | drray | I try to avoid getting to that level of involvment |
| 06:29.37 | drray | :) |
| 06:29.41 | dlynes_office | ah |
| 06:29.58 | dlynes_office | yeah...i've been compiling my own kernels for linux now for probably 12 years |
| 06:30.00 | drray | I mean, I put the card in, and got my zttest |
| 06:30.11 | drray | and forgot all about it |
| 06:30.19 | drray | I tend to treat hardware like running water |
| 06:30.26 | dlynes_office | heh |
| 06:30.57 | drray | I did however buy 10 MB's for storage |
| 06:31.20 | dlynes_office | 10MB's for storage? |
| 06:31.26 | drray | 10 motherboads |
| 06:31.29 | dlynes_office | damn? what can you store on that? 2 pictures? |
| 06:31.32 | dlynes_office | oh |
| 06:31.54 | drray | I assume putting a second govarion card in will screw things up |
| 06:32.09 | dlynes_office | not necessarily |
| 06:32.20 | dlynes_office | some people are running four quad t1 cards in one machine |
| 06:32.30 | drray | I'm eyeballing that 8 port sangoma |
| 06:32.36 | dlynes_office | heh |
| 06:32.36 | litecode | dlynes_home, looks like i'm going to use FastAGI and AMI to do this... I'll have a call queue AGI that sits and collects calls, then I'll have a queue manager that sits and watches for any queued calls, when it sees one, it will try the outside number and if busy, not bridge the call, but otherwise bridge it. Is the AMI the way to control asterisk like this? |
| 06:32.38 | dlynes_office | it's not out yet |
| 06:32.44 | drray | there are only 2 pci slots in the mb |
| 06:32.45 | dlynes_office | won't be out until June 18th |
| 06:33.17 | drray | I've been very happy with the tor2 card |
| 06:33.29 | drray | but then again, I was happy with teh zhone channel bank for a bit too |
| 06:33.39 | dlynes_office | drray: well, if you like plug and play |
| 06:33.47 | drray | heh |
| 06:33.47 | dlynes_office | drray: i don't know much you'll like sangoma |
| 06:33.49 | dlynes_office | :) |
| 06:33.59 | drray | well, it can't be harder than getting the zplex configured |
| 06:34.10 | drray | or the ivtv drivers working |
| 06:34.12 | dlynes_office | well, my system is so incompatible |
| 06:34.17 | drray | for the hauppauge 350 |
| 06:34.30 | dlynes_office | Sangoma's had to call in their head tech or whatever to take a look at my problem |
| 06:35.56 | terrapen_ | how is ivtv these days |
| 06:36.06 | drray | it's come a long way |
| 06:36.13 | terrapen_ | when it tried it with my a hauppage 350 about 3 years ago, it was dreadful |
| 06:36.21 | dlynes_office | terrapen_: is that the new tv show where you watch iv users inject themselves? |
| 06:36.23 | drray | it was picky a year ago |
| 06:36.29 | terrapen_ | dlynes: yeah |
| 06:36.44 | drray | you had to decompress the windows driver into ivtv module |
| 06:36.58 | terrapen_ | yep |
| 06:36.59 | dlynes_office | so what is it? |
| 06:37.04 | terrapen_ | ivtv.sf.net |
| 06:37.16 | drray | tvtuner/capture driver for linux |
| 06:37.22 | terrapen_ | err http://ivtvdriver.org/ |
| 06:37.31 | dlynes_office | ah...does it work better than v4l, or v4l2? |
| 06:37.46 | terrapen_ | i don't even watch tv these days so theres not much point in building another mythtv box |
| 06:37.55 | drray | well, I don't use mythtv |
| 06:37.56 | terrapen_ | it would be cool to have a TV-based SIP phone though |
| 06:37.57 | dlynes_office | I'm having a hell of a time trying to get my Phillips 7334 or whatever to work |
| 06:37.59 | terrapen_ | (if one existed) |
| 06:38.25 | drray | http://petscii.livejournal.com/22457.html |
| 06:38.25 | terrapen_ | i want a videophone, just like the Mooninites |
| 06:38.27 | drray | I use that |
| 06:38.27 | x86 | terrapen: mythtv + eyebeam? |
| 06:39.02 | terrapen_ | yeah, that would probably work x86 |
| 06:39.47 | terrapen_ | We need to send a message-er |
| 06:39.51 | *** join/#asterisk af_ (n=af@ip-164-240.sn2.eutelia.it) |
| 06:39.54 | terrapen_ | I mean, a "message comma err" |
| 06:40.04 | terrapen_ | the mooninites rule |
| 06:40.09 | drray | there is no reason that I can think of that asterisk could not stream tv |
| 06:40.15 | drray | I could be an idiot |
| 06:40.19 | *** join/#asterisk MACscr (i=user@adsl-70-235-7-81.dsl.peoril.sbcglobal.net) |
| 06:40.22 | MACscr | hello everyone |
| 06:40.28 | MACscr | wow, just found this place |
| 06:40.30 | MACscr | cool |
| 06:43.31 | xbmodder_lappy | how do I only play like 88 seconds of a video, and not the whole thing? |
| 06:45.28 | drray | what are you talking about? |
| 06:47.14 | litecode | if i set one call on musiconhold, can i go to the AMI and dial up another call, and then bridge the two together (after stopping the musiconhold) using the ami? |
| 06:48.31 | terrapen_ | We are the Mooninites from the inner core of the moon |
| 06:48.39 | litecode | is google freaking out for anybody elsE? |
| 06:48.41 | terrapen_ | Our race is hundreds of years ahead of yours |
| 06:48.42 | MACscr | lol |
| 06:48.49 | MACscr | its working fine for me |
| 06:49.19 | litecode | i'm getting a nice google Server Error page |
| 06:49.23 | litecode | on any search |
| 06:49.25 | litecode | weird, yo |
| 06:49.38 | asterboy | ya, I gotta say that TV is sure going down hill these days. |
| 06:49.39 | litecode | all good now :P |
| 06:49.41 | drray | considering we had a 1000 year dark age period due to assholes, being 100's of years ahead of us does not seem that impressive |
| 06:49.45 | MACscr | if anyone is bored and wants to give me some advice. I just posted here: |
| 06:49.47 | MACscr | http://www.pbxinfo.com/forums/showthread.php?p=93283#post93283 |
| 06:49.48 | asterboy | nothing really to look forward to watching. |
| 06:50.06 | MACscr | you can reply here or there, whatever works for you =P |
| 06:50.14 | asterboy | and the new X-men sucked |
| 06:50.14 | MACscr | either way, i appreciate it |
| 06:50.35 | *** join/#asterisk yxa (n=diablo@58.185.90.101) |
| 06:50.38 | terrapen_ | ATHF, Curb Your Enthusiasm...there are a few good shows left |
| 06:50.57 | asterboy | Clifford the Big Red Dog? |
| 06:51.15 | drray | I like The WIre |
| 06:51.34 | asterboy | Actually, I stated recording "How William Shattner changed the world" |
| 06:51.39 | drray | but you can get it on DVD and watch at yor leisure |
| 06:51.41 | asterboy | ReplayTV |
| 06:52.03 | asterboy | and Nova and some survival shows. |
| 06:52.18 | asterboy | but just about everything is downloadable via torrent or YouTube |
| 06:52.21 | terrapen_ | ok bedtime |
| 06:52.24 | drray | Nova has gone downhill |
| 06:52.31 | asterboy | so I'm actually thinking of killing cable |
| 06:52.41 | asterboy | ya Nova is loosing its edge |
| 06:52.48 | terrapen_ | hehe |
| 06:52.49 | terrapen_ | http://www.youtube.com/watch?v=HUhrkuLo8y4&search=mooninites |
| 06:53.03 | drray | the typhoid mary one was the last interesting one |
| 06:53.18 | drray | tehy spend a lot of time at the north pole/mountain climbers |
| 06:53.20 | asterboy | Discovery Channel seems like a paid advertisement for up&coming tech sellers |
| 06:53.20 | drray | for some reason |
| 06:53.38 | drray | WHEN MAILBOXES EXPLODE |
| 06:53.42 | drray | it could happen |
| 06:53.46 | asterboy | night |
| 06:53.48 | terrapen_ | Commence re-mooning at once! |
| 06:54.11 | MACscr | ROFL |
| 06:56.18 | *** join/#asterisk AltnTab (n=ecs@nrjsoft13.networx-bg.com) |
| 06:56.52 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 06:57.45 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 07:05.45 | xbmodder_lappy | http://www-306.ibm.com/e-business/ondemand/us/advertising/advert_linux.shtml for these three commericals what would you say is the best for intro to a speech? |
| 07:06.22 | *** join/#asterisk alucard064 (n=vircuser@ABayonne-152-1-62-154.w83-193.abo.wanadoo.fr) |
| 07:06.38 | alucard064 | re all |
| 07:06.53 | alucard064 | someone as already use asterisk java |
| 07:07.21 | x86 | asterisk java? |
| 07:07.29 | alucard064 | yes |
| 07:07.50 | alucard064 | http://www.asteriskjava.org/latest/tutorial.html |
| 07:07.59 | alucard064 | here the site? |
| 07:08.58 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 07:10.18 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 07:12.11 | *** join/#asterisk FreezeS (n=Gladius@82.208.156.94) |
| 07:23.39 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 07:28.02 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 07:28.18 | cekc | anyone awake? |
| 07:28.50 | x86 | nevar! :) |
| 07:29.16 | cekc | know anything about getting sip to work behind a nat? |
| 07:30.12 | dlynes_home | cekc: there's plenty of information on that on the wiki |
| 07:30.32 | dlynes_home | cekc: the two biggest things you need are nat=yes and canreinvite=no in your sip.conf |
| 07:30.53 | cekc | i got that |
| 07:31.09 | *** join/#asterisk Shaun2222 (n=ndci@ip68-5-63-223.oc.oc.cox.net) |
| 07:31.26 | dlynes_home | then, what's the problem? |
| 07:31.33 | cekc | where is the wiki? |
| 07:31.38 | dlynes_home | ~wikis |
| 07:31.46 | jbot | i guess wikis is http://www.voip-info.org |
| 07:31.51 | dlynes_home | ~docs |
| 07:31.52 | jbot | docs is probably 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 |
| 07:32.00 | Shaun2222 | each phone thats conneacted via SIP is there a CALLERID(num) value in the database or somwhere for each phone? |
| 07:32.25 | Shaun2222 | basically i want to change the phones CALLERID(num) to the agents number when a agent logs into that phone |
| 07:32.27 | dlynes_home | Shaun2222: you mean for realtime? |
| 07:32.28 | cekc | the problem now is i can dial the extension and it rings, if it dials me it gets a busy signal, and when i call it and it's picked up there is no audio |
| 07:32.58 | dlynes_home | cekc: try adding a qualify=300 to the sip entry for the phones in quesiton |
| 07:33.18 | Shaun2222 | that way when a agent calls somebody the extention/number shows up as the agents ID and not the phone extension... that way if the agent moves and somebody trys to call the person back to follows them |
| 07:33.39 | dlynes_home | Shaun2222: i'll repeat again....do you mean for realtime? |
| 07:33.46 | Shaun2222 | ya i guess |
| 07:33.54 | dlynes_home | Shaun2222: yeah...then i don't have a clue |
| 07:34.10 | dlynes_home | Shaun2222: if it's not realtime, in sip.conf, there's a setting called 'callerid=' |
| 07:34.21 | Shaun2222 | when i dial from one phone to the next it gets set some how to the phones physical extention that it was setup with |
| 07:34.24 | dlynes_home | Shaun2222: that callerid field will allow you to adjust it |
| 07:34.29 | Shaun2222 | so i imagine it's something that can change realtime |
| 07:34.37 | dlynes_home | Shaun2222: more than likely |
| 07:34.45 | dlynes_home | Shaun2222: i just don't know what or where it would be |
| 07:35.52 | dlynes_home | anyways...i'm gonna head out before i fall asleep in my chair |
| 07:37.20 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 07:40.13 | Shaun2222 | doesnt look like it's set in the db... |
| 07:40.37 | *** join/#asterisk littlejohn (n=little@host31-85.pool877.interbusiness.it) |
| 07:45.00 | *** part/#asterisk littlejohn (n=little@host31-85.pool877.interbusiness.it) |
| 07:53.48 | cekc | my asterisk log is showing the internal nat IP of the remove sip device |
| 07:54.04 | alucard064 | no one knows about asterisk-java |
| 07:54.07 | alucard064 | ? |
| 07:55.16 | mitcheloc | why would you use that? java = =( |
| 07:55.58 | *** join/#asterisk qdk (n=qdk@213.237.44.34) |
| 08:13.20 | Shaun2222 | how can i delete a database entry in the extensions.conf |
| 08:13.44 | Shaun2222 | i tryed SET(DB(family/key)=) but that doesnt work |
| 08:15.01 | *** join/#asterisk darkskiez (n=mbryars@194.247.78.146) |
| 08:21.53 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 08:23.43 | *** join/#asterisk L|NUX (n=linux@202.5.145.56) |
| 08:25.11 | L|NUX | hello every one |
| 08:25.19 | *** join/#asterisk watchy (n=watchy@h236.176.255.206.cable.cmdn.cablelynx.com) |
| 08:25.26 | watchy | anyone with fxotune knowledge around |
| 08:26.10 | L|NUX | can some one tell me how can i add record for enum in my dns |
| 08:26.45 | *** join/#asterisk Makenshi (n=chaz@gw-212-219-188-68.ne-worcs.ac.uk) |
| 08:28.59 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 08:29.25 | L|NUX | y0 RoyK |
| 08:29.43 | RoyK | ~nickometer L|NUX |
| 08:29.48 | L|NUX | :( |
| 08:31.05 | drray | ~dickometer RoyK |
| 08:31.14 | L|NUX | its his bot :P |
| 08:31.18 | drray | heh |
| 08:31.26 | L|NUX | ~nickometer drray |
| 08:31.35 | L|NUX | ~nickometer RoyK |
| 08:31.45 | *** join/#asterisk Eric-xx (n=jcc@cm83.epsilon192.maxonline.com.sg) |
| 08:31.47 | L|NUX | ~nickometer l|nux |
| 08:31.53 | L|NUX | hehe |
| 08:31.59 | L|NUX | can some one tell me how can i add record for enum in my dns |
| 08:32.02 | drray | I assume it's the pipe in your name |
| 08:32.10 | L|NUX | ya |
| 08:34.25 | L|NUX | any one answer my question :) |
| 08:35.05 | *** join/#asterisk MGSsancho (n=user@adsl-67-122-137-205.dsl.irvnca.pacbell.net) |
| 08:35.07 | cekc | are there any ports other than 5060 that I should be forwarding? |
| 08:35.33 | zoa | royko |
| 08:35.42 | zoa | hows the trunk going for you ? |
| 08:35.55 | zoa | or were you on 2.6.1 now ? |
| 08:35.59 | zoa | euh |
| 08:36.03 | zoa | im talking rubbish here |
| 08:38.20 | darkskiez | cekc: all in your rtp range - see rtp.conf |
| 08:40.13 | kapsel | uhm, very simpel question: im using PAP2 adapters to use my old existing dect phones for ip, however im having problems with servicecodes, ie. *90 (call forward on busy). i think that i might have to disable the servicecodes in the adapter itself, to let it pass them on to the asterisk, am i totally wrong here? |
| 08:44.15 | *** join/#asterisk Modcuts (n=bob@82.133.98.155) |
| 08:47.50 | *** join/#asterisk dyerf (i=dyer@cube.sut.ru) |
| 08:51.05 | mitcheloc | zoa: good morning, and good night ;) |
| 08:51.32 | zoa | hey ho |
| 08:54.02 | nextime | let's go |
| 08:55.28 | RoyK | zoa: hi |
| 08:58.10 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 08:58.39 | swytch | Q: when using the action "UserEvent" with the '^' character to brak lines (as documented), i just get the '^' character instead of CRLF in the output of the Manager API. what do i do wrong? is there another way to break output lines in the UserEvent body? |
| 09:00.04 | L|NUX | time> let's go |
| 09:00.04 | L|NUX | <R |
| 09:00.30 | L|NUX | RoyK : i have done dns thing succcessfully but now when i try to dial using my enum it will not dialing |
| 09:00.33 | L|NUX | any idea |
| 09:00.34 | L|NUX | :) |
| 09:01.28 | L|NUX | RoyK : ummm |
| 09:01.32 | L|NUX | RoyK : okay thanks |
| 09:01.36 | L|NUX | any one else |
| 09:04.09 | *** join/#asterisk abatista (n=Ariel@dsl-20-177.cofs.net) |
| 09:04.31 | *** join/#asterisk nfi|ermes (n=ermsewrk@217.220.121.62) |
| 09:08.52 | *** join/#asterisk razu_ (n=razu@tln-kontor.norby.ee) |
| 09:13.12 | *** join/#asterisk speedwagon (n=Ariel@dsl-20-177.cofs.net) |
| 09:24.14 | *** join/#asterisk Antoine67 (n=FreePBX6@212.103.11.106) |
| 09:24.17 | Antoine67 | hi there |
| 09:24.53 | Raph | I've a pb with my misdn configuration, can anyone help me ? |
| 09:26.42 | Raph | How can I know what mode ishould use ? nt or te ? |
| 09:29.10 | Raph | nobody ? |
| 09:29.13 | zoa | i know |
| 09:29.16 | zoa | where do you connect to |
| 09:29.17 | zoa | ? |
| 09:29.30 | zoa | if you connect to the carrier, it will be TE |
| 09:29.34 | Raph | my asterisk is connect to a pabx |
| 09:29.36 | zoa | if you connect to phones its NT |
| 09:29.39 | zoa | then you can choose |
| 09:29.42 | zoa | one needs to be MT |
| 09:29.46 | zoa | NT and the other one TE |
| 09:30.16 | Raph | ok, thanks |
| 09:30.27 | Eric-xx | does anyone know what this means : Got SIP response 481 "Subcription Does Not Exist" back from 2 |
| 09:39.27 | RoyK | Eric-xx: rtfrfc :) |
| 09:42.41 | *** join/#asterisk psk (n=psk@golia.caltanet.it) |
| 09:43.55 | *** join/#asterisk emrah (n=fc@adslgva0491.worldcom.ch) |
| 09:44.15 | emrah | Hello !! |
| 09:45.17 | emrah | I'd like to put my voicemail files on a shared nas but I'm not able to do it because of the ast_lock_path function. Do you know if it is possible to disable that without modifying the source code? Or do you know a method to write into the shared smbfs folder? |
| 09:46.15 | Eric-xx | k :) |
| 09:48.49 | *** join/#asterisk Lino` (n=Lino@i577BD072.versanet.de) |
| 09:48.51 | emrah | Eric-xx: any idea? |
| 09:50.01 | Eric-xx | nope sorry |
| 09:59.01 | hypnox | emrah it's quite a simple tweak to take the call to lock_path out of the code |
| 10:04.43 | *** join/#asterisk stoffell (n=stoffell@fw.catsanddogs.com) |
| 10:07.05 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 10:15.56 | *** join/#asterisk cma (n=cma@garfield.amster.com) |
| 10:27.33 | *** join/#asterisk Sonderblade (n=muh@host-213.131.147.169.addr.tdcsong.se) |
| 10:30.27 | *** part/#asterisk halorgium (n=tim@202.50.176.27) |
| 10:31.37 | *** join/#asterisk r0d3nt (n=RatMan@foster.stonedcoder.org) |
| 10:33.48 | *** join/#asterisk michael-i (n=michael@141.41.38.58) |
| 10:34.09 | *** join/#asterisk Nobbie (n=no@fwb003.fw.is.co.za) |
| 10:34.12 | Nobbie | heya =) |
| 10:35.29 | Nobbie | When dialing voicemail and using G729 codec, the VM doens't detect the button presses (tones) eg: 2 to change folder. but it works when using ulaw/alaw. why's that and how can i fix it ? |
| 10:36.30 | Nobbie | ahh, is it the pass-thru issue ? |
| 10:36.33 | Nobbie | and license ? |
| 10:38.34 | X-Rob | Probably becase you're using dtmf inband |
| 10:38.43 | X-Rob | make sure dtmfmode is rfc2883 |
| 10:38.46 | *** join/#asterisk bjohnson (n=bjohnson@i216-58-62-114.cybersurf.com) |
| 10:38.59 | Nobbie | ta |
| 10:40.41 | Nobbie | Is it possible to specify G711 should be used for calls on the same LAN, and G729 for calls over a WAN ? all i see it the allow= option. how does asterisk acutally decide whch codec to use ? |
| 10:44.48 | X-Rob | mmmm |
| 10:45.03 | X-Rob | it's difficult |
| 10:46.25 | Nobbie | where can i read abou tit ? |
| 10:46.32 | Nobbie | about it even =) |
| 10:47.46 | RoyK | zoa: hello, sir |
| 10:48.07 | *** part/#asterisk queuetue (n=scott@toronto-HSE-ppp4122670.sympatico.ca) |
| 10:52.47 | Nobbie | ahh, http://www.voip-info.org/wiki/index.php?page=Asterisk+G.729+pass-thru explains why G729 won't work with voicemail if it's in pass-thru mode |
| 10:52.57 | tzafrir | Nobbie, one thing is to use different peeers for LAN and for WAN, if this is applicable |
| 10:53.40 | *** join/#asterisk Gunnar (n=gunnar@bkkb-gw.bitcon.no) |
| 10:53.48 | tzafrir | or to have the remote caller in the WAN insist on g729 and not g711 |
| 10:55.46 | Nobbie | ahh ok, the peers options can work. but that will need a different dialplan, which will require different phone numbers on the LAN and WAN ? |
| 10:58.04 | Pj_ | Is there a way to tell an E1 from a T1 from the asterisk server ? (like an alarm that should be set or smthg like that) |
| 10:58.40 | Pj_ | I'm in France and they delivered a "T2" but I've always had E1 so I'm not quite sure how I should set the signalling & co |
| 10:59.00 | RoyK | Pj_: i seriously doubt you get T1 in france |
| 10:59.06 | Pj_ | Me too |
| 10:59.27 | Pj_ | But they said it's 24 channels |
| 10:59.31 | Pj_ | so if I could just make sure |
| 11:00.58 | Pj_ | so I tried playing around with the zaptel.conf but, well, I'm not so sure :D |
| 11:01.38 | *** join/#asterisk jahani (n=k@adsl196-69-227-217-196.adsl196-16.iam.net.ma) |
| 11:02.19 | jahani | what is the stable version is from this link http://svn.digium.com/svn/asterisk/trunk asterisk or this one http://svn.digium.com/svn/asterisk/branches/1.2 ? |
| 11:03.45 | nextime | jahani : the second one |
| 11:04.07 | jahani | ok thank you |
| 11:08.36 | *** join/#asterisk |dennis| (i=dennis@200.32.215.84) |
| 11:09.17 | *** join/#asterisk Dr-Linux (n=Nothing@202.125.141.6) |
| 11:13.23 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 11:15.22 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 11:18.29 | *** join/#asterisk Eladamri (n=Ablaidan@cs181074111.pp.htv.fi) |
| 11:19.03 | *** join/#asterisk apardo (n=apardo@87.217.145.95) |
| 11:24.06 | Nobbie | s/quit . |
| 11:33.48 | *** join/#asterisk b00mer (i=fwuser@blackhole.c5i.com) |
| 11:36.15 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 11:52.42 | *** join/#asterisk cma (n=cma@garfield.amster.com) |
| 11:54.03 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 11:55.20 | *** join/#asterisk MatsK (n=mats@141.221.181.62.in-addr.dgcsystems.net) |
| 11:57.59 | *** join/#asterisk SHad|Work (n=kvirc@84-255-228-2.dynamic.dsl.t-2.net) |
| 11:58.05 | SHad|Work | hi |
| 11:58.30 | SHad|Work | I've got problem compiling chan_h323 anyone here with gcc 4.x that has succeded? |
| 12:00.02 | emrah | Anyone knows how to lock a directory in a mounted smbfs? (To store vm) |
| 12:00.20 | emrah | How to enable th locking option * |
| 12:00.22 | *** join/#asterisk MikeJ[Laptop] (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 12:02.04 | tzafrir | emrah, loop-mount another file-system on it? |
| 12:02.56 | tzafrir | I know: performance hits and such. But at least it should provide locking |
| 12:03.45 | emrah | tzafrir: I don't understand you |
| 12:04.00 | emrah | What do you mean by loop-mount? |
| 12:04.58 | tzafrir | generate a huge file (dd if=/dev/zero of=that/file bs=1024 count=10M) |
| 12:05.19 | tzafrir | then generate a filesystem in it: mkfs.ext3 that/file |
| 12:05.49 | tzafrir | now: mount -o loop that/file /mnt/point |
| 12:06.49 | emrah | tzafrir: May I talk to you in a private window? I'll not disturb you much, I just don't understand what you exactly mean... |
| 12:06.50 | tzafrir | Linux will consider that file just as if it were a real partition / block device |
| 12:07.18 | tzafrir | ask here or in a general Linux channel (or in one for your distro) |
| 12:07.29 | *** join/#asterisk aze (n=aze@ACayenne-101-1-3-206.w81-248.abo.wanadoo.fr) |
| 12:08.10 | tzafrir | basically it is an ugly workaround in this case. It does not really solve your problem. It just changes the filesystem |
| 12:08.16 | emrah | tzafrir: I'd like to share my vm files on a nas... The nas allows both nfs and smbfs, but they do not provide the locking feature... So I'm not able to record vm properly |
| 12:09.03 | tzafrir | Won't this locking work with NFS? |
| 12:09.36 | emrah | Let me try, thanks a lot |
| 12:14.16 | *** join/#asterisk Ansonmus (n=ahaeser@a213-84-26-148.adsl.xs4all.nl) |
| 12:16.27 | *** join/#asterisk [TK]D-Fender (n=joe@toronto-HSE-ppp4122655.sympatico.ca) |
| 12:16.57 | *** join/#asterisk RoyK (n=roy@213.160.242.91) |
| 12:18.29 | *** part/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:18.33 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 12:21.45 | florz | tzafrir: As long as there aren't multiple clients accessing the same files, why shouldn't that actually solve the problem? |
| 12:22.17 | SHad|Work | florz: are you the florz-bristuff patch author? :) |
| 12:22.20 | tzafrir | it does. It's just ugly. And there are probably some performance hits |
| 12:22.25 | *** part/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 12:22.27 | florz | SHad|Work: yep |
| 12:22.42 | *** join/#asterisk jpeeler (n=jpeeler4@host86-129-192-76.range86-129.btcentralplus.com) |
| 12:22.47 | florz | tzafrir: Well, probably, yeah :-) |
| 12:22.53 | SHad|Work | florz: so I could bother you a bit about it :) |
| 12:23.35 | SHad|Work | florz: I tried patching bristuff 0.3.0-pre1o |
| 12:23.40 | *** join/#asterisk rogger (n=rogger@209.104.162.252) |
| 12:23.58 | Ansonmus | florz: Tank you for the patch! Are there plants to update it so it will work with laterst builds? |
| 12:24.04 | SHad|Work | florz: but afterwards my HFC-PCI cards don't react to anything, I can't dial, or recieve calls |
| 12:24.28 | SHad|Work | is there a possibility I did something insanely stupid? |
| 12:25.05 | florz | Ansonmus: Erm, there is anything the patch doesn't apply on yet? |
| 12:25.24 | SHad|Work | no it applied to all the 3 files without a problem |
| 12:25.50 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 12:25.51 | florz | SHad|Work: Erm, it worked (somehow) with the unpatched zaphfc? |
| 12:26.04 | SHad|Work | yes |
| 12:26.18 | SHad|Work | if I just compile the bristuff patched asterisk it works |
| 12:26.19 | florz | SHad|Work: So, no config changes or anything? |
| 12:26.38 | SHad|Work | but I get my logs flooded about messasges about cpu throttling |
| 12:26.43 | SHad|Work | nope nothing |
| 12:26.53 | SHad|Work | just patched the source and recompiled |
| 12:27.45 | florz | SHad|Work: What does the system log say when loading the driver? |
| 12:27.49 | *** join/#asterisk jono (n=jono@mail.openadvantage.org) |
| 12:28.08 | jono | hi |
| 12:28.09 | SHad|Work | hm |
| 12:28.16 | florz | SHad|Work: I mean: does it find all the cards and are all the settings it's telling you correct? |
| 12:28.29 | jono | is there a list of all the numbers you dial to access different bits of asterisk, such as the number to dial your voicemail |
| 12:28.55 | SHad|Work | florz: yes, everything looks fine |
| 12:29.20 | florz | SHad|Work: So, what's the exact symptoms then of it "not working"? |
| 12:29.28 | jono | anyone know ? |
| 12:29.42 | florz | jono: Are you looking for your extensions.conf? |
| 12:30.04 | SHad|Work | florz: well if a call comes in to the bri line, nothing happens |
| 12:30.24 | SHad|Work | the ISDN phone then timeouts after a while with congestion |
| 12:30.26 | jono | I was just looking for a list on a wiki or webpage of all the numbers for dialing different bits of * |
| 12:32.31 | florz | SHad|Work: if you load the driver with debug=1, does it output some more messages about received/sent d channel messages? |
| 12:32.53 | SHad|Work | the zaphfc module? |
| 12:32.59 | *** part/#asterisk jono (n=jono@mail.openadvantage.org) |
| 12:33.07 | florz | SHad|Work: (which look like "RX [...]" and "TX [...]", respectively) |
| 12:33.10 | florz | SHad|Work: yep |
| 12:33.19 | SHad|Work | just a moment, I'll try |
| 12:35.11 | RoyK | anyone using sangoma here? gcc: installation problem, cannot exec `cc1plus': No such file or directory |
| 12:36.08 | RoyK | ~nickometer [TK]D-Fender |
| 12:36.35 | [TK]D-Fender | RoyK : But I have more Karma that you can ever hope to attain :) |
| 12:36.38 | SHad|Work | RoyK: suse? sounds like a dist problem |
| 12:37.03 | *** join/#asterisk wunderkin (i=kev@ip68-226-113-228.ph.ph.cox.net) |
| 12:37.04 | RoyK | SHad|Work: debian |
| 12:37.11 | RoyK | SHad|Work: amd64 |
| 12:37.13 | RoyK | sarge |
| 12:37.30 | *** join/#asterisk Kis (i=vlad@p5080D8AB.dip.t-dialin.net) |
| 12:37.38 | *** join/#asterisk saftsack (n=saftsack@p54A7FB97.dip.t-dialin.net) |
| 12:37.42 | RoyK | i've been using sarge for several other boxes |
| 12:37.48 | SHad|Work | you've got gcc-c++ installed? |
| 12:38.16 | RoyK | yes |
| 12:38.21 | zoa | who needs karma anyway |
| 12:39.02 | [TK]D-Fender | zoa : My karma ran over your dogma :D |
| 12:39.03 | *** join/#asterisk saftsack (n=saftsack@p54A7FB97.dip.t-dialin.net) |
| 12:39.23 | Ansonmus | Where can I found the latest Florz patch? |
| 12:39.26 | *** join/#asterisk jono (n=jono@mail.openadvantage.org) |
| 12:39.31 | jono | what do I dial to acces smy voicemail |
| 12:39.42 | zoa | just checked, a jitter buffer does not give karma :) |
| 12:39.44 | zoa | im +5 |
| 12:39.45 | zoa | whii |
| 12:39.46 | zoa | :) |
| 12:39.59 | zoa | not that i really care |
| 12:40.05 | zoa | never got the whole deal |
| 12:40.08 | zoa | royk |
| 12:40.12 | zoa | how is this jb thing going now ? |
| 12:40.31 | RoyK | very, very good, zoa |
| 12:40.54 | RoyK | ~karma zoa |
| 12:40.54 | jbot | zoa has neutral karma |
| 12:40.56 | zoa | so it was just the damn debugging ? :/ |
| 12:41.00 | RoyK | ~karma royk |
| 12:41.00 | jbot | royk has karma of 1 |
| 12:41.02 | zoa | haha |
| 12:41.10 | zoa | roy is better than me |
| 12:41.12 | zoa | :) |
| 12:41.19 | zoa | speak to me master! |
| 12:41.31 | zoa | and how is that memory leak ? |
| 12:41.34 | zoa | do you still see that ? |
| 12:41.35 | RoyK | gone |
| 12:41.39 | RoyK | prolly a sip/sip thing |
| 12:41.45 | zoa | strange |
| 12:41.50 | zoa | oh well, good news :) |
| 12:41.57 | RoyK | :) |
| 12:42.06 | zoa | didnt get any news from anybody else about the jb |
| 12:42.07 | florz | Ansonmus: http://zaphfc.florz.dyndns.org/ |
| 12:42.18 | florz | Ansonmus: Or google, for that matter =:-) |
| 12:42.37 | Ansonmus | yeah I googled but the url is dead for me |
| 12:42.49 | Ansonmus | maybe our firewall |
| 12:42.54 | florz | gnah |
| 12:43.01 | florz | nah, probably rather my dyndns client |
| 12:43.25 | SHad|Work | florz: I've tried now, and absolutely nothing happens :) |
| 12:43.35 | florz | I think I should finally fix that =:-) |
| 12:43.43 | florz | Ansonmus: OK, try again, should work now |
| 12:44.28 | florz | SHad|Work: Not even when you try to make a call? |
| 12:44.30 | Ansonmus | ahh now it works |
| 12:44.48 | SHad|Work | then I get: Channel 0/1, span 2 got hangup, cause 42 |
| 12:45.06 | florz | SHad|Work: I meant in the syslog? |
| 12:45.15 | florz | s/\?/!?/ |
| 12:45.22 | SHad|Work | no |
| 12:45.26 | SHad|Work | nothing in the syslog |
| 12:46.23 | florz | SHad|Work: weird, indeed |
| 12:46.27 | Pj_ | Does VN6 rings a bell to someone ? |
| 12:46.56 | florz | SHad|Work: what about the interrupt counters of the respective interrupt(s) in /proc/interrupts? |
| 12:47.14 | florz | SHad|Work: Is one of them increasing at a rate of ~ 8000/second? |
| 12:47.53 | SHad|Work | doesn not seem to |
| 12:48.04 | SHad|Work | if that's the 2nd column |
| 12:48.24 | florz | SHad|Work: yep |
| 12:48.40 | SHad|Work | nope they are all about 250 |
| 12:48.46 | SHad|Work | and slowly going up |
| 12:48.56 | florz | SHad|Work: hmmm[tm] |
| 12:49.12 | SHad|Work | it seems one of the cards has a shared irq with an eth card |
| 12:49.14 | Ansonmus | For your information (maybe you get questions). In Holland the telco KPN has set the ISDN lines default to non line-sync. Asterisk + BRIstuff wouldn't work OK with that (problems with hangup and other connection problems). You can call KPN and ask if the set the option line-sync on. THen it is OK |
| 12:49.15 | florz | SHad|Work: so, what does you setup look like? like, how many cards and in which modes? |
| 12:49.25 | *** join/#asterisk ghenry (n=ghenry@mail.suretecsystems.com) |
| 12:49.42 | SHad|Work | 4 cards |
| 12:49.48 | SHad|Work | connected to bri lines |
| 12:49.54 | SHad|Work | TE mode? |
| 12:50.15 | SHad|Work | (I'm new to telephony speak :)) |
| 12:50.30 | florz | if it's as in "BRI lines provided by the telco", then they should be in TE mode, yeah |
| 12:50.44 | SHad|Work | yup, that that's that |
| 12:51.11 | SHad|Work | the syslog also says something about master mode |
| 12:51.18 | SHad|Work | but that's beyond me :) |
| 12:51.55 | SHad|Work | Could this be a problem: kernel: zaphfc: no version for "zt_receive" found: kernel tainted. |
| 12:54.34 | florz | SHad|Work: Well, what that is about is described on the patch's website, but there is no other choice than master mode for TE-mode cards anyway ... |
| 12:54.51 | *** join/#asterisk swytch (n=ezcall@LNeuilly-152-22-86-193.w193-251.abo.wanadoo.fr) |
| 12:55.03 | SHad|Work | couldn't get to the site for the past few days :) |
| 12:55.35 | SHad|Work | I've been using this setup for a while now and I got annoyed by the growing logs |
| 12:55.38 | Eladamri | does anyone happen to have any experience with varphonex.com? |
| 12:55.52 | SHad|Work | so I read somwhere your patch my fix that and improve overall performance |
| 12:56.21 | swytch | is it possible to use SIP via asterisk, and route the RTP besides the asterisk machine, so that the RTP is not going thru asterisk, only all the SIP messages for these sessions ? |
| 12:56.26 | Eladamri | i can't get my proxy server tries to work with varphonex... |
| 12:56.51 | florz | SHad|Work: Well, probably that stupid race condition in the script that calls the dyndns-client (packet filter doesn't allow outgoing connections yet when it is started) is something I really should fix =:-) |
| 12:57.28 | *** join/#asterisk groogs (n=greg@d38-54-164.commercial1.cgocable.net) |
| 12:57.43 | SHad|Work | but there's just no free time :) |
| 12:58.00 | *** join/#asterisk aze (n=aze@ACayenne-101-1-3-206.w81-248.abo.wanadoo.fr) |
| 12:58.30 | florz | SHad|Work: Well, as far as your setup is concerned, try loading the driver with a different timer_card-parameter |
| 12:58.47 | florz | SHad|Work: Basically, anything from 1 to 3 |
| 12:59.02 | florz | SHad|Work: and see whether that changes something |
| 12:59.06 | SHad|Work | hm lemme try |
| 13:00.36 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-8-107.w81-248.abo.wanadoo.fr) |
| 13:00.39 | SHad|Work | by the way |
| 13:00.50 | SHad|Work | is there any way to unload the module? :) |
| 13:01.02 | swytch | does asterisk 'insist having rtp routed thru it when using SIP ? \-8 |
| 13:01.04 | SHad|Work | or is rebooting the only way? |
| 13:01.18 | SHad|Work | swytch: depends on your settings |
| 13:01.22 | florz | SHad|Work: well, rmmod should work fine!? |
| 13:01.22 | blitzrage | SHad|Work: unload <module>.so |
| 13:01.30 | [TK]D-Fender | swytch : "canreinvite=yes" |
| 13:01.38 | SHad|Work | florz: no matter what I do it say's it's in use |
| 13:01.55 | swytch | [TK]D-Fender: but then, would still the SIP messages go thre asterisk after the re-INVITE ?? |
| 13:02.11 | blitzrage | SHad|Work: you can't unload a module that is in use, or is a dependency of something else |
| 13:02.23 | blitzrage | swytch: yes -- signalling always goes through Asterisk |
| 13:02.30 | SHad|Work | yeah the weird thing is it says it's in use by [permanent] |
| 13:02.33 | [TK]D-Fender | swytch : Yes |
| 13:02.38 | *** join/#asterisk myiagy (n=myiagy@mail.voffice.com.br) |
| 13:02.41 | SHad|Work | which I guess would me it can't be unloaded :) |
| 13:02.43 | blitzrage | SHad|Work: pastebin the message and what you're doing |
| 13:02.53 | SHad|Work | pastebin? |
| 13:03.05 | [TK]D-Fender | ~pb |
| 13:03.10 | 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, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 13:04.02 | florz | SHad|Work: hmm, you've got module unloading support in your kernel? |
| 13:04.31 | swytch | blitzrage: so with reivite my viop gateway can talk sip via asterisk but having the rtp besides it . (i got it right?) |
| 13:04.33 | SHad|Work | blitzrage: http://pastebin.com/767253 |
| 13:04.55 | SHad|Work | florz: it's a stock suse 10.1 kernel I guess it should be |
| 13:05.27 | [TK]D-Fender | swytch : Correct |
| 13:05.45 | swytch | thanks folks. |
| 13:07.52 | myiagy | if i call somewhere, and the telco gives me a message like : "the number you dialed is not available" |
| 13:07.57 | myiagy | do i get charged for that? |
| 13:09.48 | blitzrage | SHad|Work: thought you were talking about a dialplan module -- you can't unload it because something is using it |
| 13:10.05 | SHad|Work | no, sorry a kernel module :) |
| 13:10.40 | blitzrage | SHad|Work: what distro? if Redhat based, you can do 'make config' in the zaptel source directory and use 'service zaptel stop' to unload everything, although the fact its zaphfc, which I don't recognize, makes me think its a third party module |
| 13:12.07 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 13:13.37 | b00mer | anybody having issues with Teliax recently? |
| 13:14.00 | *** join/#asterisk brad_mssw (n=brad@shop.monetra.com) |
| 13:16.59 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 13:17.01 | *** join/#asterisk mvdk (i=mvdk@60-240-15-230-nsw-pppoe.tpgi.com.au) |
| 13:18.16 | mvdk | Using Debian testing, I get messages (relating to a connection with VoIP provider Engin) "chan_sip.c: Don't know how to indicate condition 9" |
| 13:18.47 | mvdk | I wouldn't ordinarily worry, but the user has been complaining of regular dropouts at times coinciding with the indication of the condition |
| 13:18.52 | mvdk | Any ideas? |
| 13:21.12 | *** join/#asterisk jsaunders (i=jsaunder@S01060060971c5817.va.shawcable.net) |
| 13:23.16 | jono | how do I call a queue from the phone? |
| 13:23.30 | *** join/#asterisk Qwell[] (i=north@unaffiliated/qwell) |
| 13:23.46 | Pj_ | I keep getting primary D-channel on span 1 {up|down}, and not alternatively (like several "up" before a "down")... wtf could that mean ? I'd understand if line went up then down but "that"... |
| 13:25.52 | *** join/#asterisk RoyK (n=roy@80.239.107.70) |
| 13:27.34 | *** join/#asterisk MACscr (n=MACscr@66.73.154.70) |
| 13:27.38 | MACscr | hello everyone |
| 13:27.57 | RoyK | <PROTECTED> |
| 13:28.11 | *** join/#asterisk Meaty (n=cp_simbu@office.abi.ca) |
| 13:29.45 | MACscr | If anyone has a couple extra seconds, I would love to get some feedback on my post here about an asterisk solution: |
| 13:29.48 | MACscr | http://www.pbxinfo.com/forums/showthread.php?t=18186 |
| 13:29.57 | MACscr | well, i guess a couple extra minutes |
| 13:30.07 | florz | SHad|Work: Well, until I've found out what this [permanent] marking is about, how about rebooting, then? |
| 13:30.11 | [TK]D-Fender | Pj_ : Some telco's cause a regular reset of the PRI when idle... As for the downs it may be just due to cycle time. I'd say call them up and ask what they're doing... |
| 13:30.13 | MACscr | reply here or there, i dont care, either way i appreciate it |
| 13:30.29 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 13:30.54 | Pj_ | [TK]D-Fender: ok thanx |
| 13:30.58 | mvdk | MACscr: just use agents |
| 13:31.10 | mvdk | The documentation is everywhere, it should be a no-brainer |
| 13:31.18 | mvdk | Just google for agents.conf |
| 13:31.32 | MACscr | thanks mvdk. Will definately check that out |
| 13:31.59 | mvdk | And auto-attendant: Well, everyone has a tutorial for that, too |
| 13:32.49 | mvdk | Unfortunate that I've got no-one with any clue about SIP floating about, though :( |
| 13:32.53 | SHad|Work | florz: tried the timer_card doesen't help |
| 13:33.11 | SHad|Work | florz: could it be an issue with the cpu being an AMD64? |
| 13:33.12 | florz | SHad|Work: So, still no increasing counter in /proc/interrupts? |
| 13:33.19 | SHad|Work | nope |
| 13:34.20 | florz | SHad|Work: Well, AFAIK it _should_ not, on the other hand I don't know of anyone yet for sure that it is working, so ... |
| 13:34.25 | MACscr | mvdk: what do you mean by "SIP floating about" |
| 13:34.46 | mvdk | Well, do you know what "condition 9" on a line is? |
| 13:35.11 | mvdk | It tells me that it "doesn't know how to indicate condition 9" |
| 13:35.19 | florz | Or is there possibly anyone around who is using zaphfc on an AMD64? =:-) |
| 13:35.26 | mvdk | I've googled for this, and it's worthless |
| 13:35.27 | *** join/#asterisk anthm (n=anthmct@CPE-69-76-83-52.wi.res.rr.com) |
| 13:35.27 | *** mode/#asterisk [+o anthm] by ChanServ |
| 13:35.33 | florz | with my patch, that is ... |
| 13:35.54 | mvdk | It seems that no-one encounters this, and I am not a telephone guy |
| 13:37.39 | SHad|Work | I guess I'll try to get an i386 machine and test on that too |
| 13:38.22 | motu | how do i supress the "thankyou" message from being played after having recorded a voicemail message and pressed pound key? |
| 13:38.45 | *** join/#asterisk key2 (n=ashdown@sd-420.dedibox.fr) |
| 13:38.59 | *** join/#asterisk klasstek (n=nunyobiz@ip67-154-143-190.z143-154-67.customer.algx.net) |
| 13:39.02 | SHad|Work | florz: oh and I get a warning when compiling |
| 13:39.13 | SHad|Work | florz: which I don't get without the patch |
| 13:39.16 | Splat | motu: you hang up instead of pressing the pount key... ? heh |
| 13:39.25 | florz | SHad|Work: that is? |
| 13:39.55 | SHad|Work | florz: http://pastebin.com/767317 |
| 13:40.07 | motu | heh, i dont want callers to hear that woman's voice if they happen to press pound key |
| 13:40.20 | *** join/#asterisk tamp4x (n=Lab@www.vonworldwide.com) |
| 13:40.27 | motu | any way to do it? |
| 13:40.42 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 13:40.54 | X-Rob | motu, change the 'thankyou.gsm' file |
| 13:40.55 | X-Rob | ? |
| 13:40.56 | Damin | motu: Well, if there isn't an option, then you can simply comment that out in app_voicemail. |
| 13:41.11 | florz | SHad|Work: Which version is that? |
| 13:41.13 | Damin | motu: Or just put a second of silence in the thanyou.gsm file. |
| 13:41.32 | Splat | no idea.. I'm more concerned about making sfftobmp to compile or find a centos package for it.. heh |
| 13:41.34 | SHad|Work | florz: the most recent one 0.3.0-pre1p |
| 13:41.40 | motu | can i delete thankyou.gsm? |
| 13:41.48 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:42.23 | motu | where are the sound files located? |
| 13:42.26 | SHad|Work | florz: I diffed all the files in the zaphfc dir with pre1o and they were the same |
| 13:43.12 | florz | SHad|Work: yep. But those warnings are just about some printk() parameter, so that's unlikely to be the problem ... |
| 13:44.14 | SHad|Work | hm no other weird things, that I can find |
| 13:45.34 | *** join/#asterisk robin_sz (n=robin@212.243.40.130) |
| 13:45.54 | *** join/#asterisk iulius (n=iulius@mail1.technologieshq.com) |
| 13:45.55 | robin_sz | remind me .. the pattern in extensions.conf that matches a digit other than 0? |
| 13:46.18 | *** join/#asterisk coppice (n=chatzill@44.199.17.210.dyn.pacific.net.hk) |
| 13:46.32 | robin_sz | N? |
| 13:47.01 | *** join/#asterisk loopt (n=pt@gw1.sanyo.hu) |
| 13:47.22 | robin_sz | ahh, Z |
| 13:47.44 | florz | SHad|Work: but you do see all four cards in /proc/interrupts? |
| 13:47.54 | SHad|Work | yes |
| 13:48.16 | SHad|Work | and if I use the unpatched module the interrupt number do go up at about 8khz |
| 13:48.49 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 13:49.32 | florz | SHad|Work: you did call ztcfg? =:-) |
| 13:49.39 | SHad|Work | yes |
| 13:51.39 | motu | there is no thankyou file, only thank-you-cooperation and thank-you-for-calling |
| 13:51.58 | *** join/#asterisk C4T3l (n=rcall01@216.54.143.2) |
| 13:52.21 | *** join/#asterisk TheCompWiz (n=TheCompW@wsip-68-109-200-102.mc.at.cox.net) |
| 13:52.30 | TheCompWiz | anyone in here good with the voicemail system? |
| 13:52.38 | TheCompWiz | I can't figure out what's wrong. |
| 13:52.47 | SHad|Work | florz: I'll be back to bother you later, thank you for the help :) |
| 13:53.05 | robin_sz | TheCompWiz, does it have a whiny american on it? |
| 13:53.32 | MACscr | wow, im taking a guess your french? |
| 13:53.34 | MACscr | =P |
| 13:53.40 | TheCompWiz | hehe... no. Every time I get to the point where it is supposed to leave a message... it says "The person at extension" & hangs up |
| 13:53.40 | florz | SHad|Work: Well, yeah, I've got no real clue yet what to try next anyway =:-) |
| 13:53.42 | *** join/#asterisk burizaa (n=freeee@cm56.omega110.maxonline.com.sg) |
| 13:53.56 | robin_sz | MACscr, s/your/you're/ |
| 13:54.06 | TheCompWiz | it never says the extension #... and never lets me leave a message. |
| 13:54.44 | MACscr | Thanks for the spelling correction, I appreciate it. |
| 13:54.49 | RoyK | robin_sz: he's prolly american |
| 13:55.16 | robin_sz | pas probleme mes amis, vous etes en pris |
| 13:55.44 | RoyK | ja, sant, det er bare tull med franskmenn, amerikanere og kanadere |
| 13:56.18 | *** join/#asterisk CodyC (n=cody@207.200.23.194) |
| 13:56.52 | TheCops | lol robin_sz |
| 13:56.56 | TheCops | vous etes en pris |
| 13:56.57 | TheCops | cest quoi sa |
| 13:57.06 | mvdk | TheCompWiz: create /var/spool/asterisk/voicemail, and give it the right permissions |
| 13:57.10 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 13:57.45 | MACscr | eh, whatever. Asterisk was create by an American company, so say what you will. |
| 13:57.48 | TheCompWiz | what permissions do I need to give it? |
| 13:57.53 | mvdk | Then go to the VoiceMailMain() application for the relevant mailbox, and you should be right |
| 13:58.36 | mvdk | TheCompWiz: if the user you run asterisk has 7, everythings OK |
| 13:58.54 | mvdk | I recommend you make it something like asterisk:root with permissions 700 |
| 13:59.02 | *** join/#asterisk Ash-OK (n=ashok@247.15.187.81.in-addr.arpa) |
| 13:59.03 | robin_sz | MACscr, s/create/created/ |
| 13:59.12 | TheCompWiz | mvdk... it was asterisk:asterisk 700 |
| 13:59.19 | mvdk | That works fine |
| 13:59.25 | TheCompWiz | still doing it. |
| 13:59.36 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 13:59.37 | robin_sz | running * as root would be just slightly crazy |
| 13:59.47 | mvdk | It would |
| 13:59.56 | TheCompWiz | I also tried 777 for everything... and it still is not working. |
| 14:00.09 | Vorondil | hi all, quick question, if you do an attended transfer, and the person you're transferring to is "do not disturb" (so it goes to voice mail), what are you supposed to do? is there any way to get back to the caller? |
| 14:00.14 | mvdk | Did you record a message for it through the voicemailmain application? |
| 14:00.36 | TheCompWiz | mvdk... how do I record a message... if I can't record a message? |
| 14:00.47 | mvdk | Through the voicemailmain application |
| 14:00.53 | mvdk | Not the voicemail application |
| 14:01.29 | mvdk | On the internal context, you might have an extension like ##, or something, that goes to VoiceMailMain() |
| 14:01.50 | TheCompWiz | how would I find that? |
| 14:01.57 | mvdk | In extensions.conf |
| 14:02.03 | mvdk | This is part of your dial plan |
| 14:02.24 | mvdk | How do you plan to retrieve messages? This is one way to do so.... |
| 14:02.52 | mvdk | God, have you set up an internal context? |
| 14:04.01 | mvdk | TheCompWiz: Still there? |
| 14:04.06 | TheCompWiz | yeah... looking. |
| 14:04.10 | Vorondil | nm, found it. (*0) |
| 14:04.31 | mvdk | OK, have you found where you direct calls from "internal" phones to? |
| 14:05.32 | TheCompWiz | yeah. |
| 14:06.02 | mvdk | Well, have you put a new extension in there? |
| 14:06.16 | TheCompWiz | yes. |
| 14:06.29 | mvdk | Something like a line that says exten => ##,1,VoiceMailMain()? |
| 14:06.31 | *** join/#asterisk ToTo (n=ToTo@81.174.33.2) |
| 14:06.40 | *** join/#asterisk apardo (n=apardo@63.Red-88-0-68.dynamicIP.rima-tde.net) |
| 14:07.12 | mvdk | You may make it whatever you like, I just find ## nice and easy for people to remember :) |
| 14:07.48 | *** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net) |
| 14:07.54 | pdavid | morning all |
| 14:08.07 | mvdk | OK, have you proceeded to dial that extension, TheCompWiz? |
| 14:08.11 | *** join/#asterisk malcolmd (n=malcolmd@pdpc/sponsor/digium/malcolmd) |
| 14:08.14 | mvdk | After reloading asterisk |
| 14:09.05 | pdavid | i am getting an error that the: Function CURL is not registered |
| 14:09.19 | pdavid | is there some way to register curl to * |
| 14:09.20 | mvdk | TheCompWiz: Success? |
| 14:09.41 | Nivex | pdavid: from experience, it sounds like you didn't have the curl headers present when you built asterisk |
| 14:09.54 | pdavid | nivex: thanks, so i guess a re-compile is in order |
| 14:10.17 | asterboy | ok, so who has zasterisk working yet so I can get my fix? |
| 14:10.34 | *** join/#asterisk viler (i=1000@200.114.70.228) |
| 14:10.59 | mvdk | TheCompWiz: Any luck? |
| 14:11.04 | TheCompWiz | hold on... |
| 14:11.13 | mvdk | Oh cool.... |
| 14:11.39 | TheCompWiz | ... um replace "hold on...." with "wait a few moments please" |
| 14:11.59 | watchy | i wish someone would hug me |
| 14:12.24 | mvdk | Watchy: You might try the hugs channel, they're far more likely to be affectionate :) |
| 14:12.25 | watchy | today is a bad day so far |
| 14:12.33 | watchy | 9 billion people have called me asking me stuff |
| 14:12.57 | TheCompWiz | really? 9 billion? shoulda asked a dollar from each of 'em... you'd be a very rich man ;) |
| 14:13.04 | TheCompWiz | (or woman... as the case may be) |
| 14:13.11 | b00mer | anyone from asterlink here? |
| 14:13.28 | watchy | yea |
| 14:14.31 | mvdk | TheCompWiz: You've put in the extension, added the voicemailmain application, reloaded asterisk, rung that extension, left a busy message, and an unavailable message, and you're done |
| 14:14.41 | mvdk | Which steps haven't you completed? |
| 14:14.45 | watchy | this guy who controls the domain of one of my cleints changed all his a records on his domain |
| 14:14.57 | TheCompWiz | mvdk... my phone keeps saying "404 not found" now... |
| 14:14.58 | watchy | so now my clients email is broke and they are a big time .gov contractor |
| 14:15.06 | *** join/#asterisk Tili (n=Tili@cm109.gamma248.maxonline.com.sg) |
| 14:15.15 | watchy | so they are far from happy atleast they arent pissed at me |
| 14:15.25 | mvdk | Are you talking about a SIP phone? |
| 14:15.30 | TheCompWiz | yeah. |
| 14:15.44 | watchy | so i told this guy ad 4 a records to the dns server |
| 14:15.55 | watchy | he changed the mail a record to the ips i gave him haha |
| 14:15.57 | mvdk | What extension did you add? |
| 14:15.57 | watchy | wtf? |
| 14:16.18 | TheCompWiz | I havn't added any yet. |
| 14:16.23 | TheCompWiz | these were the existing ones. |
| 14:16.40 | watchy | i hate idiots |
| 14:16.42 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 14:16.58 | mvdk | Yeah, but you added a line that went something like "exten => ##,1,Voicemailmain(), right? |
| 14:17.36 | mvdk | Haven't you? |
| 14:17.50 | TheCompWiz | just wait a few seconds... it's in a different office. |
| 14:18.04 | mvdk | God, hasn't everyone heard of SSH by now? |
| 14:18.11 | robin_sz | apparently not |
| 14:18.21 | watchy | ssh is made for gods of the internet |
| 14:18.30 | TheCompWiz | ... I was using *98 instead of ## ... |
| 14:18.31 | file | lalala good morning everyone |
| 14:18.36 | TheCompWiz | but yes. |
| 14:18.51 | mvdk | I'm perfectly happy working on a server halfway across the city, you know, or even halfway across the country :) |
| 14:19.09 | mvdk | OK, so you dialed *98 after reloading asterisk |
| 14:19.15 | TheCompWiz | sorry.... it's not available unless you can get through 2 vpn tunnels ;) |
| 14:19.22 | TheCompWiz | yeah... and now *98 says 404 |
| 14:19.22 | robin_sz | watchy, if they havent heard of, and use, ssh, they are probably not going to be capable of sorting out an * dialplan anyway. |
| 14:19.31 | mvdk | Did you reload asterisk? |
| 14:19.33 | TheCompWiz | yes |
| 14:20.00 | mvdk | Have you checked out the server, and seen what context it happens to be in at that time, etc? |
| 14:20.40 | *** join/#asterisk LakeSolon (n=blake@12-227-169-99.client.mchsi.com) |
| 14:20.43 | robin_sz | and you know you can test from the console by dial ##@context |
| 14:21.12 | TheCompWiz | ok... got rid of the 404... but now it won't accept mailbox/password |
| 14:21.30 | mvdk | Ah, have you checked voicemail.conf? |
| 14:22.13 | TheCompWiz | I can't make heads/tails of it. |
| 14:22.25 | mvdk | Have you looked at the documentation? |
| 14:22.31 | mvdk | I didn't find it hard at all.... |
| 14:22.35 | TheCompWiz | quite a bit... but I can't follow it very well. |
| 14:22.46 | mvdk | OK, one moment |
| 14:23.03 | [TK]D-Fender | TheCompWiz: Pastebin your dialplan and voicemail.conf |
| 14:23.40 | *** join/#asterisk fri (n=fri@port84.ds1-sdb.adsl.cybercity.dk) |
| 14:23.46 | [TK]D-Fender | ~pb |
| 14:23.47 | jbot | i heard pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 14:25.16 | mvdk | Well, a typical entry would be [Name of voicemail plans] \n <vmailbox> => <vmailpass>, <RealName of user>, <email address of user> |
| 14:26.07 | mvdk | That means: [default] <newline> 185 => 567,CoolGuy,CoolGuy@example.com |
| 14:26.35 | mvdk | That creates mailbox 185 with password 567 for CoolGuy, with email address CoolGuy@example.com |
| 14:27.02 | mvdk | And that's in the voicemail context 'default', which is what you get if you don't specify one |
| 14:27.19 | TheCompWiz | the voice mail acts like it's not recieving any of the digits I dial... |
| 14:27.27 | asterboy | Does anyone have a good suggestion for a motherboard that will give the best zttest scores? |
| 14:27.57 | mvdk | asterboy: Whose dollars are you spending? |
| 14:28.05 | *** join/#asterisk stkn_ (n=stkn@gentoo/developer/pdpc.active.stkn) |
| 14:28.09 | key2 | when qualify=yes, asterisk checks the state of the interface all the time ? |
| 14:28.10 | asterboy | The client |
| 14:28.24 | asterboy | which of course translates to mine |
| 14:28.28 | mvdk | Then I recommend you get an Opteron server motherboard |
| 14:28.42 | mvdk | Almost any will work well |
| 14:28.46 | asterboy | cause right now I have a Gigabyte motherboard that sucks shit...and is causing me grey hair |
| 14:28.55 | feld_ | TheCompWiz, I've been told menus/voicemail can sometimes fail to hear your dialing if you have certain codecs/compression/other factors |
| 14:28.57 | *** join/#asterisk Arno[Slack] (n=root@66-163-12-60.ip.tor.radiant.net) |
| 14:28.58 | b00mer | if money wasn't a question... still Opteron |
| 14:28.58 | *** join/#asterisk thieumS (n=Mathieu@nor75-12-82-230-173-27.fbx.proxad.net) |
| 14:29.01 | b00mer | ? |
| 14:29.03 | mvdk | Or an intel one, point is, server motherboards are what will give you best results |
| 14:29.23 | TheCompWiz | feld_... what's the "recomended" codec/compression? |
| 14:29.27 | mvdk | That's true... |
| 14:29.33 | mvdk | I hadn't thought of that |
| 14:29.45 | feld_ | TheCompWiz, I dont have the experience to give that answer. Hopefully others can fill you in. |
| 14:29.47 | mvdk | Well, just try ulaw, at first |
| 14:29.50 | asterboy | oh ANY might work, but not if you define work as low zttest scores which translate to choppy or intermittent dtmf problems. |
| 14:29.55 | mvdk | That's completely uncompressed |
| 14:30.16 | feld_ | TheCompWiz, analog phones, SIP phones, what are you testing from? |
| 14:30.38 | TheCompWiz | well... I've got choices of: PCMU, PCMA, G.723.1, G.729A/B and GSM.... |
| 14:30.49 | TheCompWiz | feld_ they're the grandstream phones.... |
| 14:30.50 | mvdk | And ulaw, you'll find |
| 14:30.52 | asterboy | ya server motherboards should work well |
| 14:31.02 | feld_ | TheCompWiz, try this in your sip.conf in the phone section: dtmfmode=rfc2833 |
| 14:31.25 | mvdk | That puts the dtmf out of band |
| 14:31.25 | coppice | get a gigabyte. they are one of the biggest server board makers :-) |
| 14:31.26 | feld_ | I think that can solve the dialing recognition issue for some situations. |
| 14:31.36 | asterboy | Tyan, Supermicro, AsRock anyone? |
| 14:31.43 | feld_ | mvdk, can u explain more about that? maybe I've been misinformed. |
| 14:31.45 | mvdk | coppice: Have you considered that you might be talking out your butt? |
| 14:31.56 | feld_ | asterboy, Tyan |
| 14:32.03 | feld_ | with opterons |
| 14:32.05 | mvdk | feld_: Out of band as opposed to inband |
| 14:32.12 | asterboy | now that I have a GigaByte board which is really poorly working...I won't touch that brand. |
| 14:32.17 | coppice | gigabyte really is one of the biggest server board makers. personally i'd choose tyan |
| 14:32.27 | thieumS | i'm looking for a way to get the answeredtime value from the manager api |
| 14:32.31 | feld_ | asterboy, i had one melt @ the P1 connector and I sent it in to RMA and it never returned. |
| 14:32.34 | coppice | every board makers makes some stinkers |
| 14:32.46 | asterboy | ya, I was kinda thinking P4 was the way to go...but it seems the more important factor is NOT the CPU but the PCI Bus. |
| 14:32.50 | mvdk | Yeah, but I've found Tyan pretty good |
| 14:33.10 | feld_ | I have a K8WE Thunder for my home PC workstation. It's a monster. |
| 14:33.12 | asterboy | feld_, you talking the Tyan that melted |
| 14:33.18 | feld_ | asterboy, no gigabyte |
| 14:33.18 | coppice | I've seen some stinkers from tyan, but they are generally reasonable |
| 14:33.22 | asterboy | ah |
| 14:33.34 | *** part/#asterisk kmilitzer (n=km@office-gw.westend.com) |
| 14:33.35 | feld_ | Chaintech = Gigabyte too if anyone's wondering |
| 14:33.40 | mvdk | Anyway, point is, sending DTMF out of band is a far better idea |
| 14:33.41 | asterboy | I want a brand that consistently performs. |
| 14:33.43 | b00mer | are there any specific DELL server recommendations? |
| 14:33.51 | coppice | there isn;t one |
| 14:33.52 | mvdk | Yeah: don't :) |
| 14:33.56 | *** join/#asterisk paryl (n=chatzill@216-201-177-82.res.logixcom.net) |
| 14:34.07 | feld_ | b00mer, build your own lol |
| 14:34.21 | mvdk | Unless management has decreed it so |
| 14:34.34 | mvdk | But in that case, you can likely get a Sun, or something, anyway :) |
| 14:34.36 | asterboy | I don't use Dell...once you let them in with your logos...the company will start to call them direct. |
| 14:34.39 | feld_ | more control, no cheap power supply, no junk ram rebranded ram made in a sweat shop..... |
| 14:34.39 | docelmo | b00mer depends on what you wanna do you have many options to buils an asteris box |
| 14:34.45 | b00mer | someone told me the guys at switchvox have found some of the dell servers better than others due to the PCI bus |
| 14:34.53 | docelmo | from small solidstate to robust SUN boxes |
| 14:35.01 | docelmo | yes |
| 14:35.09 | feld_ | b00mer, dell doesnt make motherboards themselves. just find out what they're using and look for those boards. |
| 14:35.10 | b00mer | mgmt requires dell |
| 14:35.17 | mvdk | No, I'm just talking about their opteron stuff.... |
| 14:35.24 | feld_ | "Dude you're gettin a Dell!" |
| 14:35.28 | docelmo | But mainly its the interaction of the chipset your looking for |
| 14:35.30 | mvdk | b00mer, have you broached the topic of Sun to them? |
| 14:35.40 | feld_ | I hear Sun rocks |
| 14:35.43 | feld_ | I want one :( |
| 14:35.58 | asterboy | Ya, I don't think the CPU is near as important as the PCI Bus. |
| 14:36.06 | coppice | Dells are just the machines the Taiwanese makers would be embarassed to put their own names on |
| 14:36.07 | TheCompWiz | for mailbox /password do I need to end the mailbox # with a #? or just dail the numbers? |
| 14:36.20 | mvdk | Just dial it..... |
| 14:36.25 | *** join/#asterisk fulgas (n=fulgas@207.226.175.2) |
| 14:36.27 | TheCompWiz | didn't werk. :( |
| 14:36.36 | asterboy | Here is a list of Hardware Recommendations: http://www.voip-info.org/wiki/view/Asterisk+hardware |
| 14:36.37 | mvdk | Try ending it with a hash |
| 14:36.49 | TheCompWiz | hash? |
| 14:36.53 | asterboy | would be nice if there was a more comprehensive uptodate list though. |
| 14:36.57 | mvdk | Yeah, the # symbol |
| 14:37.03 | TheCompWiz | I did. still ignored it. |
| 14:37.21 | mvdk | Hmm, odd..... |
| 14:37.25 | TheCompWiz | yup. |
| 14:37.50 | mvdk | I've got to say, I don't usually use those kinds of phones.... |
| 14:38.02 | mvdk | Have you tried with an IAX soft phone first? |
| 14:38.04 | {zombie} | TheCompWiz: what have you got configured for DTMF mode? |
| 14:38.11 | asterboy | ~digium |
| 14:38.12 | jbot | well, digium is evil |
| 14:38.18 | asterboy | lol |
| 14:38.40 | mvdk | Just so we can see whether it's the last inch, or a couple of other inches? |
| 14:38.52 | {zombie} | TheCompWiz: in your sip.conf dtmfmode= ? |
| 14:39.35 | TheCompWiz | dtmfmode=rfc2833 |
| 14:39.43 | {zombie} | and your phone is set to that also? |
| 14:39.44 | asterboy | jbot, digium is also reachable here: http://www.digium.com/en/company/contact.php |
| 14:39.45 | jbot | okay, asterboy |
| 14:40.08 | RoyK | ~disclaimer? |
| 14:40.10 | jbot | I disclaim all of you!, or "fortune -m 'Void where'" |
| 14:41.36 | TheCompWiz | spiffy! it worked! |
| 14:41.49 | mvdk | Excellent! |
| 14:42.02 | mvdk | You changed the DTMF mode on the phone then, too? |
| 14:42.08 | TheCompWiz | just did... |
| 14:42.31 | mvdk | Excellent, IAX soft phones spoil you :) |
| 14:42.49 | mvdk | So do FXS sockets :) |
| 14:43.41 | saftsack | hi |
| 14:43.48 | TheCompWiz | I'm still getting the "The person at extension" <hangup> when I get transfered to voicemail after no answer. |
| 14:43.53 | saftsack | are there any guys who have a knowledge about the b410p isdn card? |
| 14:44.08 | mvdk | Well, have you recorded your new message? |
| 14:44.13 | TheCompWiz | yeah. |
| 14:44.16 | *** join/#asterisk DagoBlok (n=Dago@dD5771892.access.telenet.be) |
| 14:44.19 | *** part/#asterisk scrubb (n=scrubb@IP-216-37-19-40.nframe.com) |
| 14:44.20 | [TK]D-Fender | saftsack : Doesn't exist yet... just wait... |
| 14:44.25 | saftsack | ^^ |
| 14:44.26 | mvdk | And you don't hear it played? |
| 14:44.37 | saftsack | so i have to wait until winter? ^^ |
| 14:44.49 | [TK]D-Fender | TheCompWiz : You need to call your VM box with a prefix fo "b" to play the "busy" message, or "u" for the "unavailable" one... |
| 14:44.59 | [TK]D-Fender | saftsack : Nexdt Spring sharp! |
| 14:45.12 | TheCompWiz | isn't there a default "busy" message? |
| 14:45.21 | mvdk | Yeah, there is.... |
| 14:45.46 | mvdk | Of course, the fact that it's a cheesy American chick doesn't bother you at all.... |
| 14:45.46 | [TK]D-Fender | TheCompWiz : You need to tell * which messge to play... not everyone want to use distinct messages for busy & unavailable... |
| 14:46.22 | TheCompWiz | [TK]D-Fender ... in the dialing plan... that appears to be fine. |
| 14:46.23 | mvdk | Sorry, I'm nodding off |
| 14:46.30 | [TK]D-Fender | TheCompWiz : Pastebin it.... |
| 14:46.39 | *** join/#asterisk Vorondil (n=jkiffmey@miro.voltaiccommerce.com) |
| 14:48.26 | key2 | [TK]D-Fender: when qualify=yes, asterisk checks the state of the interface all the time ? right ? |
| 14:48.37 | mvdk | Not all the time.... |
| 14:48.43 | key2 | mvdk ? |
| 14:48.49 | mvdk | It uses some compile time default |
| 14:48.51 | [TK]D-Fender | key2 : Correct. if it times out you won't be able to receive calls on it till it reregisters again |
| 14:49.19 | mvdk | So it's like filling out qualify=1000, if 1000 were the compile time default |
| 14:49.36 | [TK]D-Fender | key2 : "qualify=yes" defaults to 2000ms. you can specify the frequency yourself like "qualify=4000" |
| 14:50.06 | mvdk | I've never used the default: I prefer to specify it :) |
| 14:50.15 | mut | i use 3000ms myself |
| 14:50.18 | [TK]D-Fender | mvdk : No reason to change the compile default, just set it in [general] and override where needed. |
| 14:50.25 | C4T3l | has anyone setup paging on Pcoms with sip ver 1.6.6??? |
| 14:50.47 | key2 | my question is if I have a type=peer, and I have let say 5 users on a proxy that are Queue Member, how can asterisk know their statut |
| 14:50.49 | mvdk | No, I mean, I specify qualify=<myfavouritenumber> in general |
| 14:50.53 | jahani | why when i change the port 5060 to 5065 i can't register to asterisk? |
| 14:50.59 | mvdk | I don't specify "qualify=yes" |
| 14:51.05 | key2 | since in a queue, asterisk needs to know the status of the user before sending someone to him |
| 14:51.12 | *** join/#asterisk aze (n=aze@ACayenne-101-1-10-77.w81-248.abo.wanadoo.fr) |
| 14:51.23 | *** join/#asterisk BudaH (n=buda@urano.mhnet.com.br) |
| 14:51.47 | mvdk | Anyway, I'll see you all later |
| 14:51.50 | mvdk | exit |
| 14:51.50 | [TK]D-Fender | key2 : * knows their status because * is the one that handles all calls in/out of the device. |
| 14:51.52 | *** join/#asterisk aze (n=aze@ACayenne-101-1-10-77.w81-248.abo.wanadoo.fr) |
| 14:52.06 | TheCompWiz | hm... sounds like the default messages aren't working correctly. After I set an unavailable message... it works fine... but I can't get the "the person at extension" to read the numbers & such... |
| 14:52.30 | [TK]D-Fender | TheCompWiz : Where's my pastebin? messages work just fine for everybody else.... |
| 14:52.45 | key2 | [TK]D-Fender: but what if the phone is not registered on asterisk but on a proxy ? |
| 14:52.56 | *** join/#asterisk jono (n=jono@mail.openadvantage.org) |
| 14:53.22 | Dr-Linux | Jun 8 07:39:45 WARNING[31412]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 14:53.31 | [TK]D-Fender | key2 : Well * will assume its the only source to send a call there so yeah, it'll send calls to busy people if thats not the case. |
| 14:53.40 | Dr-Linux | why i'm getting this while i use allow=gsm codec in sip.conf? |
| 14:53.45 | Dr-Linux | anybdoy have any idea? |
| 14:55.01 | [TK]D-Fender | Dr-Linux : down your server, manually wipe the modules folder. do a complete recompile and install. |
| 14:55.46 | Dr-Linux | [TK]D-Fender: what kind of solution is this? |
| 14:56.39 | [TK]D-Fender | Dr-Linux : makes sure you aren't using a bad link somewhere... you're the only person getting this so far... so do a clean reinstall. Can't hurt |
| 14:57.16 | Dr-Linux | [TK]D-Fender: only GSM codec makes this warnings |
| 14:57.29 | *** join/#asterisk BudaH (n=buda@urano.mhnet.com.br) |
| 14:57.33 | Dr-Linux | [TK]D-Fender: so how can i re-install the entire things |
| 14:58.04 | [TK]D-Fender | Dr-Linux : By doing exactly what I said. |
| 14:59.10 | Dr-Linux | [TK]D-Fender: if you were, whould you re-complile everything if you are getting GSM warning? |
| 14:59.43 | paryl | i'm having strange issues with my queues... i have the strategy set to leastrecent, but some agents will get all calls right in a row for say, 4-5 calls, and then it seems to even out |
| 15:00.13 | *** join/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 15:00.13 | paryl | i thought when i updated to the latest version that would go away, but it's been an issue for quite some time |
| 15:00.15 | key2 | [TK]D-Fender: well the problem is that when my Queue Member is in communication, as soons as someone gets into the queue, asterisk sends the INVITE to the Queue Member so he hears a bip while being in communication |
| 15:00.29 | key2 | [TK]D-Fender: is there a way from the Dialplan to set the status of a user ? |
| 15:00.42 | _Sam-- | Dr-Linux : its not that hard of a solution...cd /usr/lib/asterisk/modules mv modules modules.sav make install |
| 15:00.43 | key2 | [TK]D-Fender: like can I force someone to be In Use ? |
| 15:01.40 | JackEstorm | key2: read up on agents and {Un}PauseQueueMember |
| 15:02.23 | C4T3l | has anyone setup paging on Pcoms with sip ver 1.6.6??? I can't seem to make the phone auto-answer. The wiki shows help for an outdated sip version |
| 15:02.29 | [TK]D-Fender | key2 : * should send calls to someone on a call routed by * already... |
| 15:02.40 | MikeJ[Laptop] | ~seen jbot |
| 15:02.56 | jbot | jbot is currently on #asterisk-doc (1d 17h 43m 44s) ##t42 (1d 17h 43m 44s) #how (1d 17h 43m 44s) #ol (1d 17h 43m 44s) #flyspray (1d 17h 43m 44s) #asterisk (1d 17h 43m 44s) #byumug (1d 17h 43m 44s) #va (1d 17h 43m 44s) #orkut (1d 17h 43m 44s) #nslu2-linux (1d 17h 43m 44s) ##ducleague ... |
| 15:02.56 | TheCompWiz | hm... it's not reading ANY numbers... |
| 15:03.01 | _Sam-- | key2: i think what are you talking is on the phone side. |
| 15:03.04 | *** join/#asterisk jpeeler (n=jpeeler4@host86-129-192-76.range86-129.btcentralplus.com) |
| 15:03.22 | Dr-Linux | _Sam--: same i unload the "format_gsm.so" and "codec_gsm.so" |
| 15:03.29 | Dr-Linux | not let me load them again, and lets see |
| 15:03.29 | TheCompWiz | for instance... if I have 2 messages in my inbox... it says "you have messages in your inbox" .... and 1 message = "you have message in your inbox" |
| 15:03.44 | MikeJ[Laptop] | busy bot |
| 15:03.56 | *** join/#asterisk ToyMan (n=stuq@rrcs-24-97-206-117.nys.biz.rr.com) |
| 15:04.28 | [TK]D-Fender | TheCompWiz : I'm betting your "digits" folder is screwed |
| 15:04.52 | TheCompWiz | I think so too. |
| 15:05.14 | _Sam-- | check /usr/share/asterisk/sounds/digits |
| 15:05.21 | _Sam-- | at least, thats where mine are. |
| 15:05.52 | Dr-Linux | really this problem is killing me :( |
| 15:05.56 | [TK]D-Fender | should be in /var/lib/sounds/asterisk/digits |
| 15:06.13 | [TK]D-Fender | Dr-Linux : Just flush it all and reinstall already.... |
| 15:06.35 | Dr-Linux | [TK]D-Fender: can i do it with only GSM codec? |
| 15:07.01 | Dr-Linux | [TK]D-Fender: bcoz it's gonna peak hours start and i can't take risk, maybe something goes wrong :S |
| 15:07.29 | RoyK | Dr-Linux: what's wrong with alaw? |
| 15:07.29 | TheCompWiz | is there a .conf someplace that keeps track of location of the digits folder? |
| 15:07.39 | RoyK | asterisk.conf |
| 15:07.40 | _Sam-- | you could make modules then copy over codec_gsm.so to the modules dir? |
| 15:07.56 | Dr-Linux | RoyK: all works fine, but only GSM gives error |
| 15:07.58 | _Sam-- | TheCompWiz: locate digits |less and look for asterisk |
| 15:08.14 | TheCompWiz | _Sam--... I've found the folder... and everything looks fine. |
| 15:08.22 | key2 | JackEstorm: so it would mean that if a queue member is not registered on asterisk and just added by AddQueueMember(Queue_name|SIP/contact@my_proxy_where_he_is_registered)) |
| 15:08.30 | RoyK | Dr-Linux: if it hurts, don't do it |
| 15:08.37 | Dr-Linux | RoyK: but alaw tooks lot of bandwidth |
| 15:08.44 | RoyK | takes |
| 15:08.54 | RoyK | Dr-Linux: what sort of error? |
| 15:09.28 | Dr-Linux | Jun 8 07:52:14 WARNING[31706]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 15:09.47 | Dr-Linux | i get this loop for hundred of time, while codec executes |
| 15:10.09 | *** join/#asterisk tamp4x (n=Lab@64.201.13.51) |
| 15:10.27 | RoyK | Dr-Linux: perhaps the client doesn't talk the same dialect of gsm as asterisk does |
| 15:10.37 | RoyK | or perhaps there's something wrong with asterisk's gsm support |
| 15:10.52 | *** join/#asterisk r124 (n=r00t@203.88.88.237) |
| 15:11.00 | TheCompWiz | yipee! permission problems. |
| 15:11.00 | Dr-Linux | should i copy codec_gsm.so module from my other asterisk server? |
| 15:11.04 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 15:11.05 | r124 | hi |
| 15:11.13 | r124 | can u guys help me ? |
| 15:11.23 | TheCompWiz | can you ask a question that relates to your problem? |
| 15:11.36 | JackEstorm | key2: you need to add some logic to your dial plan, so that when a phone is dialed or dials it is paused in the queues it belongs to (or you can configure your phones to disable callwaiting, but that only works with phones that support that) |
| 15:12.03 | r124 | i figured out, can i set up asterisk to connected to some PSTN line |
| 15:12.07 | r124 | ?? |
| 15:12.10 | feld_ | does it require extra settings to make the "availability" work for X-lite & Asterisk ? It doesnt seem to ever show that someone is online, though I have it set in X-Lite to show that. |
| 15:12.12 | feld_ | r124, yes |
| 15:12.25 | _Sam-- | feld: hint() |
| 15:12.28 | r124 | with generic modem ? |
| 15:12.49 | feld_ | _Sam--, i think i have hinting enabled |
| 15:12.51 | feld_ | let me check |
| 15:12.52 | JackEstorm | feld: I only use Polycom phones and Cisco ATA's right now |
| 15:12.59 | feld_ | r124, no not with a generic modem |
| 15:13.09 | _Sam-- | its in extensions.conf -- its not something you really enable |
| 15:13.10 | [TK]D-Fender | feld : You need to set up your dial-plan "hint"'s |
| 15:13.15 | r124 | maybe some winmodem with lucent or realtek chipset ? |
| 15:13.21 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 15:13.43 | [TK]D-Fender | r124 : No, only Intel 537 based winmodems are supported. |
| 15:13.49 | r124 | wow |
| 15:13.56 | r124 | intel 537 ?? |
| 15:13.57 | [TK]D-Fender | r124 : And they aren't so great |
| 15:14.02 | *** join/#asterisk wunderkin (n=wunderki@69.26.192.234) |
| 15:14.10 | [TK]D-Fender | r124 : </echo> |
| 15:14.31 | r124 | :p |
| 15:14.47 | r124 | thanks anyway |
| 15:15.02 | r124 | i'll to search the modem |
| 15:15.02 | JackEstorm | so is it normal for Set(CALLERID(number)=${AGENTBYCALLERID_${CALLERID(num)}} to kill all globals? or did I just find another agent related bug? |
| 15:15.07 | r124 | i'll try |
| 15:15.08 | [TK]D-Fender | r124 : Thats the only "modem" that works, there are specialty cards for everything else. |
| 15:15.11 | r124 | thanks |
| 15:15.15 | _Sam-- | feld: http://www.voip-info.org/wiki/index.php?page=Asterisk+standard+extensions |
| 15:15.18 | _Sam-- | has info about hints |
| 15:15.33 | *** part/#asterisk r124 (n=r00t@203.88.88.237) |
| 15:16.43 | *** join/#asterisk fourcheeze (n=rich@82.153.23.79) |
| 15:17.30 | fourcheeze | coppice: thanks for the advice re: faxes yesterday - looks like I'm going to have to bite the bullet and go for t.38 - what do you recommend as a client endpoint? |
| 15:18.56 | iq | Hi, Anyone using h323 with Asterisk? Document says that I MUST build PWLib AND Openh323 fromt he source. |
| 15:19.14 | *** join/#asterisk Faithful (n=Faithful@ns.linuxterminal.com) |
| 15:19.43 | CunningPike | Morning all |
| 15:19.55 | fourcheeze | iq, I hated h323 with asterisk so I cheated and use Yate as an h323->sip signalling proxy |
| 15:19.59 | [TK]D-Fender | CunningPike : y0 |
| 15:20.05 | iq | Any idea where 'bison.simple' comes from |
| 15:20.05 | *** join/#asterisk littlejohn (n=little@host31-85.pool877.interbusiness.it) |
| 15:20.24 | *** join/#asterisk salviadud (n=ralfalfa@201.133.207.93) |
| 15:20.33 | CunningPike | Hey, [t |
| 15:20.39 | iq | fourcheeze: thats an option as well. I wanted to give it a try first. |
| 15:20.48 | Hmmhesays | go around a time or two, just to waste my time with you |
| 15:21.52 | fourcheeze | hmm is there a t.37 howto for asterisk? |
| 15:22.08 | Hmmhesays | Wu Mart making waves as retailer in China. Tawget, Crostco, and Blest Bly not far behind |
| 15:22.27 | MrChimpy | hmm, I'm a bit confused about trunk config on my 4 port PRI ISDN cards |
| 15:22.42 | Kis | finally... vacancy... |
| 15:22.46 | MrChimpy | I'm dialling on stuff like Zap/g1 |
| 15:23.05 | MrChimpy | which is fine, but I need it to find a line on any of my four trunk groups |
| 15:23.08 | *** join/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.216.Dial1.SanJose1.Level3.net) |
| 15:23.27 | MrChimpy | do I do that in the Dial command, or should I just be defining one trunk group? |
| 15:23.47 | robin_sz | ok, be this about my poxy SNOM360 explaining ... i have MWI and a retreive button .. when the MWI is unlit, I press retreive and it dials 8501 *which is what I need it to do* and it works .. when the MWI is flashing, I press retreive and the danm thing dials asterisk@192.168.1.1 ?? whys that ?? |
| 15:24.15 | CunningPike | MrChimpy: Try a single group= line before all your channel => lines |
| 15:24.34 | file | robin_sz: configure the option in sip.conf to change it to 8501 |
| 15:24.50 | robin_sz | its in sip.conf? |
| 15:24.52 | robin_sz | coo. |
| 15:24.54 | file | vmexten |
| 15:24.57 | file | in the general section |
| 15:25.08 | robin_sz | wow. |
| 15:27.07 | CunningPike | Good to know, CodyC |
| 15:30.36 | noky | somebody test the performance of MeetMe's Application ? |
| 15:30.57 | feld_ | ok guys for hinting I am doing the following. can you please tell me if there's a one-liner that takes care of both? : |
| 15:31.17 | CunningPike | noky: Check the list - I think someone has done some testing with concurrent users/concurrent conferences |
| 15:31.21 | feld_ | exten => 2000,hint,${PHONE0} |
| 15:31.21 | feld_ | exten => 2000,1,Macro(oneline,${PHONE0}) |
| 15:31.36 | asterboy | where do you start with Festival setup in * |
| 15:31.42 | asterboy | the docs are not so clear |
| 15:31.43 | feld_ | can that be put in one line like 2000,1,hint,Macro(.... or will that fail? |
| 15:32.06 | robin_sz | file: cool. works. I didn t have that option even commented out in my sip.conf, thats probabyll how I missed it |
| 15:32.08 | asterboy | I have the app loaded, however, there is no festival.conf or festival.scm |
| 15:32.13 | CunningPike | feld_: It will fail - you can only have one priority per line |
| 15:32.14 | file | ah |
| 15:32.20 | feld_ | CunningPike, thanks! |
| 15:32.21 | asterboy | the source is in contribs |
| 15:32.34 | C4T3l | festival uses up a lot of proc |
| 15:32.54 | asterboy | ya but if I want to play zasterisk...I must have it. |
| 15:33.32 | C4T3l | have you tried to run it at linux CLI? does it work there? |
| 15:33.43 | asterboy | what is the syntax? |
| 15:33.46 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 15:33.51 | *** part/#asterisk CodyC (n=cody@207.200.23.194) |
| 15:34.10 | robin_sz | should I go around all 40 phones and configure them .. or is there someting I can do to * to make it less worried? |
| 15:34.35 | C4T3l | festival <enter> then type: SayText("hello world") <enter> |
| 15:35.03 | asterboy | no such command |
| 15:35.42 | C4T3l | hmm, then you need to install it :) |
| 15:36.26 | asterboy | where are the docs on the install? |
| 15:36.31 | robin_sz | sigh ... bring me a decent SIP termination |
| 15:36.55 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 15:36.57 | asterboy | this does not seem to be much help: http://www.voip-info.org/wiki-Asterisk+Festival+installation |
| 15:37.44 | *** join/#asterisk burizaa (n=freeee@cm56.omega110.maxonline.com.sg) |
| 15:37.51 | C4T3l | asterboy: what distro you use? |
| 15:38.06 | asterboy | lfs |
| 15:38.11 | asterboy | ~festival |
| 15:38.14 | jbot | well, festival is a general multi-lingual speech synthesis system developed at CSTR. See http://www.cstr.ed.ac.uk/projects/festival/, or festival lite a much more compact festival http://www.speech.cs.cmu.edu/flite/index.html |
| 15:40.52 | *** join/#asterisk Neptune__ (n=foo@zux221-156-100.adsl.green.ch) |
| 15:42.06 | asterboy | man, too complicated....festival needs a hell of a bunch of requirements. |
| 15:42.06 | *** join/#asterisk unixgeek (n=unixgeek@216-220-234-197.exploremaine.com) |
| 15:42.12 | asterboy | so much for zasterisk |
| 15:45.01 | Neptune__ | what ports do i need to allow for *? 5060 and the range in rtp? |
| 15:47.52 | *** join/#asterisk Mother (n=m@253.Red-88-12-225.dynamicIP.rima-tde.net) |
| 15:47.58 | Mother | greetings |
| 15:48.08 | Qwell[] | Neptune__: yes |
| 15:48.15 | Qwell[] | Neptune__: unless you use anything besides sip |
| 15:48.16 | Mother | anyone know the payload type for Speex in RTP? |
| 15:49.08 | file | yeekz, as a software developer putting together a desk I feel... silly |
| 15:49.29 | Qwell[] | file: pfft |
| 15:49.29 | Mother | consider it bit assembly |
| 15:49.40 | Qwell[] | You have the Makefile, right? |
| 15:49.43 | file | ooh |
| 15:49.45 | file | make desk |
| 15:49.46 | Qwell[] | and a compiler? |
| 15:50.01 | Qwell[] | ./configure --with-cdrack |
| 15:50.02 | Qwell[] | such a nub |
| 15:50.08 | Mother | ## unresolved dependency: size 10 spanner |
| 15:50.27 | file | I'm using a Digium/Asterisk screwdriver :D |
| 15:50.39 | Mother | ah! then you should be OK |
| 15:50.40 | Qwell[] | checking for phillips head screwdriver: yes |
| 15:50.42 | TheCompWiz | Mother shoulda used yum. |
| 15:50.45 | Neptune__ | Quell - would that be tcp or udp? |
| 15:50.46 | Qwell[] | checking for phillips head screwdriver usability: no |
| 15:50.49 | Qwell[] | Neptune__: udp |
| 15:50.50 | Mother | lol |
| 15:50.55 | Mother | yum install table |
| 15:50.55 | Neptune__ | ok thanks |
| 15:51.03 | Qwell[] | BAH! |
| 15:51.17 | Qwell[] | USE="glass" emerge -u desk |
| 15:51.32 | Mother | http://www.iana.org/assignments/rtp-parameters <- I don't see Speex here |
| 15:51.50 | Mother | so just looking for a creative idea before I set it as 1337 |
| 15:54.11 | rpm | how do allow my phones in asterisk to retain their caller ID and not have asterisk set it? |
| 15:54.14 | Dr-Linux | Qwell[]: a call needs good upload speed or download speed? |
| 15:54.15 | file | I can't find a part |
| 15:54.16 | file | hrm |
| 15:54.23 | Qwell[] | Dr-Linux: both, it needs the same amount of each |
| 15:54.33 | noky | somebody test the performance of MeetMe's Application ? |
| 15:54.39 | Qwell[] | file: ./configure --without-doorhinge |
| 15:54.40 | noky | i don't found in the list |
| 15:54.56 | Dr-Linux | Qwell[]: i'm having some issues, not sure what do do |
| 15:55.03 | Qwell[] | ~consultant |
| 15:55.05 | jbot | Hire a consultant. |
| 15:55.05 | Qwell[] | :D |
| 15:55.13 | Qwell[] | Dr-Linux: What's the issue? |
| 15:55.25 | Dr-Linux | heh :) i'm .. i'm not bad ;) |
| 15:55.38 | Mother | file: I bet it's from Ikea |
| 15:55.42 | file | nah |
| 15:55.43 | Dr-Linux | Qwell[]: when i use GSM codec in the sip.conf |
| 15:55.44 | file | Office Depot |
| 15:55.46 | Mother | ah |
| 15:55.48 | Dr-Linux | i get loop : |
| 15:55.48 | Dr-Linux | Jun 8 08:18:24 WARNING[32223]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 15:55.49 | Qwell[] | file: worse |
| 15:55.50 | Dr-Linux | Jun 8 08:18:24 WARNING[32223]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 15:55.53 | file | I think I'm missing side frames |
| 15:56.00 | Mother | they tend to do the "missing part" thing rather often |
| 15:56.01 | Dr-Linux | hundreds of time .. |
| 15:56.04 | Mother | that's more serious then |
| 15:56.05 | Qwell[] | file: I once bought a chair, with two of the same arm |
| 15:56.11 | Qwell[] | so, like...it didn't fit, heh |
| 15:56.13 | file | neat |
| 15:56.41 | Dr-Linux | also, sometime i hear like someone is talking from understand water |
| 15:56.46 | Mother | the other day I saw a friend got a bookcase - required items as per the instructions: a screwdriver, a hammer and a friend |
| 15:56.57 | Dr-Linux | Qwell[]: do you think it's RTP or bandwidth issu? |
| 15:57.00 | Mother | so if you had no friends you had to return the bookcase |
| 15:57.03 | Qwell[] | Dr-Linux: neither |
| 15:57.08 | Dr-Linux | Qwell[]: my US clients are just fine with the same server |
| 15:57.15 | Qwell[] | fun |
| 15:57.34 | Dr-Linux | Qwell[]: neither? |
| 15:58.09 | *** join/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 15:58.27 | *** join/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 15:58.33 | file | wait wait |
| 15:58.34 | Kerry_G | anyone know of a way to listen to voicemail when sent as email to a blackberry? |
| 15:58.39 | file | I think I know where they are |
| 15:58.45 | Qwell[] | file: in the box?! |
| 15:59.23 | *** join/#asterisk mut (n=animenod@65.111.222.120) |
| 15:59.26 | mut | yay services! |
| 15:59.31 | Hmmhesays | Kerry_G: you mean you can read audio data as a text file and decode it in your brain? |
| 15:59.34 | Hmmhesays | *can't |
| 15:59.47 | noky | where can i found the last bugs of asterisk ? |
| 16:00.28 | *** join/#asterisk lorinc (n=ang@caracas-0343.adsl.interware.hu) |
| 16:00.40 | Qwell[] | noky: bugs.digium.com |
| 16:00.59 | Dr-Linux | file: do you have any clue why i'm getting this warnings ? |
| 16:01.56 | file | got them! |
| 16:02.07 | file | Dr-Linux: I'm busy putting together a desk, please heave your message after the beep |
| 16:02.09 | Qwell[] | file: where were they? in the desk?! |
| 16:02.12 | file | er leave |
| 16:02.20 | Qwell[] | heave your message? heh |
| 16:02.28 | file | Qwell[]: well there's this foam thing... with a big glass top on top, and under neath there's all sorts of stuff |
| 16:02.33 | Qwell[] | ooo |
| 16:02.35 | Dr-Linux | file: ok, i'll catch you next time |
| 16:03.36 | swytch | q about dtmf. i set dtmfmode=info in sip.conf [global] however, in all SIP messages (100,180,183 and 200) from asterisk i dont have Allow: INFO. In the INVITE that i sends i have... |
| 16:06.06 | Kerry_G | no no no, I just want to play the sound file attachment |
| 16:06.12 | CunningPike | rpm: Just make sure there is no callerid line in sip.conf...... |
| 16:06.22 | CunningPike | Kerry_G: Blackberries can't do that..... |
| 16:06.24 | Kerry_G | and blackberrys dont recognize the default format |
| 16:06.44 | swytch | anyone have a hint why i may not get "Allow: INFO" from asterisk when i set dtmfmode=info ? |
| 16:07.06 | Kerry_G | it sees the attachment, but wont open it |
| 16:07.22 | rpm | CunningPike: and it will take the callerID from the phone? |
| 16:08.07 | CunningPike | rpm: Yes - we want it the other way around, so I know that's what happens :D |
| 16:08.46 | CunningPike | rpm: All this assumes that your PSTN connection allows you to set your own CID |
| 16:08.49 | *** join/#asterisk Alric (n=nbowyer@wireless-062.1stel.com) |
| 16:09.37 | *** join/#asterisk crich1999 (n=crich@pd956852e.dip0.t-ipconnect.de) |
| 16:10.14 | *** join/#asterisk bancus (n=treed@static-71-160-206-211.lsanca.dsl-w.verizon.net) |
| 16:10.14 | Neptune__ | why am i always getting a error on the CLI about failed registration for the local sip peers? - it always gives me a "username/authname mismatch" |
| 16:10.36 | *** join/#asterisk pa (n=paolo@unaffiliated/pa) |
| 16:10.51 | zoa | because you are using the wrong username / pass |
| 16:10.53 | zoa | or none at all |
| 16:10.54 | bancus | Quick question: I'm designing a PBX system for a company. (Actually three closely related companies, so there'll be multiple phone numbers involved.) Can anyone recommend good reliable PTSN<->SIP/IAX2 providers for the US? |
| 16:11.11 | bancus | I use broadvoice for our own company's PBX, and occaisionally have issues with them. |
| 16:11.14 | Kerry_G | NexVortex, Teliax |
| 16:11.58 | Neptune__ | zoa - no im positive that it is correct |
| 16:13.42 | swytch | quit |
| 16:14.08 | rpm | CunningPike: yes it does. |
| 16:14.12 | rpm | CunningPike: thanks |
| 16:14.20 | zoa | no its not |
| 16:16.51 | Dr-Linux | anthm: active? :) |
| 16:16.56 | anthm | ? |
| 16:17.38 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 16:17.41 | Dr-Linux | anthm: as i reported my last day GSM codec problem to you, do you remember? |
| 16:17.53 | Dr-Linux | Jun 8 08:18:24 WARNING[32223]: codec_gsm.c:194 gsmtolin_framein: Invalid GSM data |
| 16:18.26 | Dr-Linux | so had changed the codec in the sip.conf andi couldn't see this warning again .. |
| 16:18.38 | *** join/#asterisk Assid (i=assid@203.115.83.214) |
| 16:18.45 | Dr-Linux | but you were something right, it looks RTP bad packets/link problem |
| 16:19.23 | Dr-Linux | anthm: we are facing very bad voice quailty problem, like someone is talking from understand the water etc |
| 16:19.38 | Dr-Linux | anthm: my US users are very fine with the same server |
| 16:20.11 | Neptune__ | zoa i tried this a couple of times - the username/password in the sip.conf is the same as in the config of the phone |
| 16:20.28 | zoa | check for other users in sip.conf that dont have a pass |
| 16:21.22 | Dr-Linux | anthm: i was told to recompile my asterisk, but i'm doubted that it's my network/bandwidth problem |
| 16:21.55 | *** join/#asterisk tanvalmg (n=mguevarr@67-134-234-194.dia.static.qwest.net) |
| 16:21.59 | zoa | recompling asterisk probably will not work |
| 16:22.06 | zoa | i'd say go for ethereal |
| 16:22.09 | zoa | and look how the packets look |
| 16:22.13 | zoa | see if they are the same size |
| 16:22.23 | zoa | and they come every 20ms |
| 16:22.31 | zoa | and the timestamps look reasonable |
| 16:23.23 | brettnem | hey anyone know if there are problems with sending RTP over bonded pipes.. like bonded T1s, MPPP, or bonded SDSL lines? |
| 16:23.34 | emrah | exit |
| 16:23.52 | zoa | exit yourself :p |
| 16:23.53 | anthm | what is the call connected to on the other end |
| 16:24.36 | CunningPike | rpm: Great - glad it worked |
| 16:24.47 | brettnem | Session Border Controller->riverstone router->Asterisk->Riverstone Router->Paradyne DSLAM->Ethernet CPE->Polycom |
| 16:26.00 | tanvalmg | Hi, we're having problem installing our Digium TDM400P card on our IBM x346 server. We are getting kernel messages of "Uhhuh. NMI received for unknown reason 25 n CPU 0....Dazed and confused, but trying to continue...Do you have a stange power saving mode enabled?" |
| 16:27.01 | tanvalmg | ...when we do a "modprobe wcfxs" the system reboots |
| 16:29.20 | *** join/#asterisk CodyC (n=cody@207.200.23.194) |
| 16:29.47 | Dr-Linux | [TK]D-Fender: i just verified that's not asterisk problem, that's our DSL provider's problem |
| 16:30.00 | CodyC | anyone had a problem with a sip phone's dialplan screwing up an outbound call? |
| 16:30.37 | Dr-Linux | [TK]D-Fender: but odd thing is that, all other things work fine, but only voice stuff have problems, not only with asterisk, even with all our other voice devices. |
| 16:31.04 | *** part/#asterisk mhnoyes (n=mhnoyes@dialup-4.246.235.216.Dial1.SanJose1.Level3.net) |
| 16:31.04 | Neptune__ | i now got the grandstream phone to seamingly be registered - it shows when i do a "sip show peers" on the * cli - but the phone's status page says it isnt registered |
| 16:31.10 | Neptune__ | could this be a firewall issue? |
| 16:31.36 | *** join/#asterisk JakBeatZ (n=JakBeatZ@trek.tor1.ebit.ca) |
| 16:33.59 | *** join/#asterisk iq__ (n=iq@71-215-63-190.omah.qwest.net) |
| 16:34.25 | *** part/#asterisk iq__ (n=iq@71-215-63-190.omah.qwest.net) |
| 16:34.58 | *** join/#asterisk iq (n=iq@71-215-63-190.omah.qwest.net) |
| 16:35.54 | JakBeatZ | Hi Folks, I'm having an issue with a new asterisk box I'm setting up. I have it set up like this (PSTN)-- [ASTERISK1] --(IAX)-- [ASTERISK2] --(IAX)-- [ASTERISK3]. I have a number that's answered on Asterisk 1 via PSTN, it forwards it to asterisk 2 and I just setup IAX beteeen Asterisk 2 and Asterisk 3 and told Asterisk 2 to send the call to Asterisk 3, but for some reason, when I call I see this message on astterisk 2 "Jun 8 12:30:35 NOTICE[52 |
| 16:35.54 | JakBeatZ | 77]: chan_iax2.c:7183 socket_read: Host (asterisk 1 IP) failed to authenticate as (asterisk 3 IP)". I'm confused as to why asterisk asterisk 1 is trying to authenticate to aseterisk 3?! Is there some sort of proxy action I have to setup, somewhere? |
| 16:39.18 | Assid | heya tkd |
| 16:40.32 | Assid | okay question on dialplan for poly501.. when a user dials the dialplan.. the last digit is being dialled when they dial with another prefix besides 1xxxxxxxxxx |
| 16:41.35 | salviadud | why send it to asterisk 2? |
| 16:41.43 | salviadud | you can send it directly to asterisk 3 |
| 16:41.53 | salviadud | looks kinda pointless |
| 16:42.01 | JakBeatZ | salviadud: because I don;t own asterisk 1 |
| 16:42.20 | salviadud | :( |
| 16:42.30 | JakBeatZ | and I'm just doing some testing at the moment so I don't want to swing the DIDs around until I'm done the testing |
| 16:42.41 | JakBeatZ | so that's why I just thought I could do the multi-iax hops |
| 16:43.07 | *** part/#asterisk Kerry_G (n=Kerry_G@ip70-187-129-227.oc.oc.cox.net) |
| 16:43.25 | *** join/#asterisk jtodd (n=jtodd@ti.fox-den.com) |
| 16:46.59 | *** join/#asterisk littlejohn (n=little@host31-85.pool877.interbusiness.it) |
| 16:48.22 | rajiv|work | this "hacker" who resold voip service ... anyone know which company it was? |
| 16:48.55 | *** join/#asterisk anonymouz666 (i=anonymou@200.218.196.5) |
| 16:49.26 | dlynes_home | rajiv|work: huh? |
| 16:50.00 | dlynes_home | rajiv|work: to what are you referring? I don't see any mention of hacker for at least the last three screens or so |
| 16:50.21 | rajiv|work | dlynes_home: a slashdot article, nto mentioned here int he channel |
| 16:50.30 | dlynes_home | ah |
| 16:50.30 | rajiv|work | sorry i wasnt clear |
| 16:50.56 | *** join/#asterisk JINDAL (n=root@220.226.36.2) |
| 16:50.57 | dlynes_home | And when you say hacker, do you mean the media definition, or some other definition? |
| 16:51.20 | rajiv|work | media |
| 16:51.30 | dlynes_home | ah...no idea about that then |
| 16:51.33 | rajiv|work | http://it.slashdot.org/article.pl?sid=06/06/07/1949258 |
| 16:52.22 | dlynes_home | heh |
| 16:52.23 | dlynes_home | cool |
| 16:53.07 | JINDAL | hi guys... |
| 16:53.10 | dlynes_home | damn...that 1M would just barely get him a piece of property here |
| 16:53.21 | dlynes_home | he wouldn't have any money left over for a boat, and two cards |
| 16:53.26 | dlynes_home | s/cards/cars/ |
| 16:53.38 | *** join/#asterisk meshuga (i=meshuga@c-71-231-139-8.hsd1.or.comcast.net) |
| 16:55.26 | noky | where can i found the last bugs of asterisk ? |
| 16:55.33 | meshuga | anyone got a moment to look at a bizaare error? |
| 16:56.10 | JINDAL | am looking to asterisk as a solution for 500 people call centre..... but dont hav an idea abt the kinda hardware required...... any ideas |
| 16:56.22 | meshuga | JINDAL 3 servers |
| 16:56.38 | meshuga | 1 SER box and 2 asterisk boxes at minimum |
| 16:56.50 | meshuga | http://pastebin.com/767696 |
| 16:56.54 | dlynes_home | rajiv|work: damn..that guy had a good score going on |
| 16:57.00 | meshuga | so i am using DISA and trying to dial out using voipjet |
| 16:57.03 | rajiv|work | dlynes_home: heh ya |
| 16:57.04 | meshuga | it works fine from my sip client |
| 16:57.11 | JINDAL | okey... am supposing to handle 100+ concurrent calls |
| 16:57.19 | rajiv|work | dlynes_home: but the article doesnt say what his "company" name was |
| 16:57.25 | meshuga | but when i dial out using another SIP extension (ie remote dialtone) it fails |
| 16:57.33 | dlynes_home | rajiv|work: bet he really pissed off those ten carriers though :) |
| 16:57.41 | meshuga | JINDAL * can handle 300ish on good hardware. this is a pci bus limitation, not an asterisk limitation. |
| 16:57.59 | meshuga | well, a bus limitation, not necessarily pci :) |
| 16:58.23 | dlynes_home | rajiv|work: but 500K calls doesn't amount to the amount of money they're insinuating, either, unless they're all international calls |
| 16:59.01 | JINDAL | okey... meshuga, can ye elabarate on good hardware / ser boxes or pass a link -> |
| 16:59.08 | dlynes_home | rajiv|work: and even then, if that was the case, he would've made a lot more than $1M |
| 16:59.48 | *** join/#asterisk phin (n=mcgee@net2.netexp.com) |
| 16:59.52 | phin | hello |
| 17:00.26 | phin | i was wondering, before i install, and taking hardware into consideration still, would people recommend asterix on a ubuntu/debian based system, or freebsd? |
| 17:01.13 | [Airwolf] | phin, depends. |
| 17:01.25 | [Airwolf] | Are you planning on using any special hardware ? |
| 17:01.33 | meshuga | JINDAL thats too broad of a question to even answer. have you done any experimenting with * yourself? |
| 17:01.46 | [Airwolf] | As far as I know, zaptel drivers aren't avalible voor bsd. |
| 17:01.59 | phin | [Airwolf]: ahh, ok, thanks |
| 17:02.06 | viperdude | hi, anybody got any ideas on how to monitor asterisk boxes and how to send a email if there is a problem with them? |
| 17:02.11 | meshuga | anyone have any idea why voipjet would work from a SIP extension, but not thru DISA with remote dialtone? http://pastebin.com/767705 |
| 17:02.11 | phin | well i suppose i'll go the ubuntu or debian route |
| 17:02.28 | meshuga | viperdude: same with you monitor any other unix server. |
| 17:02.45 | meshuga | s/with/way |
| 17:02.50 | [Airwolf] | phin, nothing against ubuntu. But Ubuntu is kinda new. |
| 17:03.07 | viperdude | ok so is the a nagios plugin or the like? |
| 17:03.18 | phin | right |
| 17:03.19 | viperdude | or jffnms |
| 17:03.28 | meshuga | viperdude: exactly. |
| 17:03.37 | meshuga | or bigbrother or any network monitoring tool |
| 17:03.48 | viperdude | they can monitor SIP? |
| 17:03.55 | meshuga | heh |
| 17:04.01 | JINDAL | yup am on it presently..... and wanna buy hardware to start full time testing.............. just suggest me the minimum hardware i should blindly go for at this moment (500 people arnd 200+ concurrent PSTN calls) |
| 17:04.10 | meshuga | those apps monitor services by telnetting to the port. |
| 17:04.11 | Nugget | telnet is eeeeeeevil! |
| 17:04.26 | viperdude | ok |
| 17:04.36 | meshuga | thats a perfectly acceptable way of testing connectivity. |
| 17:04.52 | *** join/#asterisk S4w (n=saw@adsl-3-138-152.mia.bellsouth.net) |
| 17:04.53 | meshuga | JINDAL: my old employer use to run asterisk on sgi altix hardware. |
| 17:05.01 | meshuga | well, they probably still do. |
| 17:05.17 | *** join/#asterisk shido6 (n=shido6@d221-68-200.commercial.cgocable.net) |
| 17:05.28 | S4w | hey guys, with an x100p can I pickup the call waiting and transfer the first call and the second call to different sip Phones? |
| 17:05.33 | LokeshIndian | JINDAL: On digium website they mentioned about compatibel hardware with zaptel cards..better look there |
| 17:06.07 | JINDAL | oks |
| 17:06.09 | meshuga | LokeshIndian he never said he was using t1 boards. |
| 17:06.15 | meshuga | JINDAL what are you using for transport? |
| 17:07.13 | LokeshIndian | i bought wrong server to work with TE405P card |
| 17:07.23 | meshuga | te405p isnt going to get close to 200 calls. |
| 17:07.25 | JINDAL | gigbit ethernet inside the call centre... not decieded on anything else |
| 17:07.47 | meshuga | JINDAL heh, that doesnt matter. |
| 17:07.55 | meshuga | you could run 10mbit switched and be fine. |
| 17:08.03 | meshuga | this is voice, not streaming HD video. :) |
| 17:08.22 | meshuga | JINDAL you need to decide what your transport is |
| 17:08.30 | JINDAL | hehe!! |
| 17:08.32 | meshuga | what you plan onsupporting 200 i assume PSTN inbound calls on |
| 17:08.38 | meshuga | that matters more then the pc ever will. |
| 17:08.38 | JINDAL | GSM/SIP |
| 17:08.42 | S4w | is there any way to disable callwaiting on x100p FXO so that the second caller doesnt hear and endless ringing? |
| 17:08.57 | meshuga | so gsm is magically going to appear on the asterisk box? no conversion process needed? |
| 17:09.06 | meshuga | no sip provider with hordes of internet? |
| 17:09.09 | *** join/#asterisk feld_ (n=feld@12.148.212.157) |
| 17:09.18 | meshuga | i would be scared to provide 200 sip calls on voip |
| 17:09.22 | [Airwolf] | dlynes_home, have you ever worked with Diva isdn cards ? |
| 17:09.27 | meshuga | S4w: did you try callwaiting=no? |
| 17:09.31 | meshuga | in the zaptel.conf. |
| 17:09.38 | meshuga | or zapata.conf, i dont remember |
| 17:10.08 | S4w | meshuga: it makes the callwaiting sound on the phone anyways and the caller still hears the reinging and no answr ofcourse |
| 17:10.09 | JINDAL | ya am looking towards T-1/E-1 digital lines or subscribe to voip providers whichever be more economical |
| 17:11.08 | meshuga | JINDAL so, what hardware are you going to use? even a te405p full of t1s isnt going to get close to 200 concurrent |
| 17:11.09 | meshuga | heh |
| 17:11.11 | S4w | meshuga: if asterisk would send a busy tone I would be happy, is there any way of doing this? |
| 17:11.14 | meshuga | its not about economical |
| 17:11.23 | meshuga | its about quality and reliably. |
| 17:11.26 | meshuga | reliability. |
| 17:11.30 | meshuga | S4w: yes. but i dotn know how. |
| 17:12.05 | meshuga | JINDAL: i think i'd worry about your inbound far before the servers |
| 17:12.14 | meshuga | and how the hell youre going to bring in 200 calls concurrently |
| 17:12.14 | JINDAL | ya can go more than one te405p ......... dats wat i need to deciede abt what and how much do i need |
| 17:12.20 | meshuga | JINDAL you can? |
| 17:12.40 | meshuga | you're going to put 4 in a pci |
| 17:12.44 | meshuga | and then buy 16 t1s? |
| 17:12.46 | meshuga | and not a t3? |
| 17:13.16 | meshuga | not counting i've never seen a te405p work completely full reliably |
| 17:13.20 | meshuga | but maybe thats just me. |
| 17:13.28 | DagoBlok | what is t1/t3 actually ? |
| 17:13.28 | meshuga | much less multiples of them :P |
| 17:13.36 | LokeshIndian | In India I guess there is no T3 |
| 17:13.51 | JINDAL | hmm i may go for t3 if its availiable |
| 17:14.01 | LokeshIndian | if JINDAL is setting up this in India |
| 17:14.20 | meshuga | DagoBlok t1 is 23 voice channels, 1 data. or its 1.544mbit data, typically a PRI. t3 is 45mbit, dunno how many voice channels |
| 17:14.21 | JINDAL | yup in india |
| 17:14.26 | meshuga | JINDAL: usign what hardware ? |
| 17:14.34 | DagoBlok | oh ok |
| 17:14.53 | DagoBlok | some sort of internet connection then |
| 17:14.58 | meshuga | DagoBlok: asterisk uses t1s for voice, its essentially 23 phonelines terminating, but you can assign multiple dids |
| 17:15.01 | meshuga | no. |
| 17:15.02 | JINDAL | am looking for all the options presently....... |
| 17:15.17 | DagoBlok | ahh |
| 17:15.25 | meshuga | http://en.wikipedia.org/wiki/Digital_Signal_1 |
| 17:15.26 | DagoBlok | pstn |
| 17:15.35 | meshuga | not necessarily, but usually. |
| 17:15.47 | meshuga | to get pstn termination reliability 90% of pbxes use t1s |
| 17:15.50 | meshuga | hell i'd say 99% |
| 17:15.57 | JINDAL | most probably i hav to go on T-1s and then scale up |
| 17:16.01 | DagoBlok | they use it in europe too? |
| 17:16.01 | meshuga | its the traditional way of sharing multiple lines |
| 17:16.09 | blitzrage | europe uses E1 |
| 17:16.11 | DagoBlok | never heard of it before |
| 17:16.12 | meshuga | DagoBlok : e1 in europe |
| 17:16.12 | DagoBlok | oh ok |
| 17:16.14 | meshuga | 2mbit, same idea. |
| 17:16.18 | blitzrage | which has more channels than T1s |
| 17:16.20 | meshuga | its fairly old |
| 17:16.29 | meshuga | like 20 |
| 17:16.30 | blitzrage | 30B+2D if I remember correctly |
| 17:16.32 | meshuga | 20+ years |
| 17:16.33 | *** join/#asterisk cekc (n=cekc@rrcs-24-199-36-210.west.biz.rr.com) |
| 17:16.40 | *** join/#asterisk spatulamaan (n=ggilmore@207.188.8.252) |
| 17:16.55 | meshuga | JINDAL: good luck on getting a te405p to handle multiple t1s, and then getting multiple te405ps working in one machine. |
| 17:17.05 | meshuga | you got many hours of pain and anguish. |
| 17:17.21 | bancus | why the hell can't these sip providers list rates on their website? |
| 17:17.38 | meshuga | who doesnt? |
| 17:17.44 | meshuga | all the us ones i use do |
| 17:17.51 | bancus | who do you use? |
| 17:17.52 | meshuga | so, anyone use voipjet/DISA much? |
| 17:17.54 | JINDAL | should i use multiple te405ps on one machine......... i think its better to go for multiple servers |
| 17:17.58 | blitzrage | if they don't list prices, don't use them |
| 17:18.02 | meshuga | if so, please take a look at this |
| 17:18.03 | meshuga | http://pastebin.com/767705 |
| 17:18.08 | bancus | blitzrage: that's about what I think |
| 17:18.17 | bancus | the few I've found that do list prices are kinda expensive |
| 17:18.24 | meshuga | JINDAL: thats assuming you can get a te405p to work reliably singly... |
| 17:18.35 | bancus | My company uses Broadvoice at $35/mo, which is cheaper than anything I've seen listed so far. |
| 17:18.47 | meshuga | what? |
| 17:18.49 | meshuga | www.axvoice.com |
| 17:18.51 | meshuga | www.telasip.com |
| 17:18.52 | bancus | But I can't recommend them to the company I'm building this PBX for, because of intermittant auth issues. |
| 17:18.59 | meshuga | $35/mo is pretty expensive. |
| 17:19.14 | bancus | For business? |
| 17:19.18 | meshuga | unless youre going legit and not using residental plans for business :P |
| 17:19.22 | bancus | Hahaha. |
| 17:19.50 | meshuga | bancus i dont recommend voip for any companeis i do buildouts for. |
| 17:19.52 | bancus | meshuga: you use both of those? |
| 17:19.59 | meshuga | except when interconnecting offices |
| 17:20.00 | meshuga | yes. |
| 17:20.01 | bancus | meshuga: Working on a budget here. |
| 17:20.08 | meshuga | i use voipjet, freedigits, fwd, etc |
| 17:20.25 | meshuga | bancus: hopefully the business dont rely on phone service then. |
| 17:20.32 | bancus | Voip's been working fine for my company, except when the LA proxy stops authenticating us. |
| 17:20.41 | meshuga | wow you're lucky |
| 17:20.48 | meshuga | you shoulda been a broadvoice cust last year |
| 17:20.52 | bancus | I was. |
| 17:21.01 | meshuga | during gblx issues? |
| 17:21.08 | meshuga | when they decided not to pay their bill for 2 years? |
| 17:21.09 | bancus | Although we didn't use it much until August/September. |
| 17:21.15 | bancus | :o |
| 17:21.22 | meshuga | and they still aren't paying their level3 bill rumor has it? |
| 17:21.30 | bancus | urgh |
| 17:21.51 | meshuga | i've used voip for nearly 8 years |
| 17:22.14 | meshuga | and i still can't recommend it to companies for PSTN termination. |
| 17:22.26 | bancus | My only problem with broadvoice so far is that occaisionally our local proxy will refuse our auth for about four hours and I'll have to switch to a proxy on the other side of the country, which introduces a bit too much lag. |
| 17:22.53 | bancus | And they've only got the one west coast proxy. |
| 17:22.55 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 17:23.01 | meshuga | not bad |
| 17:23.17 | meshuga | i cant say i've had that much luck with BV. |
| 17:23.35 | bancus | So you really recommend against it, then? |
| 17:23.35 | meshuga | especially when it took me almost 2 months to release a block of 30 dids from them |
| 17:23.37 | JINDAL | meshuga,what kinda issue will i face wil te405p as i suppose them to completely compliant wid asterisk as they are made by digium |
| 17:23.53 | meshuga | JINDAL: good luck. |
| 17:23.55 | bancus | They're looking at getting three lines in. |
| 17:24.02 | meshuga | bancus: use copper. |
| 17:24.04 | meshuga | 3 lines aint shit. |
| 17:24.15 | meshuga | i use copper for inbound on small setups like that |
| 17:24.17 | bancus | Hoping to save them money. |
| 17:24.18 | meshuga | and voip outbound for the LD |
| 17:24.20 | JINDAL | :) |
| 17:24.21 | meshuga | heh |
| 17:24.29 | bancus | But then I have to get a special card to have it go into the PBX? |
| 17:24.32 | meshuga | putting in an ip pbx isnt saving enough? |
| 17:24.41 | meshuga | yes, you need something with FXO ports |
| 17:24.46 | meshuga | weather its a pci card or a breakout box |
| 17:24.48 | bancus | How much do those run? |
| 17:24.57 | meshuga | x100p is $15 from ebay. |
| 17:25.00 | meshuga | for 1 line. |
| 17:25.06 | bancus | Well, they don't have a PBX as it stands. |
| 17:25.09 | bancus | They're about to expand. |
| 17:25.18 | bancus | Without me, they'd probably be fine just having a bunch of different phones. |
| 17:26.02 | meshuga | heh well, if thats a case, let them go that route |
| 17:26.03 | meshuga | if you are comparing against that, you will always lose. |
| 17:26.03 | bancus | Heh. |
| 17:26.03 | meshuga | asterisk outperforms any centrex or key system solution easily |
| 17:26.04 | meshuga | but nitpicking on cost will always make you look bad. |
| 17:26.13 | bancus | Oh, they're not nitpicking on cost. |
| 17:26.17 | meshuga | outperforms/more cost efficient/etc |
| 17:26.29 | bancus | As it stands, the system will probably come out well under-budget. |
| 17:26.35 | bancus | Even if I have to get some FXO ports. |
| 17:26.44 | meshuga | bancus well, roughyl a business line should run about $25/mo, so you're look at, say, $120/mo for telecom stuff. |
| 17:26.57 | meshuga | which woould be the exact same they would pay for junch a bunch of phones |
| 17:27.01 | meshuga | with alot of added functionality. |
| 17:27.13 | bancus | Yeah. |
| 17:27.20 | meshuga | i try to convert companies to pure voip ever. |
| 17:27.20 | bancus | There's the added expense of the FXO card. |
| 17:27.24 | meshuga | heh |
| 17:27.32 | bancus | I wasn't sure how much they'd run. |
| 17:27.43 | meshuga | i dont consider $45 an added expense, but maybe thats me :) |
| 17:27.44 | bancus | And SIP's been working reasonably well for my company. |
| 17:27.45 | *** join/#asterisk MatsK (i=MatsK@83.233.97.229) |
| 17:27.50 | meshuga | youre lucky. |
| 17:27.52 | bancus | $45 is for which card? |
| 17:27.53 | meshuga | thats all it is. |
| 17:28.01 | meshuga | as i stated previously, x100p is $15 each |
| 17:28.06 | meshuga | therefore 3x$15 == 45$. |
| 17:28.15 | bancus | I doubt the box could fit three PCI cards. |
| 17:28.32 | meshuga | well, then spend $200 on the sangoma expandable ones |
| 17:28.41 | meshuga | er i mean $500 |
| 17:28.51 | meshuga | (i've had alot better luck with sangomas then digium hardware) |
| 17:28.55 | meshuga | smoke break brb |
| 17:29.31 | pdavid | is a hardware timer required for meetme & moh? |
| 17:30.56 | DagoBlok | i heard the sipura spa-3000 is excellent |
| 17:31.19 | DagoBlok | much better than a x100p |
| 17:31.51 | *** join/#asterisk trelane_ (n=trelane@pdpc/supporter/sustaining/trelane) |
| 17:32.26 | pdavid | what options should i look at if i dont have hardware timer, and no uhci-usb |
| 17:32.28 | pdavid | :( |
| 17:32.56 | dlynes_home | DagoBlok: x100p doesn't have fxs, and it's also a total piece of crap |
| 17:33.03 | dlynes_home | DagoBlok: so it wouldn't take much to be better than it |
| 17:33.11 | DagoBlok | lol, ok :) |
| 17:33.17 | DagoBlok | never tried the x100p :) |
| 17:33.21 | MACscr | What do you think is better for remote agents, ATA's or Voip Phones? |
| 17:33.27 | dlynes_home | but yeah, the spa-3000 is ok |
| 17:34.06 | bancus | does digium not sell the x100p anymore? I don't see it on their site |
| 17:34.13 | *** join/#asterisk angler (n=angler@pdpc/sponsor/digium/angler) |
| 17:34.19 | meshuga | pdavid no you can use the module ztdummy. |
| 17:34.33 | pdavid | meshuga: i am having a hard time compiling the darn module, though |
| 17:34.43 | bancus | aha, apparently it was replaced by the tdm400p |
| 17:34.48 | meshuga | uh what? |
| 17:34.49 | bancus | anyone have any experience with that? |
| 17:34.52 | meshuga | yes. |
| 17:35.06 | meshuga | you only use ztdummy if you have no zaptel hardware. |
| 17:35.23 | pdavid | right, but i cannot get ztdummy to compile for some reason |
| 17:35.28 | meshuga | if you use the hardware, you'll use hardware timer and don't need to use the RTC, which is what ztdummy does |
| 17:35.36 | pdavid | meshuga: right, got it so far |
| 17:35.48 | pdavid | meshuga: i am trying to compile the module from svn snapshots |
| 17:35.53 | meshuga | uhm, if you can't get ztdummy to compile, you got system/distribution/os issues :) |
| 17:35.54 | meshuga | why |
| 17:35.56 | *** join/#asterisk cytrak (n=btorch@208.63.19.179) |
| 17:35.57 | meshuga | use releases. |
| 17:35.59 | dlynes_home | bancus: lotsa peeps on here using the tdm400 |
| 17:36.14 | meshuga | bancus: buy a x100p from ebay |
| 17:36.18 | meshuga | i've bought tons of clones |
| 17:36.22 | meshuga | never had a problem with one |
| 17:36.48 | bancus | As I said, I don't think the system has 3 open PCI ports. |
| 17:36.51 | cytrak | does anyone know how I can connect my asterisk with a good voip provider? I understand I have to create a peer register |
| 17:36.58 | bancus | I'd prefer a card that could supply all three in one. |
| 17:37.12 | meshuga | then you're going to spend $500+ |
| 17:37.15 | bancus | :/ |
| 17:37.46 | meshuga | buying a el-cheapo computer (since x100ps are less picky about dedicated irqs then t1 boards) |
| 17:37.54 | meshuga | would be more cost efficient |
| 17:37.56 | cytrak | any suggestions on a good voip provider that can give me say a block of voip extensions that can be redirected to my asterisk box.. I guess i'm looking for a getaway provider ? |
| 17:37.57 | meshuga | but less of a path to ugprade |
| 17:38.23 | meshuga | cytrak: sounds like you need to do some reading. http://www.nerdvittles.com/ is suggested. |
| 17:38.51 | meshuga | http://nerdvittles.com/index.php?p=130 |
| 17:38.53 | cytrak | please point me where to do the reading |
| 17:38.54 | meshuga | this is one of my favorite pages |
| 17:39.05 | dlynes_home | meshuga: i've found the opposite...the x100p's will share an irq, but if the irq is shared, after a period of time, it ceases to register with the driver |
| 17:39.10 | *** join/#asterisk r_evolution (i=_evoluti@208.251.203.246) |
| 17:39.18 | meshuga | cytrak: the voip provider reviews and config tips |
| 17:39.46 | dlynes_home | meshuga: with my four port pri card, it was sharing an irq, and it never ceased to register with the driver |
| 17:39.47 | meshuga | dlynes_home: I guess it matters whats sharing it. but i can definitely say i've had much less issues with irq sharing on x100p's then tdm boards |
| 17:40.04 | dlynes_home | meshuga: however, both of them would drop calls periodically because of the interrupt sharing, too |
| 17:40.10 | meshuga | i havent used a t1 board in a year, so maybe they got better |
| 17:40.28 | dlynes_home | meshuga: our t1 card is about 3 or 4 years old |
| 17:40.29 | meshuga | it all depends how the mobo is designed |
| 17:40.35 | meshuga | and how irq steering on the mobo works |
| 17:40.42 | meshuga | and if ACPI is trying to take it over, etc |
| 17:40.44 | dlynes_home | ah |
| 17:40.51 | dlynes_home | you mean APIC? |
| 17:41.03 | *** join/#asterisk moprilo (n=mop@201.198.78.23) |
| 17:41.06 | meshuga | i've had supermicro/tyan boards fail because they implemented their own technology |
| 17:41.15 | CunningPike | Quick hints question: would this work? exten=> 1234,hint,SIP/1234&SIP/2345&SIP/3456? |
| 17:41.17 | meshuga | yea, which is controlled by APCI still it hought |
| 17:41.21 | cytrak | meshuga, cool thanks |
| 17:41.30 | *** join/#asterisk Blake0PS (n=blake@c-66-41-195-142.hsd1.mn.comcast.net) |
| 17:41.31 | *** join/#asterisk ramo (n=ramo@59.92.133.168) |
| 17:41.34 | meshuga | cuz if you dont load ACPI modules, APIC doesn't load either. |
| 17:41.50 | meshuga | CunningPike should with a dial line. i dunno how your hint macro is setup though |
| 17:41.53 | r_evolution | HAH! |
| 17:41.54 | r_evolution | SCORE! |
| 17:41.55 | dlynes_home | APIC == advanced programmable interrupt controller; ACPI == something to do with power management |
| 17:41.59 | meshuga | i'd replace hint with dial |
| 17:42.04 | dlynes_home | CunningPike: yes |
| 17:42.14 | CunningPike | dlynes_home: Thanks :D |
| 17:42.17 | r_evolution | hey dlynes... remember how i was in her ethe other day saying about how in 1.2.8 asterisk crashed out when you tried to park a call? |
| 17:42.22 | dlynes_home | CunningPike: i use something almost exactly like that |
| 17:42.24 | r_evolution | well apparently it got fixed in 1.2.9.1 |
| 17:42.25 | r_evolution | SCORE! |
| 17:42.40 | CunningPike | dlynes_home: Perfect. And how come you're at home? |
| 17:42.44 | dlynes_home | r_evolution: heh...that's why you should always use the latest version before reporting a bug |
| 17:42.50 | dlynes_home | CunningPike: just on my way into the office |
| 17:42.52 | r_evolution | well |
| 17:42.57 | r_evolution | 1.2.8 WAS the latest version at the time |
| 17:42.57 | moprilo | hi, i just upgraded from 1.0.9 to 1.2.9, and when i transfer a zapincomming call from a sip-phone to a zap-fxs-phone it answers and automaticly hangups any idea_]? |
| 17:43.00 | meshuga | dlynes_home: stands for advanced configuration and power interface. replaces pnp on isa |
| 17:43.03 | CunningPike | dlynes_home: Well, make it snappy ;) |
| 17:43.04 | dlynes_home | CunningPike: I was at a job site until 11pm or so last night |
| 17:43.04 | r_evolution | the day after i bitched about the issue in here for a good hour |
| 17:43.04 | meshuga | http://en.wikipedia.org/wiki/ACPI |
| 17:43.09 | r_evolution | they released 1.2.9.1 |
| 17:43.13 | dlynes_home | CunningPike: all the way the hell out in surrey, too |
| 17:43.15 | meshuga | power management is common, yes, but it also does device config and all that kazz |
| 17:43.17 | meshuga | jazz |
| 17:43.22 | CunningPike | dlynes_home: Surrey == hell |
| 17:43.23 | meshuga | i've never seen APIC work without ACPI. |
| 17:43.30 | meshuga | so i assumed APIC was a subset of ACPI. |
| 17:43.42 | dlynes_home | meshuga: you need to enable SMP in the kernel before you can enable APIC |
| 17:43.55 | dlynes_home | meshuga: even if you're on a uniprocessor system |
| 17:43.58 | meshuga | uhm what? |
| 17:44.01 | meshuga | then i never use APIC :) |
| 17:44.02 | *** join/#asterisk _DAW (n=bob@adsl-150-59-108.msy.bellsouth.net) |
| 17:44.04 | meshuga | i only use ACPI |
| 17:44.11 | meshuga | which handles the pci steering and timings on most mobos |
| 17:44.15 | dlynes_home | yeah...ACPI is the new improved replacement for APM |
| 17:44.17 | meshuga | at least that follow the intel design. |
| 17:44.29 | meshuga | dlynes_home: as the wikipedia link stated, its much more then that. |
| 17:44.41 | dlynes_home | ~wiki acpi |
| 17:44.50 | dlynes_home | ~wiki apic |
| 17:45.00 | SplasPood | Does anyone here have the gizmo client talking to an asterisk box? |
| 17:45.01 | meshuga | but i never enable SMP |
| 17:45.05 | meshuga | and i see APIC messages |
| 17:45.08 | meshuga | so i dunno if thats right |
| 17:45.37 | dlynes_home | anyways |
| 17:45.41 | meshuga | damn, they are beyond dmesg |
| 17:45.43 | dlynes_home | i need to head into the office |
| 17:45.48 | dlynes_home | dmesg | grep APIC |
| 17:46.01 | meshuga | heh as i said, its beyond what is in dmesg |
| 17:46.02 | meshuga | so i cant |
| 17:46.02 | dlynes_home | that'll tell you if you have APIC enabled or not |
| 17:46.06 | meshuga | no it wont. |
| 17:46.15 | meshuga | because dmesg has all kernel messages, and they've scrolled off :) |
| 17:46.24 | meshuga | thats what i mean when i say 'its beyond dmesg' |
| 17:46.29 | meshuga | since its a finite buffer |
| 17:46.33 | dlynes_home | but if you grep it |
| 17:46.39 | dlynes_home | you can see it |
| 17:46.45 | dlynes_home | dmesg is a command at the linux prompt |
| 17:46.56 | meshuga | no. |
| 17:47.09 | dlynes_home | meshuga: then you're using some really weird distro |
| 17:47.13 | meshuga | dude |
| 17:47.15 | dlynes_home | every distro i've used has that command |
| 17:47.15 | meshuga | you dont understand. |
| 17:47.22 | meshuga | of course i have dmesg |
| 17:47.28 | meshuga | as i stated, dmesg has a finite buffer |
| 17:47.37 | meshuga | and those startup messages have scrolled out of the buffer |
| 17:47.40 | meshuga | therefore, i can't grep it. |
| 17:47.47 | meshuga | its common when your machine is up for more then a few days |
| 17:47.52 | meshuga | and you use alot of apps that report to the kernel |
| 17:47.55 | meshuga | say, smb. |
| 17:48.01 | meshuga | since smb errors so much |
| 17:48.01 | Blake0PS | One of my Zap channels has been failing and I'm wondering which error is the culprit. "WARNING[811]: zt hook failed: Device or resource busy" or "WARNING[4394]: CallerID returned with error on channel 'Zap/3-1'" |
| 17:48.49 | meshuga | dmesg is a great utility |
| 17:49.01 | *** join/#asterisk oej (n=oej@apollo.webway.se) |
| 17:49.02 | meshuga | but you can rely on it accurately to report hardware on a machine that isnt freshly booted up. |
| 17:49.06 | meshuga | cant^ |
| 17:49.16 | meshuga | thats why you should do *.kernel in syslog |
| 17:49.39 | *** part/#asterisk liran_ (n=Coll@212.199.177.203.static.012.net.il) |
| 17:49.47 | TheCops | [TK]D-Fender? |
| 17:49.50 | TheCops | There ? |
| 17:49.58 | *** part/#asterisk ramo (n=ramo@59.92.133.168) |
| 17:50.00 | meshuga | Use a buffer of size bufsize to query the kernel ring buffer. This is 16392 by default. |
| 17:50.01 | meshuga | <PROTECTED> |
| 17:50.01 | meshuga | <PROTECTED> |
| 17:50.02 | meshuga | <PROTECTED> |
| 17:50.18 | meshuga | so you only get 16392 characters. |
| 17:51.14 | MACscr | do you guys like packet8? |
| 17:51.27 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 17:51.54 | coppice | sad sad company packet8. it has lost its investors a large fortune |
| 17:52.06 | mishehu | al teshageA et hasechel |
| 17:52.07 | mishehu | heh |
| 17:52.30 | meshuga | coppice: welcome to the wonderful world of telecom. |
| 17:52.30 | *** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it) |
| 17:52.38 | *** join/#asterisk schirpich (n=dvs@ip21.farheap.net) |
| 17:52.40 | MACscr | any better providers that offer unlimited plans and work with asterisk? |
| 17:52.52 | MACscr | i have 3 lines with vonage |
| 17:52.59 | MACscr | but obviously i cant use them =( |
| 17:53.11 | schirpich | Is there a different context for the "first in line" hold music versus all the other hold music contexts? |
| 17:53.43 | SplasPood | hrm, gizmo doesn't seem to wanna talk to asterisk at all |
| 17:53.57 | meshuga | MACscr: theres quite a few, but better is too subjective |
| 17:54.16 | meshuga | MACscr: axvoice. telasip. broadvoice. dialpad. just to name a few |
| 17:54.39 | meshuga | schirpich: not by default |
| 17:54.47 | MACscr | thanks mesh |
| 17:54.50 | mishehu | meshuga: like I said, al teshageA et hasechel ;-) |
| 17:54.53 | meshuga | you should be able to script it though, since queue participation is a variable |
| 17:55.09 | meshuga | mishehu: i dont speak yiddish/hebrew/whatever |
| 17:55.19 | meshuga | so you can stop trying to speak to me in your native tongue :) |
| 17:55.25 | mishehu | meshuga: hebrew. and your nick is hebrew so I thought I'd try it out |
| 17:55.37 | meshuga | yea i pulled it out of a book 10+ years ago :) |
| 17:55.45 | mishehu | means you're nuts. |
| 17:55.57 | schirpich | meshuga: say there's 10 people in a queue. callers 2-10 can hear hold music. caller 1 hears nothign but silence. Where is this specified? |
| 17:56.00 | meshuga | senseless is my preferred translation |
| 17:56.03 | mishehu | not totally crazy, but nuts. |
| 17:56.11 | schirpich | if its not by default? |
| 17:56.14 | meshuga | schirpich: sounds like a configuration issue |
| 17:56.21 | meshuga | thats not default configured. |
| 17:56.24 | RF_MIA | queues.conf |
| 17:56.26 | meshuga | but i havent looked at queue stuff since 1.0.2 |
| 17:56.29 | meshuga | so maybe i'm wrong |
| 17:56.37 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 17:56.37 | *** part/#asterisk KranZ (n=user@sme.bestline.net) |
| 17:56.39 | *** join/#asterisk KranZ (n=user@sme.bestline.net) |
| 17:56.52 | meshuga | mishehu: what exactly is the difference in hebrew language? |
| 17:56.58 | meshuga | between nuts and totally crazy |
| 17:57.05 | mishehu | meshuga: meturaf is totally crazy. |
| 17:57.19 | meshuga | <mishehu> meshuga: meturaf is totally crazy. |
| 17:57.20 | meshuga | er |
| 17:57.51 | mishehu | itchy mouse finger eh/ |
| 17:57.53 | mishehu | heh |
| 17:58.27 | tanvalmg | Hi, does anybody know if the Digium TDM400P is compatible with IBM xSeries 346? |
| 17:58.54 | meshuga | i'm surprised someone hasnt made a hardware compatibility page on a wiki somewhere. |
| 17:59.05 | mishehu | tanvalmg: hard to say, in all honesty. did you do a google search or look on digium's compat list? |
| 17:59.16 | meshuga | digium has a valid one? |
| 17:59.17 | *** join/#asterisk saftsack (n=saftsack@p54A7D890.dip.t-dialin.net) |
| 17:59.22 | mishehu | I've been unlucky with the digium cards causing PCI PERR# in the past. |
| 17:59.46 | mishehu | welp. lunch time |
| 17:59.48 | meshuga | mishehu yea, thats crappy pci steering |
| 17:59.52 | tanvalmg | The Digium site says it is partially compatible with xSeries 345, we xSeries 346 |
| 18:00.08 | meshuga | i use sangomas myself |
| 18:00.13 | mishehu | time is an illusion, lunchtime doubly so. |
| 18:00.14 | tanvalmg | we have IBM xSeries 346 |
| 18:00.21 | meshuga | i've seen too many tdm400p's die by static |
| 18:00.39 | meshuga | man wtf |
| 18:00.42 | mishehu | I do not recommend any of the tigerjet chipset digium cards |
| 18:00.44 | meshuga | http://pastebin.com/767705 |
| 18:00.50 | meshuga | i cannot get this to work |
| 18:00.54 | mishehu | I've had too many problems with them. |
| 18:00.57 | meshuga | mishehu dude, can you take a look at this? |
| 18:02.17 | mishehu | meshuga: you put up the log but not the extensions config |
| 18:02.44 | mishehu | anyway, I'll be back in an hour. must go out to eat and head over to a client's office |
| 18:03.38 | MACscr | has anyone ever tried transfering their vonage number to another provider? |
| 18:03.44 | meshuga | mishehu heh its just amp |
| 18:04.00 | meshuga | mishehu they're setup the exact same way. |
| 18:06.33 | moprilo | asterisk is not recognizing # key on my zap channel phones (FXS ports), why can this be? |
| 18:06.35 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 18:06.42 | moprilo | i just upgraded to 1.2.9 |
| 18:07.14 | *** join/#asterisk insomni (n=insomni@x1-6-00-e0-18-6f-34-ad.k455.webspeed.dk) |
| 18:08.50 | moprilo | jmm.. |
| 18:09.02 | moprilo | out to lunch? jeje.. |
| 18:09.36 | moprilo | it only happens when i receive a call |
| 18:09.59 | moprilo | and it doesn't do it on sip phones either |
| 18:11.02 | RF_MIA | anybody have any experience with res_snmp ? |
| 18:13.03 | MACscr | Small phone network with all extensions at different locations. Should i go with IAX2 or SIP? |
| 18:13.52 | meshuga | iax2 is always better |
| 18:13.56 | meshuga | but less ip phones are IAX2 |
| 18:14.04 | meshuga | so you usually have to go SIP for 'the last mile' |
| 18:14.33 | MACscr | got any recommendations for iax2 phones that are entry level? |
| 18:14.54 | *** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it) |
| 18:15.00 | moprilo | jaja, my fault, worked.. |
| 18:15.08 | *** part/#asterisk moprilo (n=mop@201.198.78.23) |
| 18:15.20 | meshuga | maxscr: i dont even know of any |
| 18:15.27 | meshuga | i prefer polycom's for cheap ip phones |
| 18:15.30 | meshuga | and ciscos for mid grade |
| 18:15.33 | _Sam-- | there are iax phones |
| 18:15.35 | *** join/#asterisk philippel (n=p_lindhe@c-24-19-186-72.hsd1.wa.comcast.net) |
| 18:15.47 | MACscr | im familiar with polycom in general |
| 18:15.47 | meshuga | the 2 iax one i used last year were garbage |
| 18:15.53 | MACscr | i worked in telecom for a couple years |
| 18:16.04 | meshuga | nice i'd go with those for handsets |
| 18:16.08 | meshuga | did you provision by hand? |
| 18:16.15 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
| 18:16.21 | meshuga | be sure to upgrade them all to the latest firmwares |
| 18:16.34 | meshuga | cuz polycoms changed their configuration in later firmware revisions |
| 18:16.48 | meshuga | and i did a rollout once in 3 steps and half my shit didnt work and i had no idea |
| 18:17.00 | *** part/#asterisk RF_MIA (n=mw1@ip67-93-229-222.z229-93-67.customer.algx.net) |
| 18:17.08 | meshuga | why until i realized that, updated my tftp service to push firmware and all was well. |
| 18:17.15 | *** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net) |
| 18:17.18 | pdavid | grr |
| 18:17.28 | pdavid | could anyone lend me a hand figuring out why i cannot seem to build the ztdummy module? |
| 18:17.58 | MACscr | looks pretty cheap, but not bad as far as what it supports as an ATA |
| 18:17.59 | MACscr | http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=5819261173&category=61839 |
| 18:18.02 | *** join/#asterisk chapeaurouge (n=chapeaur@user-85-201-82-146.tvcablenet.be) |
| 18:18.48 | *** part/#asterisk JINDAL (n=root@220.226.36.2) |
| 18:19.00 | *** join/#asterisk brc_ (n=brc_@pdpc/supporter/basic/brc) |
| 18:19.25 | brc_ | hey hey hey |
| 18:19.29 | brc_ | guess who |
| 18:20.00 | pdavid | make runs clean without any errors |
| 18:20.19 | pdavid | so does make install |
| 18:20.24 | pdavid | but a modprobe ztdummy does nada |
| 18:20.36 | pdavid | FATAL: Module ztdummy not found. |
| 18:20.38 | pdavid | FATAL: Error running install command for ztdummy |
| 18:21.05 | C4T3l | pdavid: you running udev? |
| 18:21.21 | pdavid | yes, and did make install-udev |
| 18:21.21 | *** join/#asterisk paolob (n=donpaolo@pri-214-b7.codetel.net.do) |
| 18:21.23 | MatsK | make linux26 |
| 18:21.25 | pdavid | and rebooted |
| 18:21.31 | pdavid | same issue with make linux26 as well |
| 18:21.51 | CunningPike | pdavid: Make sure the module exists in the folder corresponding to uname -a |
| 18:21.54 | *** join/#asterisk funxion (n=nunya@63.214.236.169) |
| 18:22.07 | C4T3l | uname -r? |
| 18:22.14 | CunningPike | C4T3l: That too |
| 18:22.17 | dlynes_office | CunningPike: damned cops busted me on knight street :((( |
| 18:22.18 | C4T3l | hehe |
| 18:22.19 | paolob | Hi guys! When trying to make a voip call with wengo, asterisk tells me many times: " WARNING[8120]: chan_sip.c:2542 sip_write: Asked to transmit frame type 4, while native formats is 256 (read/write = 256/256)". What does it mean? |
| 18:22.38 | pdavid | CunningPike: as in: /lib/modues/2.6.12-10-686? |
| 18:23.27 | CunningPike | pdavid: Yes - does that match uname -r? |
| 18:23.33 | pdavid | sure does |
| 18:23.34 | MACscr | mesh: which polycom do you prefer |
| 18:24.01 | Bullseye_Network | Im using a dialer for agents and im getting calls on hold when there are agents available in the queue. What are some good ways to speed up asterisk? |
| 18:24.20 | MatsK | pdavid: no errors while compiling ? |
| 18:24.31 | pdavid | Matsk: no, no errors while compiling |
| 18:24.35 | CunningPike | pdavid: What kernel? |
| 18:24.46 | pdavid | 2.6.12-10-686 |
| 18:24.46 | CunningPike | pdavid: nm - I can actually read lol |
| 18:24.50 | pdavid | lol |
| 18:24.57 | pdavid | maybe they are going to the wrong place |
| 18:25.02 | pdavid | this is an ubuntu box |
| 18:25.09 | Bullseye_Network | Can I change the nice levels? |
| 18:25.12 | pdavid | maybe im missing a symlink somewhere? |
| 18:25.23 | CunningPike | pdavid: That would be my suspicion - I've had that happen on RHEL |
| 18:25.35 | pdavid | well, where *should* they go, then? |
| 18:25.52 | CunningPike | pdavid: In the correct folder, of course :D |
| 18:26.01 | pdavid | ahh, well that would make sense! |
| 18:26.40 | salviadud | CunningPike, are you saying red hat is red hell' |
| 18:26.43 | salviadud | ? |
| 18:26.59 | CunningPike | salviadud: Not at all - I like it |
| 18:27.31 | salviadud | i like suse better |
| 18:27.46 | salviadud | if i need a distro that does everything for me |
| 18:28.21 | salviadud | i've heard the guys from debian are like muslim fanatics |
| 18:28.55 | salviadud | they'll DOS FreeBSD servers, and stuff like that |
| 18:29.08 | dlynes_office | salviadud: you're confusing them with the guys in #perl :) |
| 18:29.29 | CunningPike | pdavid: Mine is in /lib/modules/2.6.9-34.EL/extra/, if that helps..... |
| 18:29.32 | C4T3l | salviadud: are you serious about DOS? |
| 18:29.48 | pdavid | that is where ztdummy.ko resides? |
| 18:29.55 | salviadud | no, i think it's funny to crash a server like that |
| 18:29.58 | CunningPike | pdavid: Yes |
| 18:30.05 | *** join/#asterisk TheCops (n=henri@206-248-136-187.dsl.teksavvy.com) |
| 18:30.08 | salviadud | it's basically evil |
| 18:30.15 | salviadud | bad karma and such |
| 18:30.30 | salviadud | i don't do DOS, i'm a nice guy |
| 18:31.07 | salviadud | well, regular guy, i like to prank call with .call files |
| 18:31.14 | mitcheloc | who uses dos anymore? |
| 18:31.17 | salviadud | phone terrorism is cool |
| 18:31.28 | *** join/#asterisk w32 (n=234@c-71-193-124-77.hsd1.il.comcast.net) |
| 18:31.38 | salviadud | it's even better with voipdiscount |
| 18:31.43 | salviadud | i can call the US for free |
| 18:32.04 | w32 | does anyone know of a reliable provider for origination and termination that accepts paypal ? |
| 18:32.22 | salviadud | now, if i can get the guys from #perl to help me out with a script that auto-generates call files... |
| 18:33.43 | *** join/#asterisk Blackthorn (i=blacktho@72.236.88.10) |
| 18:33.44 | *** join/#asterisk eKo1 (n=bernd@190.4.7.90) |
| 18:34.18 | eKo1 | Wow. Asterisk is moving so quickly. I just upgraded to 1.2.8 last week and now we have 1.2.9.1 |
| 18:34.33 | dlynes_office | CunningPike: so have you been busted by those damned cops on knight street yet? |
| 18:34.46 | dlynes_office | CunningPike: or do you usually take oak street? |
| 18:34.50 | CunningPike | dlynes_office: No - I'm way to careful a driver :D |
| 18:34.59 | dlynes_office | CunningPike: such a bs'er :) |
| 18:35.14 | dlynes_office | bastards busted me doing 76 on knight :((( |
| 18:35.24 | mitcheloc | eKo1: not much makes you move faster then security issues ;) |
| 18:35.30 | CunningPike | dlynes_home: When I drive, I take Granville in the morning and Knight on the way home |
| 18:35.39 | MACscr | yep, still no good iax2 phones |
| 18:35.41 | CunningPike | dlynes_office: It's a posted 50....... |
| 18:35.51 | dlynes_office | ah...lion's gate is pretty clear in the mornings? |
| 18:36.00 | CunningPike | dlynes_office: Yes, it's not bad |
| 18:36.15 | dlynes_office | so what you're telling me is you never speed? :) |
| 18:36.18 | CunningPike | dlynes_office: But I'm scooting these days |
| 18:36.27 | CunningPike | dlynes_office: 10 over - never more |
| 18:36.33 | dlynes_office | ah....that's why you never speed |
| 18:36.38 | dlynes_office | the scooter can't go that fast :) |
| 18:36.45 | CunningPike | dlynes_office: Even in the truck - 10 over |
| 18:37.00 | dlynes_office | ah |
| 18:37.11 | dlynes_office | yeah...in the van, i usually don't go more than 5 or 10 over |
| 18:37.27 | dlynes_office | but in my gti, it's a different story :) |
| 18:37.28 | CunningPike | dlynes_office: But you're right - the scooter maxes out at 65 |
| 18:37.35 | CunningPike | dlynes_office: Golf? |
| 18:37.38 | dlynes_office | yeah |
| 18:37.40 | pdavid | im pretty sure i am going to scream shortly |
| 18:37.47 | CunningPike | dlynes_office: Nice |
| 18:37.47 | *** join/#asterisk noky (n=noky@200.69.211.18) |
| 18:37.48 | noky | hi |
| 18:37.58 | dlynes_office | yeah...nice red one, too :) |
| 18:38.15 | CunningPike | pdavid: Still struggling? |
| 18:38.24 | pdavid | yep, big time |
| 18:38.37 | pdavid | cant seem to figure out where its all going haywire |
| 18:38.41 | *** join/#asterisk truz_`24 (n=truz_`24@74.129.166.232) |
| 18:38.46 | CunningPike | pdavid: Your rules-50 and permissions-50 are all OK? |
| 18:38.49 | pdavid | should probably step away from the terminal and smoke before i toss it out the window |
| 18:38.49 | noky | please i need some help, where can i found any benchmark or anything to see the performance of the application MeetMe in Asterisk? |
| 18:39.39 | pdavid | well, lemme check, and smoke. bbl |
| 18:39.41 | truz_`24 | Can asterisk determine if a call was intercepted by an operator |
| 18:40.02 | r_evolution | fuck |
| 18:40.09 | r_evolution | i like too wide a variety of music :( |
| 18:41.04 | sevard | how the hell is that sad? |
| 18:41.29 | mitcheloc | do people know about pound key?? |
| 18:41.54 | *** join/#asterisk SplasPood (n=jwb@206.252.198.101) |
| 18:41.59 | CunningPike | truz_`24: Telco operator? |
| 18:42.01 | r_evolution | it's sad because i filled up the external hard-drive i bought for music |
| 18:42.04 | r_evolution | and that annoys me |
| 18:42.08 | r_evolution | because it means i have to buy another one... |
| 18:42.12 | r_evolution | and i wanted them to all be on ONE drive |
| 18:42.16 | r_evolution | not like 4 :( |
| 18:42.21 | truz_`24 | CunningPike, yeah, and detecting answering machines and such. |
| 18:42.23 | Alric | Setup a nice NAS :) |
| 18:42.28 | CunningPike | mitcheloc: How do you mean? |
| 18:42.30 | r_evolution | someone hurry up and make a 2 Terrabyte external |
| 18:42.44 | CunningPike | truz_`24: Not sure how it could - it just knows the channel was answered |
| 18:42.52 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-215.usadatanet.com) |
| 18:43.10 | KranZ | anyone found the perfect software echo cancellation settings for a pri? |
| 18:43.37 | [TK]D-Fender | KranZ : No, but I've found a good solder one ;) |
| 18:43.47 | *** join/#asterisk jhiver (n=jhiver@LReunion-151-20-4.w193-253.abo.wanadoo.fr) |
| 18:43.47 | KranZ | u modded? |
| 18:43.51 | jhiver | hi all |
| 18:43.53 | [TK]D-Fender | KranZ : What you're asking doesn't exist :) |
| 18:44.00 | jhiver | I have a strange issue with Asterisk |
| 18:44.03 | mitcheloc | CunningPike: supposedly pk is the official linux distro from digium with asterisk in it.... |
| 18:44.11 | caloi | Hello all - anyone feel like assisting a newcomer on dial plan creation for an iax -> iax connection? |
| 18:44.12 | mitcheloc | CunningPike: i was wondering why people don't use that a hell of a lot more.... |
| 18:44.25 | [TK]D-Fender | KranZ : Echo correction needs is ALWAYS variable... and every case can have different needs. |
| 18:44.43 | [TK]D-Fender | KranZ : If it could be standardized you wouldn't be asking that question right now :) |
| 18:44.57 | jhiver | one of my customers sends me SIP traffic from a cirpak switch, and he can't hear any ringing at all |
| 18:45.07 | r_evolution | we all have strange issues with asterisk jhiver |
| 18:45.12 | r_evolution | it's a strange system |
| 18:45.18 | r_evolution | I support tin can with a string personally |
| 18:45.18 | jhiver | but when the phone is off hook the conversation goes fine |
| 18:45.19 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 18:45.20 | r_evolution | but that's just me |
| 18:45.25 | CunningPike | mitcheloc: Ah - lol. I thought you meant...... # |
| 18:45.39 | jhiver | it's just that there is no ring, which is kind of bad |
| 18:45.55 | mitcheloc | http://www.digium.com/en/products/opensourceasterisk/downloads.php |
| 18:45.57 | r_evolution | oh oh i've got it! use the r option! |
| 18:45.58 | r_evolution | haha |
| 18:45.59 | r_evolution | ;x |
| 18:46.16 | mitcheloc | look at PK on there, im downloading right now |
| 18:46.30 | KranZ | yeah, i've got a remote sip ua connected to a pbx via pri |
| 18:46.40 | KranZ | bad echo on the pstn side for cell phones |
| 18:46.46 | [TK]D-Fender | BBIAB |
| 18:46.47 | *** part/#asterisk marv[work] (n=timr@64.89.118.139) |
| 18:46.55 | jhiver | anybody else got this 'no ring issue'? |
| 18:46.55 | *** join/#asterisk marv[work] (n=timr@64.89.118.139) |
| 18:48.07 | jhiver | I have progressinband=yes in my sip.conf so my understanding is that asterisk should send some RTP? |
| 18:48.20 | jhiver | when ringing? |
| 18:48.48 | Blackthorn | Kranz are you refering to the double ring when you call form ata --> * ---> out pri? |
| 18:51.28 | KranZ | nah |
| 18:51.44 | Blackthorn | ok i just jumped into channel.. to try to ask the above statement. |
| 18:51.47 | KranZ | you can fix that with sticky183=yes on the sip ua |
| 18:52.03 | Bullseye_Network | Im using a dialer for agents and im getting calls on hold when there are agents available in the queue. What are some good ways to speed up asterisk? |
| 18:52.14 | pdavid | So, i have NO idea what I did, but there they were |
| 18:52.16 | Blackthorn | Ata to * to voice pulse, ata to * to ata... works fine. but i can't get the doule bring |
| 18:52.20 | pdavid | after a quick depmod, im in the game |
| 18:52.26 | pdavid | :( |
| 18:52.29 | pdavid | hate voodoo installs |
| 18:52.33 | meshuga | so anyone know why when I'm using DISA and i got a remote dialtone, I'm unable to dial out thru voipjet or voipbusted or anything? I never even get the 'attempting native bridge' or any progress indicatations.. |
| 18:52.48 | pdavid | cunningpike: thanks for the help, though! |
| 18:53.04 | meshuga | voipjet gives me a congested tone, and voipbuster just has me hanging |
| 18:53.12 | truz_`24 | CunningPike, dialogic can detect answering machines with a 98% accuracy... |
| 18:53.14 | meshuga | if i call from an internal sip extension, everything works fine |
| 18:53.25 | Blackthorn | Kranz: are you stating to fix the issue i metnioned I put "sticky183=yes" on the sip.conf profile for the ata? |
| 18:53.30 | truz_`24 | CunningPike, is there any such guess algorithm/plugin for asterisk? |
| 18:53.31 | meshuga | but from my DISA login it doesnt work |
| 18:53.36 | CunningPike | pdavid: Great |
| 18:54.14 | CunningPike | truz_`24: I don't believe so...... but I remember reading something either on the list or the wiki about answering machine detection.... |
| 18:54.19 | *** join/#asterisk moprilo (n=mop@201.198.78.23) |
| 18:55.01 | moprilo | is it posible i don't have my Asterisk Database installed, my call forwarding is not working.., and all the commands seem to be though |
| 18:55.45 | truz_`24 | CunningPike, can i have a link to the wiki? |
| 18:55.54 | CunningPike | ~wiki |
| 18:56.17 | CunningPike | ~thewiki |
| 18:56.18 | jbot | extra, extra, read all about it, thewiki is at http://www.voip-info.org/wiki-Asterisk |
| 18:56.31 | noky | please i need some help, where can i found any benchmark or anything to see the performance of the application MeetMe in Asterisk? |
| 18:57.08 | CunningPike | noky: Try the list archives - I have a vague recollection of a discussion there a while ago |
| 18:57.47 | CunningPike | ~thelist |
| 18:58.05 | CunningPike | ~list |
| 18:58.06 | jbot | one warez list being sent |
| 18:58.20 | CunningPike | jbot, you are a fool |
| 18:58.21 | jbot | I think you lost me on that one, CunningPike |
| 18:59.09 | Blackthorn | Kranz: Where do you plac "sticky183=yes" on the sip.conf profile for the ata? |
| 18:59.56 | truz_`24 | CunningPike, thanks |
| 19:00.22 | CunningPike | truz_`24: You're welcome |
| 19:00.44 | CunningPike | ~asterisk-users |
| 19:01.59 | CunningPike | jbot, thelist is the asterisk-users mailing list. Sign up or view archives at http://lists.digium.com/mailman/listinfo/asterisk-users |
| 19:02.01 | jbot | okay, CunningPike |
| 19:05.45 | Blackthorn | ahhh.. sigh.. i finally find somone that gives me a clue how to fix the issue... and i don't understand the answer :\ |
| 19:06.02 | Blackthorn | goggled it and found three sites in german :P |
| 19:07.52 | *** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it) |
| 19:09.52 | *** join/#asterisk squinky86 (n=squinky8@gentoo/developer/squinky86) |
| 19:10.25 | *** join/#asterisk aze_ (n=aze@ACayenne-101-1-10-77.w81-248.abo.wanadoo.fr) |
| 19:10.37 | *** join/#asterisk Samoied (n=Samoied@ip47092.static.poa.virtua.com.br) |
| 19:11.01 | KranZ | Blackthorn: google stick183 |
| 19:11.13 | KranZ | i enabled it on my linksys/sipura devices |
| 19:12.30 | Blackthorn | google only pulls up 3 websites which do talk about sipura and stiky183 but there in german. |
| 19:12.48 | Blackthorn | i even just tried to ues babblefish to translate :P didn't work. |
| 19:12.49 | MACscr | kranz: how do you like your sipura phones |
| 19:13.08 | SplasPood | hrm.. wish I could find a Universal build of some mac os x softphone |
| 19:13.09 | Blackthorn | So i'm in the sipura web interface, got a hint as to where you went? |
| 19:13.11 | KranZ | the rt31p2 works well |
| 19:13.21 | KranZ | spa-941 are good also |
| 19:13.46 | MACscr | im in the midst of planning out my first asterisk system |
| 19:13.58 | MACscr | needing it for my web hosting company |
| 19:14.09 | KranZ | i just deployed spa-941 phones with asterisk as the pbx |
| 19:14.20 | MACscr | remote agents is going to be biggest part |
| 19:14.44 | blitzrage | the only thing I don't like about the RT31P2 is the long response timeout unless you hit # |
| 19:14.57 | *** join/#asterisk websae (n=websae@209-252-79-66.ip.mcleodusa.net) |
| 19:14.59 | KranZ | blitzrage: you're dialplan isnt US |
| 19:15.01 | KranZ | er your |
| 19:15.07 | Blackthorn | ok i'm workign with the spa-2000 so perhaps that function isn't in there? |
| 19:15.11 | blitzrage | I haven't found any way of changing it (if you dial 1+areacode+number then it works) |
| 19:15.28 | blitzrage | KranZ: like I mentioned above, only matches right away on 11 digit dialing |
| 19:15.33 | KranZ | (011,xx.|*xx|[3469]11|0|00|<:1408>[2-9]xxxxxx|1[2-9]xx[2-9]xxxxxx) |
| 19:15.41 | KranZ | try that |
| 19:15.48 | blitzrage | KranZ: where is there configured in the RT31P2 ? |
| 19:15.53 | blitzrage | /s/there/that |
| 19:15.57 | KranZ | do you have an unlocked one? |
| 19:16.15 | blitzrage | what do you mean by unlocked? I guess I haven't looked at what that device can do closely enough :) |
| 19:16.31 | KranZ | the model should be rt31p2-na |
| 19:16.35 | KranZ | on the lable on the bottom |
| 19:16.42 | blitzrage | ahhh... not na afaik -- its about a year old now |
| 19:16.56 | blitzrage | whats the difference? |
| 19:17.04 | KranZ | did you get it through vonage or another provider? |
| 19:17.22 | KranZ | the diff is you have the same access as you would a regular sipura |
| 19:17.25 | blitzrage | I just bought one -- its not locked to a provider |
| 19:17.35 | blitzrage | I see what you mean now :0 |
| 19:17.39 | blitzrage | errr... :) |
| 19:17.41 | KranZ | look at the lable on the bottom of the unit |
| 19:18.03 | blitzrage | KranZ: what menu is that pattern matching under? I'm going to have to look that up when I get home this weekend... maybe I've just missed it... |
| 19:18.06 | blitzrage | hrmmm :) |
| 19:18.49 | KranZ | fyi, the pattern prefixes 1408 to a 7digit number |
| 19:19.05 | KranZ | so either remove the <:1408> or change it to your area code |
| 19:21.17 | blitzrage | hrmm... gotta find the prefix pattern... totally missed it |
| 19:22.11 | KranZ | anyone else love it when vonage's stock drops even further? |
| 19:23.27 | *** join/#asterisk darby_t (i=darby_t@aapc8.neoplus.adsl.tpnet.pl) |
| 19:25.23 | blitzrage | KranZ: I do... and I don't -- I think if Vonage goes under that will hurt VoIP adoption in the consumer market |
| 19:25.45 | blitzrage | KranZ: it will look back on everyone in the general publics eye |
| 19:26.41 | *** join/#asterisk jeffgus (n=jeffgus@greengables.zimage.com) |
| 19:27.21 | Blackthorn | YES! double ring gone.. ohh man that makes me so happy.. |
| 19:27.25 | mitcheloc | KranZ: what do you have against vonage?? |
| 19:27.27 | Blackthorn | I think i go home an hour early today :) |
| 19:27.44 | KranZ | well... from a clec standpoint... |
| 19:27.46 | sevard | how the hell is that sad? |
| 19:28.58 | Blackthorn | vonage is trying to compete on price to much, they should market a good product at decent price that they can maintain good service. |
| 19:30.29 | coppice | how could vonage charge more than they do? they are already rather expensive |
| 19:30.40 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 19:31.25 | blitzrage | agreed |
| 19:31.38 | KranZ | they're already not making money |
| 19:31.41 | blitzrage | I'm glad they are expensive -- helps me undercut them :) |
| 19:31.47 | blitzrage | they spend WAY too much money on advertising |
| 19:31.53 | *** join/#asterisk caloi (n=caloi@nat-66-218-1-215.usadatanet.com) |
| 19:31.59 | KranZ | they're losing their asses on the LD minutes |
| 19:32.01 | blitzrage | its been a money losing company since day once |
| 19:32.02 | drray | I don't see how vonage wins |
| 19:32.10 | fholmes | I am having a problem with my SIP softphone. It was working until I unregistered it and I am now trying to re-register as a different user. Now I cannot even re-register as the old user anymore. |
| 19:32.12 | coppice | most spending in telecoms today is on advertising |
| 19:32.28 | drray | because all the bells have to do is decide to crush them, and it is over for vonage |
| 19:32.37 | KranZ | fholmes: run some packet sniffer on your computer or the asterisk box |
| 19:32.39 | fholmes | I keep getting Username/auth name mismatch for some reason. |
| 19:32.42 | KranZ | www.ethereal.com |
| 19:33.24 | fholmes | I just happen to have it installed already. Let me know if there are some commands I can use that will make it as easy as possible to figure out what is going on. |
| 19:34.08 | KranZ | filter out your asterisk ip |
| 19:34.15 | KranZ | 'host x.x.x.x' |
| 19:34.53 | *** join/#asterisk AlexCTI (n=alex@adsl-074-238-025-003.sip.mia.bellsouth.net) |
| 19:37.14 | fholmes | So what am I looking for? |
| 19:37.29 | Blackthorn | how can you say 'there too expensive" and then say there loosing money on ld minutes... They've got plans out there for like $15 bucks... The few customers that I have, i am serving over wireless 802.11b for $35. |
| 19:39.03 | blitzrage | yah... but when you charge someone $35 for a plan, $20 of that is to pay for the termination, and then spend another $50 on advertising... its a losing system |
| 19:40.33 | Blackthorn | umm perhaps. but 1) i'm not paying $50 for advertising because i'm only marketing to my wireless customers and 2) i already have there e-mail. |
| 19:41.09 | Blackthorn | I'm just saying if vonage is only charging $15 and spending all that money on advertising i agree with you. can't make money that way |
| 19:41.52 | _Sam-- | i dont thik the average account at vonage is costing them 20/month for termination/orig. |
| 19:42.00 | blitzrage | I don't remember saying anything about your company... I thought we were talkinga bout vonage |
| 19:42.10 | Dr-Linux | blitzrage: Hi :) |
| 19:42.17 | sivana | vonage has been educating the public... their gap is closing |
| 19:42.21 | blitzrage | Dr-Linux: hoi |
| 19:42.33 | coppice | most telecoms companies now spend more on recruiting subscribers than on providing them service |
| 19:42.38 | *** join/#asterisk ManxPower (i=ewieling@184.sub-70-196-103.myvzw.com) |
| 19:42.41 | blitzrage | sivana: exactly -- which is why I think it'd be a bad thing if they went under |
| 19:42.47 | Blackthorn | true sivana thats the one very good thin they have done for all of us. |
| 19:42.51 | _Sam-- | and if the value per subscriber in a resale of the customers becomes more valubale than the cash flow of the customers...then you can understand their rationale |
| 19:42.56 | *** join/#asterisk chainey (n=jeremy@dsl017-031-038.lax1.dsl.speakeasy.net) |
| 19:43.08 | _Sam-- | you are only looking at cash flow, not value of the biz. |
| 19:43.27 | *** join/#asterisk castro2006 (n=khaled@83.244.91.231) |
| 19:43.39 | blitzrage | how many times have you explained VoIP by saying something like, "You know... like Vonage?" |
| 19:43.41 | sivana | blitzrage: yup, but they won't their loss/profit gap is closing -- they'll survive |
| 19:43.52 | websae | VOnage is a terrible standard |
| 19:44.01 | blitzrage | sivana: ideally... but the IPO thing really didn't make things any better -- it made things worse |
| 19:44.01 | websae | no one can possibly compare to vonage |
| 19:44.04 | KranZ | what's the latest zaptel stable ? |
| 19:44.06 | KranZ | im lazy |
| 19:44.07 | castro2006 | i need SIP soft phone to install on Debian ?? |
| 19:44.11 | sivana | blitzrage: yea, that whole employee program |
| 19:44.18 | blitzrage | websae: its not a standard you refer to -- its a reference point that people recognize |
| 19:44.21 | KranZ | ~zaptel |
| 19:44.23 | jbot | well, zaptel is zapata telephony interface. A low level interface designed to abstract hardware access to a variety of devices for BRI, PRI or analogue access. |
| 19:44.25 | websae | Vonage SOPs are completely radical from the norm of these other ma and pa shop VoIP providers |
| 19:44.47 | eKo1 | I'm a ma and pa |
| 19:44.53 | eKo1 | VoIP provider |
| 19:44.57 | eKo1 | :P |
| 19:45.12 | castro2006 | ?? |
| 19:45.23 | websae | eKo1: the best thing that could happen to you is that you get bought out |
| 19:45.25 | _Sam-- | ma and pa: did you get hacked by the dude? http://www.nytimes.com/2006/06/08/technology/08voice.html |
| 19:45.30 | eKo1 | castro2006: use xtenlite |
| 19:45.42 | eKo1 | websae: that is exactly what we're aiming for. |
| 19:45.52 | sivana | websae: what other option do you have.. you think you'll buy AT&T or any other Bell |
| 19:46.00 | blitzrage | websae: indeed :) |
| 19:46.12 | TheCompWiz | I will probably get shot for asking this... but has anyone had any luck using skypeout for outbound calls? |
| 19:46.22 | eKo1 | I hope someone buys us quick and for 1.5 times our price :) |
| 19:46.24 | websae | sivana: obviously i don't think that, as i just said my opinion |
| 19:46.29 | eKo1 | or twice even |
| 19:46.30 | castro2006 | i tried to install it but it did not work |
| 19:46.33 | r_evolution | blitz. get the rope. |
| 19:46.40 | castro2006 | it is tested on debian |
| 19:46.42 | eKo1 | castro2006: how? |
| 19:46.43 | sivana | websae: you said the "best thing that could happen".... and I'm saying.. it's the only thing |
| 19:46.58 | eKo1 | sivana: hahaha |
| 19:47.09 | websae | no you could go bankrupt.....in debt |
| 19:47.14 | websae | there are other things that could happen |
| 19:47.14 | sivana | haha.. true |
| 19:47.19 | _Sam-- | i dont think there will be much consolodation (buy outs) in the VOIP arena...why would anyone (comcast, verizon, etc) need to buy a customer base when they can take them over by default in masses over time |
| 19:47.20 | r_evolution | yeah cuz ya know... you certainly can't make money as a small company. |
| 19:47.24 | websae | which happens for quite a few |
| 19:47.28 | *** join/#asterisk techie (n=gus@antibala.com) |
| 19:47.55 | TheCompWiz | does anyone know if it can be done? (skypeout) |
| 19:48.09 | r_evolution | ... |
| 19:48.12 | websae | the best VoIP company/provider that has a chance at climbing the ladder....is one that uses VoIP to create a niche market |
| 19:48.14 | r_evolution | why's it gotta be fancy? |
| 19:48.21 | r_evolution | now let's hang and shoot CompWiz |
| 19:48.27 | r_evolution | maybe not in that order though... |
| 19:48.42 | TheCompWiz | r_evolution: have you tried? |
| 19:48.52 | r_evolution | no. we're going to be shooting and hanging you now |
| 19:49.03 | r_evolution | just out of curiousity... CompWiz. |
| 19:49.11 | r_evolution | if you're in an asterisk channel... |
| 19:49.24 | r_evolution | asterisk != skype |
| 19:49.36 | r_evolution | why exactly are you asking about using skype? |
| 19:49.40 | TheCompWiz | r_evolution... I'm trying to setup skype as a trunk for outbound dialing in asterisk... |
| 19:50.16 | TheCompWiz | so when user a... picks up his phone & dials a number... it's on skype's dime. |
| 19:50.18 | blitzrage | TheCompWiz: skype is a proprietary application -- via skypeout you can call asterisk as long as Asterisk is available via the PSTN |
| 19:50.38 | r_evolution | ^ |
| 19:50.45 | _Sam-- | you can do it though |
| 19:50.51 | _Sam-- | you need a skype phone adapter or something |
| 19:50.52 | TheCompWiz | blitzrage: I know that... but there are apps out there like "Skype2Sip" that's supposed to translate skype into SIP. |
| 19:50.53 | sivana | yes.. with an fxs/fxo |
| 19:50.55 | sivana | but it's lame |
| 19:50.56 | _Sam-- | there was someone here that did it |
| 19:51.04 | mitcheloc | if i might join in... i think the real killer for pots & voip companies, is going to be a bittorrent-like configuration for phones..... |
| 19:51.13 | sivana | an probably violates their tos |
| 19:51.28 | eKo1 | mitcheloc: please elaborate |
| 19:51.44 | mitcheloc | i mean that there will be no companies to terminate calls, just phones calling phones directly (in an easy to do manner) |
| 19:51.54 | r_evolution | where you gets parts of everyones voice from various locations? |
| 19:51.56 | r_evolution | awesome. |
| 19:51.59 | blitzrage | TheCompWiz: so go try it and tell us how it went |
| 19:52.08 | blitzrage | we don't hand hold in here |
| 19:52.18 | mogorman | mitcheloc, that wont exist |
| 19:52.23 | r_evolution | yeah... im curious to know wtf would make that work? O_o |
| 19:52.24 | sivana | especially for skype related stuff |
| 19:52.31 | *** join/#asterisk southtel (n=slester@c-67-191-211-17.hsd1.ga.comcast.net) |
| 19:52.32 | mogorman | as there is a level of quality needed that wont be there |
| 19:52.40 | r_evolution | im saying... seriously... it sounds like he's trying to setup a ITSP type solution |
| 19:52.41 | mogorman | but i do think service providers will operate like that |
| 19:52.44 | mogorman | like over dundi |
| 19:52.44 | r_evolution | except terminate to skype |
| 19:52.49 | mitcheloc | obviously the bittorrent protocol wouldn't work, but some sort of distriubuted mechanism would be great... |
| 19:52.51 | southtel | Has anyone successfully sent faxes using a TDM110P? |
| 19:53.13 | mitcheloc | dundi seems to be close to what i'm talking about, i'm thinking it'll take the next step beyond that though |
| 19:53.18 | mogorman | right without a party to trust |
| 19:53.20 | mitcheloc | think web 1.0 vs web 2.0, and so on |
| 19:53.21 | mogorman | it wont work |
| 19:53.27 | r_evolution | hehe |
| 19:53.31 | mogorman | dundi is what you are talking about |
| 19:53.33 | r_evolution | i think web 2.0 and i think userfriendly ;x |
| 19:53.43 | mogorman | but it cant work on a grand scale without people watching it |
| 19:53.44 | r_evolution | web 2.0 = omgwtfbbq! o rly? ya rly! |
| 19:53.47 | r_evolution | ;)\ |
| 19:53.49 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 19:53.57 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 19:54.06 | mitcheloc | mogorman: to be fair, i wouldn't have thought a serverless way to share files could happen... the technology will evolve... |
| 19:54.26 | mogorman | there is no such things as a serverless way to share files |
| 19:54.32 | mogorman | we just have more servers now.... |
| 19:54.52 | r_evolution | yeah i think he's trying to say he never thought there wouldn't be a server-based storage for those files |
| 19:54.53 | mitcheloc | mogorman: you know what i mean ;), there will be a way to do something like this |
| 19:54.55 | r_evolution | i.e. ftp etc |
| 19:55.09 | *** join/#asterisk techie (n=gus@brutus.voipops.net) |
| 19:55.11 | mogorman | they just made ftp easier |
| 19:55.19 | mogorman | no one has gotten rid of what ftp is |
| 19:55.23 | southtel | (TE110P, rather) |
| 19:55.27 | r_evolution | well of course not |
| 19:55.30 | r_evolution | ftp is still useful |
| 19:55.40 | mogorman | i think the closest thing to what you are talking about is email |
| 19:55.41 | mogorman | or jabber |
| 19:55.51 | r_evolution | but you can't say a lot of places don't encourage using bt in place of ftp |
| 19:55.54 | mitcheloc | to put it in those terms, when calling phones directly gets easier, you will see a huge decline in actual need for phone service |
| 19:55.55 | mogorman | but both are controlled by verisign at some level |
| 19:55.58 | r_evolution | conserve bandwidth etc |
| 19:56.03 | mogorman | there needs to be a guy in charge |
| 19:56.15 | KranZ | elect KranZ! |
| 19:56.15 | r_evolution | ill agree with that in part... |
| 19:56.22 | r_evolution | not the KranZ part. |
| 19:56.25 | KranZ | doh |
| 19:56.27 | Blackthorn | lol |
| 19:56.28 | r_evolution | ;) |
| 19:56.41 | r_evolution | people are too fucking dumb to be in charge of themselves usually |
| 19:56.49 | r_evolution | a person may be smart... but people are effin retarded |
| 19:57.02 | mitcheloc | agreed, like someone is in charge of the .com system... , it'll get easier and easier ;), eventually you'll be so seemless using msn + talk or aim+talk,or whatever, i think it's going to slowly remove the need for phone companies... |
| 19:57.14 | mitcheloc | the cell phone companies have the best security atm ;) |
| 19:57.27 | KranZ | mitcheloc: until it's reliable, commercial businesses will stay away |
| 19:57.29 | mogorman | but thats all phone companies are |
| 19:57.44 | _Sam-- | KranZ: what other type of business besides commercial is there? |
| 19:58.03 | KranZ | home office |
| 19:58.08 | *** join/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 19:58.17 | Nivex | non-profit business? |
| 19:58.23 | mogorman | someone who routes numbers to users |
| 19:58.28 | _Sam-- | its not a business if its non profit |
| 19:58.29 | mogorman | just like verisign |
| 19:58.35 | _Sam-- | its a non profit organization |
| 19:58.38 | *** join/#asterisk Soybomb (i=Soybomb@71-8-250-35.dhcp.mtvr.il.charter.com) |
| 19:58.52 | sorush20 | I want to be able make calls from my computer to lanlines and mobiles and recieve the calls trough my computer using my landline... |
| 19:59.00 | sorush20 | I want to be able record conversations.. |
| 19:59.01 | *** join/#asterisk kitche (n=el_lupo@pool-70-16-34-92.buff.east.verizon.net) |
| 19:59.02 | _Sam-- | any event sorry for splitting hairs, couldnt help it. |
| 19:59.08 | sorush20 | is this the best program for it? |
| 19:59.44 | KranZ | sorush20: yeah, i recommend you mess with asterisk at home |
| 19:59.45 | sorush20 | I have broadband.. wireless connection to a router.. and my phone line is separate from broadband cable connection.. |
| 19:59.54 | r_evolution | Hey sorush... I recommend you get a tin can with a string. |
| 19:59.59 | truz_`24 | lol |
| 20:00.22 | truz_`24 | There seems to be a lot of fluff out there concerning voip. |
| 20:00.28 | KranZ | sorush20: "Asterisk@Home" |
| 20:00.33 | sorush20 | tin can with a string does not record conversations.. |
| 20:00.34 | KranZ | ~aoh |
| 20:00.39 | r_evolution | yes it does |
| 20:00.41 | Blackthorn | tin can with a string. hehe and this form a guy with the nick evolution :P |
| 20:00.47 | KranZ | ~asterisk@home |
| 20:00.48 | jbot | from memory, asterisk@home is http://asteriskathome.sourceforge.net/, or http://www.voip-info.org/tiki-index.php?page=Asterisk+at++Home |
| 20:00.50 | r_evolution | ;x |
| 20:00.55 | Blackthorn | would that be r(reverse)_evolution? |
| 20:01.00 | r_evolution | yesssss |
| 20:01.05 | r_evolution | DEVOLUTION BITCHES! |
| 20:01.12 | r_evolution | TIN CAN WITH A STRING ALL THE WAY! |
| 20:01.15 | [TK]D-Fender|AFK | Hey I've got an emergency request I could use some quick tips on : I need to track all IP traffic from 2-3 hosts here and what I'm looking to do is set they default routes in DHCP to pass through my linux server which would then forward the packets on to our real internet router (SonicWALL). I am hoping to log EVERYTHING through Ethereal. Is this viable and are there any other tricks I'd need to do to prep for this? IP forwa |
| 20:01.24 | Soybomb | hello all, my boss just got me a 4t1 card to play with (generic tormenta clone from govarion.com) - it gets deteced as a tormenta 1 quad t1 card, but ztcfg fails with ZT_SPANCONFIG failed on span 1: No such device or address (6) |
| 20:01.31 | *** join/#asterisk stormfr (n=StorM@stardust.noc.frontier.fr) |
| 20:01.32 | *** join/#asterisk Wowzers10 (n=pbaker@nnat-gw.adeptra.com) |
| 20:02.02 | KranZ | [TK]D-Fender|AFK: sounds about right |
| 20:02.11 | KranZ | [TK]D-Fender|AFK: now you need to reset their dhcp leases |
| 20:03.06 | Wowzers10 | hello all, with straight through asterisk and no agi - is there a way to use ackcall and announce but have it play " please press pound to connect " after the call is answered? |
| 20:03.30 | Wowzers10 | IE: they dont hear silence, so they know what to do |
| 20:04.19 | sorush20 | KranZ: for everyone's info asterisk@home is now.. http://www.trixbox.org/, now who is in the know.. ? |
| 20:04.31 | blitzrage | #freepbx is in the know |
| 20:04.59 | r_evolution | i'm telling you sorush... |
| 20:05.00 | r_evolution | TIN |
| 20:05.00 | r_evolution | CAN |
| 20:05.01 | r_evolution | WITH |
| 20:05.02 | r_evolution | STRING |
| 20:05.06 | blitzrage | very effective |
| 20:05.10 | r_evolution | yes. |
| 20:05.14 | blitzrage | tin can telecom, inc. <-- my company! |
| 20:05.14 | KranZ | i recommend extra string |
| 20:05.18 | KranZ | they can break sometimes |
| 20:05.23 | blitzrage | 20lb. test |
| 20:05.27 | sorush20 | r_evolution: no thank |
| 20:05.33 | sorush20 | s |
| 20:05.50 | coppice | wired? wireless? stringed? |
| 20:05.55 | KranZ | you might have to condition the string to make sure it will carry the signal |
| 20:05.56 | r_evolution | i'm telling you sorush |
| 20:05.58 | r_evolution | tin can with a string |
| 20:06.02 | r_evolution | there's even a cat in canada |
| 20:06.04 | r_evolution | by the name of Life |
| 20:06.10 | r_evolution | he wrote a book about the tin can theory |
| 20:06.22 | KranZ | he was also on a hot tin roof |
| 20:06.26 | Soybomb | anyone know if thats a driver issue? thats my guess but its odd to see anything in the dmesg then i'd think |
| 20:06.28 | [TK]D-Fender|AFK | KranZ : Easily done. |
| 20:06.52 | r_evolution | see... you've got all these people advising you... but you won't take the advice :( |
| 20:07.16 | blitzrage | advice is overrated |
| 20:07.23 | stormfr | hello, i don't know if it's a bug or a new directive but when a call is answered on second dial command, cdr disposition is set to FAILED (billsec>0). This problem exist since around january but not before. |
| 20:07.39 | r_evolution | sometimes. |
| 20:09.35 | r_evolution | you know what's not overrated? |
| 20:09.35 | r_evolution | http://www.addictinggames.com/curveball.html |
| 20:09.37 | r_evolution | that game. |
| 20:09.44 | *** join/#asterisk tekati (n=captain@cpe-66-75-215-63.bak.res.rr.com) |
| 20:10.00 | *** part/#asterisk chainey (n=jeremy@dsl017-031-038.lax1.dsl.speakeasy.net) |
| 20:12.46 | Blackthorn | thanks for that, i'm off.. just don't ask how far |
| 20:13.11 | Blackthorn | i could have swore i wrote "thanks for the chat" |
| 20:13.24 | r_evolution | ? |
| 20:13.28 | r_evolution | you're on drugs. |
| 20:13.30 | r_evolution | ;)( |
| 20:13.36 | Blackthorn | i wish :P |
| 20:14.33 | *** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it) |
| 20:15.24 | markus99 | hi all, I have an issue on my zap modem when in conversation I hear beeps once in a while, someone said that it might have something to do with dtmf, not sure even what to look under |
| 20:15.24 | sorush20 | so why should I use freepbx rather than asterisk? |
| 20:15.58 | mitcheloc | you shouldn't |
| 20:16.03 | mitcheloc | if anything, you use them together |
| 20:16.55 | r_evolution | O_o |
| 20:16.57 | Soybomb | in my dmesg after registered tormenta2 pci i see scb2_flash: warning - can't reserve rom window, continuing" ---is that a problem related to the card? (bsd guy here) |
| 20:17.03 | r_evolution | tin can.... with string. |
| 20:19.01 | KranZ | damn curveball |
| 20:19.02 | *** join/#asterisk Delta239 (n=paparapa@201.218.116.114) |
| 20:19.12 | r_evolution | hehehehe |
| 20:19.17 | r_evolution | you're playing now arent you kranZ |
| 20:19.20 | r_evolution | it's addictive as shit |
| 20:19.21 | KranZ | mebbe |
| 20:19.24 | r_evolution | ;x |
| 20:19.28 | r_evolution | it's like geek-crack |
| 20:19.41 | *** part/#asterisk kitche (n=el_lupo@pool-70-16-34-92.buff.east.verizon.net) |
| 20:19.46 | r_evolution | the lan admin here got the net admin who got me |
| 20:19.51 | r_evolution | now i gotchu! |
| 20:20.17 | *** join/#asterisk Gunnar (n=gunnar@34.80-203-209.nextgentel.com) |
| 20:20.51 | r_evolution | ~jbot |
| 20:20.52 | jbot | rumour has it, jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot |
| 20:21.09 | r_evolution | ~blitzrage |
| 20:21.11 | jbot | from memory, blitzrage is a super cool fellow |
| 20:21.18 | r_evolution | someone lied to jbot :-\ |
| 20:22.32 | blitzrage | ouch! |
| 20:22.50 | r_evolution | HAR |
| 20:22.53 | r_evolution | http://www.addictinggames.com/curveball.html |
| 20:22.56 | r_evolution | go play blitz |
| 20:23.16 | blitzrage | probably needs java |
| 20:23.29 | blitzrage | in linux on a computer in class with no java :) |
| 20:23.38 | r_evolution | :( |
| 20:23.39 | r_evolution | flash |
| 20:24.36 | *** part/#asterisk cytrak (n=btorch@208.63.19.179) |
| 20:25.03 | *** join/#asterisk mog_home (n=mogorman@68.62.237.103) |
| 20:26.31 | Wowzers10 | hello all, with straight through asterisk and no agi - is there a way to use ackcall and announce but have it play " please press pound to connect " after the call is answered? - this is so they do not hear silence |
| 20:28.14 | r_evolution | everyone is playing curveball wowzers. |
| 20:28.15 | r_evolution | http://www.addictinggames.com/curveball.html |
| 20:28.17 | r_evolution | go play :) |
| 20:29.29 | KranZ | dammit |
| 20:29.40 | KranZ | level 8 and i click a bookmark folder |
| 20:29.45 | KranZ | half the screen is covered |
| 20:29.55 | KranZ | i managed to keep playing for 10 more secs |
| 20:30.11 | KranZ | hmm...full screen mode |
| 20:30.29 | r_evolution | heh |
| 20:30.40 | r_evolution | dude level 8 and up gets CRAZY |
| 20:30.47 | r_evolution | im saying that ball is all OVER the effin place |
| 20:30.53 | KranZ | yeah, sucks u into the zone |
| 20:31.04 | r_evolution | i know |
| 20:31.10 | r_evolution | imma laugh when i redball and fall over and twitch |
| 20:31.47 | tzafrir | anybody here uses res_zeroconf? Any published version of it? |
| 20:32.17 | KranZ | http://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf |
| 20:32.19 | KranZ | better link for it |
| 20:32.31 | r_evolution | how so? |
| 20:32.35 | r_evolution | oh |
| 20:32.36 | r_evolution | dur |
| 20:32.39 | r_evolution | straight to the flash |
| 20:33.07 | r_evolution | O_O |
| 20:33.14 | r_evolution | perfect sound track |
| 20:33.17 | r_evolution | fast paced breaks |
| 20:33.25 | KranZ | close all the toolbars and goto full screen mode |
| 20:33.34 | *** join/#asterisk JINDAL (n=root@220.226.36.2) |
| 20:33.50 | *** join/#asterisk postel (n=jp@unaffiliated/postel) |
| 20:34.21 | r_evolution | i might never come back |
| 20:34.33 | *** join/#asterisk vechers (i=vechers@64.61.117.138) |
| 20:34.53 | *** part/#asterisk vechers (i=vechers@64.61.117.138) |
| 20:35.01 | *** join/#asterisk bjohnson (n=bjohnson@jecinc.tor.istop.com) |
| 20:35.39 | r_evolution | yep |
| 20:35.41 | r_evolution | def NOT coming back |
| 20:38.02 | http | crap |
| 20:41.14 | *** join/#asterisk zotz (n=zotz@24.244.133.115) |
| 20:41.59 | *** join/#asterisk Iam8up|lpy (n=iam8up@cpe-24-210-253-66.woh.res.rr.com) |
| 20:42.18 | KranZ | gah, only 3:42 |
| 20:42.26 | r_evolution | ? |
| 20:42.27 | Iam8up|lpy | can anyone tell me where to find the pdf document that has all the IVR commands for the digium iaxy s101? |
| 20:42.28 | r_evolution | heh |
| 20:42.30 | r_evolution | addictive ;x |
| 20:42.37 | r_evolution | hey ate up |
| 20:42.38 | r_evolution | go to |
| 20:42.39 | r_evolution | http://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf |
| 20:42.39 | mogorman | ivr commands? |
| 20:42.42 | r_evolution | it's there. |
| 20:42.48 | Iam8up|lpy | the two docs on digium's website aren't what i'm lookin for, and i can't find it on voip-info.org |
| 20:43.21 | Iam8up|lpy | mogorman - so you can hit like **** on your phone while plugged into the ata and it has it's own voice menu? |
| 20:43.35 | Iam8up|lpy | i thought it was ****, but it didn't work - nor did #### |
| 20:44.23 | C4T3l | what kind of ata you have? |
| 20:44.43 | Iam8up|lpy | digium iaxy s101? |
| 20:44.52 | Iam8up|lpy | maybe the digium doesn't have one..i may be thinking of the sipura ata101 |
| 20:44.59 | Iam8up|lpy | ata1001 rather |
| 20:45.12 | C4T3l | that sounds like sipura to me |
| 20:45.19 | *** join/#asterisk raidenz (i=raiden@205-200-66-136.static.mts.net) |
| 20:45.28 | Iam8up|lpy | mmm |
| 20:45.47 | *** part/#asterisk JINDAL (n=root@220.226.36.2) |
| 20:45.48 | Iam8up|lpy | that may be right..cause i can only figure out the digium's ip address by having it dhcp and looking at the leases given... |
| 20:45.57 | mogorman | yeah the iaxy doesnt have anything like that |
| 20:46.04 | Iam8up|lpy | damn... |
| 20:46.08 | Iam8up|lpy | my mistake, sorry |
| 20:46.12 | mogorman | no prob |
| 20:46.20 | raidenz | hi guys |
| 20:47.26 | r_evolution | hey |
| 20:47.27 | r_evolution | go here |
| 20:47.27 | r_evolution | http://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf |
| 20:48.20 | dlynes_office | Anyone have any idea what this would mean? |
| 20:48.22 | dlynes_office | !! Not good - head of queue has not been transmitted yet |
| 20:48.22 | dlynes_office | Jun 8 13:36:47 WARNING[24732]: chan_iax2.c:692 jb_warning_output: Resyncing the jb. last_delay 56, this delay 17605, threshold 1072, new offset -17605 |
| 20:49.23 | raidenz | Does anyone have an updated test g729 codec that works with the latest Asterisk SVN (1.4)? Please msg me. |
| 20:51.46 | MikeJ[Laptop] | what exactly is a "test" g729 codec? |
| 20:52.16 | r_evolution | one he doesnt have to pay for? |
| 20:52.17 | r_evolution | ;x |
| 20:52.22 | raidenz | no |
| 20:52.29 | raidenz | the test gcc/ipp g729 codec. |
| 20:54.49 | docelmo | YOUR STEALING FROM DIGIUM AND BREAKING THE LAW! |
| 20:55.05 | docelmo | if your in the US anyhow |
| 20:55.13 | MikeJ[Laptop] | they are not stealing from digium.. they may be stealing from the g729 consortium |
| 20:55.21 | ids2500 | stealing from digium? |
| 20:55.21 | ids2500 | lol |
| 20:55.45 | ids2500 | NEWS FLASH: digium has no more g729 codecs today... raidenz stole them all! |
| 20:55.49 | r_evolution | yeah but regardless of who... the codec is still cheap |
| 20:56.08 | ids2500 | raidenz: you won't get any modules to work with SVN because they changed the headers and stuff... |
| 20:56.10 | KranZ | $10 a chan is not cheap |
| 20:56.14 | MikeJ[Laptop] | depending on the license of the ipp code, and what patent rights it comes with, it may be totally legal, I have never looked at it |
| 20:56.16 | r_evolution | O_o |
| 20:56.20 | r_evolution | compared to what? free? |
| 20:56.40 | KranZ | to fill a quad pri is + $960 |
| 20:57.19 | *** join/#asterisk mtaht4 (n=m@207.47.5.58.static.nextweb.net) |
| 20:57.23 | r_evolution | touche. |
| 20:57.46 | r_evolution | go play curveball |
| 20:57.49 | r_evolution | you're stopping me |
| 20:57.50 | KranZ | heh |
| 20:57.56 | KranZ | i heard footsteps |
| 20:58.03 | r_evolution | i sit facing the door :) |
| 20:58.09 | docelmo | MikeJ[Laptop] to use the IPP code the "free" one is for non-commercial personal use so inherently yes you could.. But in a business environment nope.. |
| 20:58.33 | r_evolution | i've come to the conclusion that i need a trackball |
| 20:58.35 | KranZ | use ua's which have the 726 codec |
| 20:58.49 | KranZ | works well |
| 21:00.05 | r_evolution | fuck |
| 21:00.06 | r_evolution | i quit |
| 21:00.18 | r_evolution | i got to L8 with 3 extras... and got cocky and lost em :( |
| 21:00.34 | KranZ | ha |
| 21:00.44 | KranZ | ok, 1 more game |
| 21:00.49 | r_evolution | yeah |
| 21:00.56 | r_evolution | you say that |
| 21:00.59 | r_evolution | but thats like a crackhead |
| 21:01.03 | r_evolution | ok just ONE more rock |
| 21:01.27 | *** join/#asterisk ddn_ (n=Daniel@200.84.67.165) |
| 21:01.31 | ddn_ | hi all |
| 21:01.37 | *** join/#asterisk iulius (n=iulius@adsl-152-175-71.asm.bellsouth.net) |
| 21:02.38 | De_Mon | what game? |
| 21:03.37 | ddn_ | is it possible to install a voip server over a satellital internet connection? |
| 21:03.56 | r_evolution | http://farm.addictinggames.com/D78AQSAKQLQWI9/1258.swf |
| 21:05.01 | KranZ | ddn_: yes |
| 21:05.07 | KranZ | ddn_: will it work well |
| 21:05.15 | KranZ | ddn_: probably not |
| 21:05.23 | KranZ | you have latency issues and jitter problems |
| 21:05.30 | ddn_ | KranZ, so? yes or not. |
| 21:05.40 | KranZ | yes you can "install" it |
| 21:05.41 | ddn_ | r_evolution, check zuma at popcapgames.com |
| 21:06.04 | ddn_ | KranZ, and will work fine. isnt? |
| 21:06.19 | KranZ | but most likely, the connection wont be stable enough for voice traffic |
| 21:06.33 | KranZ | are you on the sat link now? |
| 21:06.39 | ddn_ | KranZ, ohhhhhhhhhhhhhh |
| 21:06.52 | ddn_ | KranZ, no |
| 21:07.27 | KranZ | brb, curveball |
| 21:08.27 | MACscr | sat wont be good at all because of the delay |
| 21:09.02 | MACscr | because of the distance it has to travel, sat. internet is not good for voip or online games |
| 21:09.12 | MACscr | fine for browsing or downloading though |
| 21:10.30 | ddn_ | anyone interested in installing voip servers in venezuela? |
| 21:11.07 | KranZ | if the latency were consistant, it would be alright |
| 21:11.13 | KranZ | but the jitter will kill it |
| 21:11.42 | justinu | any cisco QoS experts around? |
| 21:13.12 | justinu | need help with flow based fair queueing and diffserv |
| 21:15.22 | *** join/#asterisk blebleble (i=godie@caesar.godie.net) |
| 21:16.28 | blebleble | i have an extension that should be forwarding to voicemail, however at 5:05 our autoatendant kicks and is overridding it. Basically i have 5 trunks, 4 get auto attendant and one should get voicemail, yet the voicemail one is getting auto attendant anyone have any clues? |
| 21:16.50 | De_Mon | humph lvl 5 |
| 21:17.18 | blebleble | de_mon: is that meant for me? |
| 21:17.33 | r_evolution | afw curveball |
| 21:18.31 | eKo1 | justinu: I recommend #cisco |
| 21:18.53 | TheCompWiz | SWEET... just got skype to work with asterisk. |
| 21:19.07 | TheCompWiz | both in-dialing & out-dialing.... (one line only :() |
| 21:19.53 | justinu | eKo1: i'll check it out (hopefully it exists) :) |
| 21:20.05 | eKo1 | oh, it does. |
| 21:20.13 | eKo1 | I've been there numerous times |
| 21:21.47 | justinu | cool |
| 21:22.00 | ddn_ | anyone interested in installing voip servers in venezuela? call 001 619 374 0892 |
| 21:23.16 | Wowzers10 | is there a way to use ackcall and announce to answer the call play a message, then press # to bridge the call? - this is so the person answering knows its a call for them |
| 21:23.49 | justinu | it be hot |
| 21:25.01 | r_evolution | JUSTIN! |
| 21:25.04 | r_evolution | sup homie. |
| 21:25.16 | ddn_ | anyone interested in installing voip servers in venezuela? call 001 619 374 0892. ahhh prefer spanish |
| 21:25.24 | justinu | hey man |
| 21:25.28 | justinu | how's it? |
| 21:25.29 | r_evolution | how goes? |
| 21:25.38 | justinu | pretty good |
| 21:25.38 | r_evolution | pretty good... waiting for a DS3 to come in |
| 21:25.47 | justinu | we're getting a new one as well |
| 21:25.50 | r_evolution | got a temp solution as well |
| 21:25.51 | *** join/#asterisk Ariel_ (n=Ariel@70.46.87.158) |
| 21:25.52 | justinu | new datacenter too |
| 21:25.52 | r_evolution | chya. needed. |
| 21:25.54 | *** join/#asterisk redondos (n=redondos@190.48.27.53) |
| 21:25.55 | r_evolution | nice. |
| 21:26.14 | *** part/#asterisk wrmem (n=monnin@monnin-win.ci.uiuc.edu) |
| 21:27.00 | r_evolution | im contemplating 1.2.9.1 on the primary * |
| 21:27.14 | r_evolution | it's doing ok for the in-house so far... |
| 21:27.49 | r_evolution | actually seems to do a little better as far as the qualify goes... the phone on my desk would never stay qualified to the primary... but it hasn't lost conn to the in-house |
| 21:28.47 | redondos | Can you please take a look at this log? I can't start asterisk. I've got an E200P card and an X100P card connected. http://pastebin.com/768435 |
| 21:29.10 | brettnem | hey anyone have trouble running Asterisk->SDSL->Phones which small to medium offices (5 simult calls or so) |
| 21:29.40 | C4T3l | does anyone here use broadcom for voip termination, are they reliable??? |
| 21:29.56 | brettnem | s/which/with/ |
| 21:32.56 | *** join/#asterisk Rawplayer (i=kevin@ipc31055d2.oom-killer.org) |
| 21:33.02 | *** join/#asterisk fholmes (n=fholmes@rrcs-24-227-237-197.sw.biz.rr.com) |
| 21:34.12 | *** join/#asterisk saftsack (n=saftsack@p54A7D890.dip.t-dialin.net) |
| 21:34.22 | fholmes | I am having an issue with my sip softphone. I cannot get it to register all the sudden. I have not changed anything in my configuration. |
| 21:35.00 | C4T3l | firewall?? |
| 21:35.03 | C4T3l | does anyone here use broadcom for voip termination, are they reliable??? |
| 21:35.04 | justinu | r_evolution: you do any cisco work? |
| 21:36.09 | KranZ | brettnem: what's the upstream limit? |
| 21:36.23 | fholmes | I keep getting username/auth name mismatch. |
| 21:36.41 | KranZ | make sure your username and authname are the same |
| 21:36.42 | brettnem | KranZ: 2.2M |
| 21:36.53 | KranZ | what happens when you get 5 calls |
| 21:37.08 | fholmes | Is that two different settings in my sip.conf? |
| 21:37.16 | KranZ | it's probably on the client |
| 21:37.27 | brettnem | KranZ: who are you talking to? |
| 21:37.40 | KranZ | both of u |
| 21:37.41 | KranZ | heh |
| 21:37.47 | brettnem | heh |
| 21:37.50 | KranZ | brettnem: what happens when you hit 5 calls? |
| 21:38.15 | brettnem | Kranz: all calls are choppy and staticy.. other customers on the same asterisk box same router but different DSL line are just fine. |
| 21:38.16 | *** join/#asterisk pigpen2 (n=mark@m015f36d0.tmodns.net) |
| 21:38.26 | brettnem | local LAN for the phones is isolated from PCs |
| 21:38.45 | brettnem | werid eh? |
| 21:38.57 | KranZ | oh, so you have 2 vlans? |
| 21:39.01 | KranZ | one pc and one phones? |
| 21:39.08 | brettnem | no, they arn't vlans |
| 21:39.13 | brettnem | it's |
| 21:39.24 | C4T3l | does anyone here use broadcom for voip termination, are they reliable??? |
| 21:39.25 | KranZ | just 2 physical lans |
| 21:39.33 | brettnem | asterisk->router->router->dslam=>SDSL=>2xclients |
| 21:40.00 | KranZ | so the clients aren't local to the asterisk? |
| 21:40.21 | brettnem | no, in this example, there are two clients.. each on their own DSL circuit (In different bldgs) |
| 21:40.28 | KranZ | oh |
| 21:40.30 | r_evolution | no sir i do not :( |
| 21:40.43 | KranZ | and * is attached to both of the dslams? |
| 21:40.53 | KranZ | er both routers to each dslam |
| 21:41.17 | brettnem | asterisk is attached to a router, which is attached to another router which is attached to a dslam which has 2 SDSL customers hanging off of it.. one gets good service.. the other no so good |
| 21:42.44 | MikeJ[Laptop] | it's the routers! |
| 21:42.57 | file | MikeJ[Laptop]: !!!!!!!!! |
| 21:43.02 | MikeJ[Laptop] | hello |
| 21:43.08 | MikeJ[Laptop] | it's the interweb! |
| 21:43.14 | file | no no |
| 21:43.15 | file | intarweb |
| 21:43.17 | r_evolution | INTARWEB! |
| 21:43.20 | r_evolution | exactly. |
| 21:43.32 | r_evolution | tin can with a string... that's the solution brett. |
| 21:43.47 | r_evolution | low bandwidth... low latency... virtually no jitter. |
| 21:44.32 | r_evolution | here's a scary thought... |
| 21:44.38 | r_evolution | how does the laziest supervisor in the callcenter |
| 21:44.48 | r_evolution | become the one who's in charge of motivating people to collect money? |
| 21:45.01 | nahirean | r_evolution; That's quite easy. He knows how to play the game. |
| 21:45.11 | r_evolution | she. |
| 21:45.14 | r_evolution | quoting " someone to set goals for collections, set goals for orders numbers, implement motivational activities and the list goes on" |
| 21:45.15 | nahirean | Even more so |
| 21:45.31 | r_evolution | no. the person assigning the lazy person to the 'job' is another female |
| 21:45.33 | r_evolution | and in this case |
| 21:45.37 | r_evolution | the game is very easy |
| 21:45.43 | *** join/#asterisk dropdrive (n=dropdriv@c-66-30-112-44.hsd1.ma.comcast.net) |
| 21:45.44 | r_evolution | kiss lots of ass |
| 21:45.46 | r_evolution | thats it :) |
| 21:45.48 | nahirean | Right. |
| 21:45.57 | r_evolution | haha no seriously. |
| 21:45.59 | r_evolution | lots and lots. |
| 21:46.08 | nahirean | Yeah, until it's raw and pruned, right? |
| 21:46.20 | markus99 | why would my music on hold sound like crickets? |
| 21:46.29 | brettnem | MikeJ[Laptop]: I'm trying to convince my customer it's gotta be at the customer prem |
| 21:46.35 | *** join/#asterisk ToyMan (n=stuq@user-12lcqia.cable.mindspring.com) |
| 21:46.36 | Wowzers10 | ~jbot thebook |
| 21:46.44 | Wowzers10 | ~jbot the book |
| 21:46.56 | Wowzers10 | jbot thebook |
| 21:47.00 | Wowzers10 | hmm |
| 21:47.41 | r_evolution | ~book |
| 21:47.43 | jbot | i heard book is a book called Asterisk: The Future of Telephony which is found at http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11 |
| 21:47.43 | r_evolution | ~thebook |
| 21:47.48 | r_evolution | there ya go |
| 21:47.53 | docelm0 | ~mybutt |
| 21:47.54 | jbot | methinks mybutt is HUGE and stands for some funky stuff... |
| 21:48.07 | r_evolution | did you record crickets as teh moh markus? |
| 21:48.15 | r_evolution | ~docelmo |
| 21:48.19 | r_evolution | ~docelm0 |
| 21:48.20 | r_evolution | :( |
| 21:48.56 | *** join/#asterisk asterisk-dud (n=dwwollma@64-42-247-120.mb.skyweb.ca) |
| 21:49.30 | asterisk-dud | i have three outgoing lines, i would like asterisk to look for an open line from the three and use that to dial out |
| 21:49.46 | C4T3l | how do you turn off the sip debug feature? |
| 21:50.09 | fholmes | Alright so how do I force a logged in SIP agent to unregister? |
| 21:50.34 | fholmes | I did 'database show' and there is a registration in there for a connection that is no longer good. |
| 21:50.40 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 21:50.56 | *** join/#asterisk Lord_Drachenblut (n=Lord@74.129.228.28) |
| 21:50.58 | Lord_Drachenblut | hello |
| 21:51.20 | *** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it) |
| 21:52.11 | C4T3l | how do you turn off the sip debug feature? i dont need all of that info anymore |
| 21:52.16 | asterisk-dud | i have three outgoing lines, i would like asterisk to look for an open line from the three and use that to dial out |
| 21:52.55 | r_evolution | you can delete things in teh database holmes |
| 21:53.17 | CunningPike | C4T3l: sip no debug |
| 21:53.27 | fholmes | So what exactly is the database? Is it things that are in use right now basically? |
| 21:53.35 | r_evolution | no |
| 21:53.41 | r_evolution | it's just *'s internal database |
| 21:53.46 | fholmes | compared to sip show users which will list all of the users that are in the sip.conf file? |
| 21:53.56 | r_evolution | it'll list agents and other things |
| 21:54.00 | r_evolution | well |
| 21:54.08 | r_evolution | sip show users will list the users |
| 21:54.15 | r_evolution | not just the ones in the sip.conf file |
| 21:54.15 | JackEstorm | asterisk-dud: did you define them as a group in zapata.conf? if so you can use that group on dial out. |
| 21:54.16 | r_evolution | er |
| 21:54.37 | r_evolution | game time :-D |
| 21:55.04 | *** join/#asterisk hohum (n=dcorbe@12.117.204.18) |
| 21:55.41 | fholmes | So if I do database show and the top listing is : SIP/Reistry/admin : all the connection info. Is that something I can safely delete? (The connection is not valid anymore) |
| 21:55.59 | fholmes | *SIP/Registry/admin |
| 21:56.25 | MACscr | not sure if anyone has seen this, but i just found it off another link |
| 21:56.25 | MACscr | http://www.voipcharges.com/ |
| 21:56.28 | MACscr | pretty cool |
| 21:56.30 | fholmes | In otherwords what is SIP/Registry/username? |
| 21:59.54 | *** join/#asterisk [Airwolf] (n=airwolf@83.98.235.220) |
| 22:05.30 | *** join/#asterisk symmetre (n=s@198.87.2.15) |
| 22:06.15 | symmetre | ok, i have asterisk , now i have to goto a PSTN ..can you do that for free too? |
| 22:06.29 | MACscr | no |
| 22:06.51 | *** join/#asterisk pdavid (n=chatzill@adsl-068-209-191-127.sip.mob.bellsouth.net) |
| 22:06.59 | pdavid | what exactly do i need to do to enable music on hold? |
| 22:09.20 | justinu | r_evolution: what game? |
| 22:09.33 | *** join/#asterisk _CALLNET (n=charly@140.Red-217-125-115.staticIP.rima-tde.net) |
| 22:09.40 | _CALLNET | Hello all |
| 22:10.52 | CunningPike | symmetre: No - at some point you will have to a) buy hardware b) pay a telco c) pay a termination/origination provider d) some combination of all 3 |
| 22:11.26 | CunningPike | pdavid: Create a musiconhold.conf |
| 22:11.33 | CunningPike | pdavid: Put a call on hold |
| 22:12.38 | _CALLNET | Someone have any route to sell us ? We have 60 Millons montly of traffic |
| 22:14.07 | asterisk-dud | i have a channel bank and i would like to start with channel 11 in my zapata and zaptel conf |
| 22:14.20 | asterisk-dud | for some reason it's not working |
| 22:14.48 | CunningPike | asterisk-dud: pastebin your files |
| 22:14.51 | CunningPike | ~pb |
| 22:14.52 | jbot | extra, extra, read all about it, pb is a place to paste your stuff without flooding the channel - try http://pastebin.com/, or http://pastebin.ca, or http://channels.debian.net/paste, or for #oe use http://oe.pastebin.com, or http://bzflag.pastebin.ca/ |
| 22:15.00 | asterisk-dud | asterisk accepts it but when it dials, i dials zap/11 on the 11th hole |
| 22:16.17 | pdavid | cunningpike: yeah, i have, but the sound is not quite what i expected ;) |
| 22:16.17 | *** part/#asterisk MACscr (n=MACscr@66.73.154.70) |
| 22:16.30 | pdavid | do i need to make the adjustment in the zapata.conf file? |
| 22:16.46 | CunningPike | pdavid: Possibly - what does it sound like? |
| 22:16.47 | asterisk-dud | http://pastebin.com/768539 |
| 22:17.19 | pdavid | cunningpike: not sure if its the playback, or the goofy default moh music, sort of static and music blended |
| 22:17.52 | CunningPike | pdavid: Are you using mpg123? |
| 22:18.08 | asterisk-dud | cunningpike: the zaptel is at the bottom |
| 22:18.26 | pdavid | cunningpike: was reading that v1.2 does native playing now, so not sure. using default musiconhold.conf file |
| 22:18.35 | pdavid | [default] |
| 22:18.41 | pdavid | mode=quietmp3 |
| 22:18.43 | pdavid | etc.. |
| 22:19.09 | CunningPike | pdavid: Give native MOH a go - it helped us out a lot |
| 22:19.31 | pdavid | cunningpike: is there anything special i need to do to set it up? |
| 22:19.42 | asterisk-dud | cunningpike: did u get that? |
| 22:19.59 | CunningPike | asterisk-dud: On phone - one sec |
| 22:20.08 | asterisk-dud | ok |
| 22:21.25 | CunningPike | asterisk-dud: What does your Dial() statement look like |
| 22:22.17 | Shaun2222 | what am i doing wrong with this? exten => _2XX,3,Set(CALLERID(num)=DB(/AgentsMap/${CALLERIDNUM})) |
| 22:22.43 | Shaun2222 | it's setting the caller ID to DB(/AgentsMap/301) rather than the value from the key in the db |
| 22:22.46 | *** join/#asterisk znoG (n=gs@153-129-89-200.fibertel.com.ar) |
| 22:22.49 | CunningPike | asterisk-dud: Do you have a PRI connected to this card? |
| 22:23.48 | CunningPike | pdavid: There are instructions in the wiki for setting up native MOH - we convert our MOH to ulaw and play it natively |
| 22:24.12 | pdavid | cunningpike: same instructions as at astrecipes.net (transcoding to native formats one time?) |
| 22:24.54 | CunningPike | pdavid: I haven't seen those - we use sox to convert to ulaw from Windows WAV - I can get you the command if you need |
| 22:25.10 | pdavid | http://astrecipes.net/?n=152 |
| 22:25.15 | pdavid | seems like the same idea |
| 22:25.23 | pdavid | now that i can hear the music, its wayyyy too slow |
| 22:25.28 | pdavid | ? |
| 22:25.54 | CunningPike | pdavid: That's it - sounds like your sampling rate is wrong |
| 22:26.08 | pdavid | cunningpike: yep. off to fiddle! |
| 22:26.16 | CunningPike | pdavid: Have fun! |
| 22:26.17 | *** part/#asterisk znoG (n=gs@153-129-89-200.fibertel.com.ar) |
| 22:26.29 | CunningPike | asterisk-dud: Do you have a PRI? |
| 22:27.08 | CunningPike | asterisk-dud: OK - you said before - a channelbank |
| 22:28.54 | asterisk-dud | CunningPike: can u see anything worng, or can't this be done? |
| 22:30.09 | *** part/#asterisk sorush20 (n=sorush20@82-43-184-143.cable.ubr07.newm.blueyonder.co.uk) |
| 22:30.31 | CunningPike | asterisk-dud: I'm not 100% sure, but your zapata.conf looks like a mix up between FXO and PRI - but I don't have a channelbank, so I'm not sure how these are meant to be set up |
| 22:31.08 | asterisk-dud | well i'm using fxo ports in my channel bank, what is pri? |
| 22:31.37 | *** join/#asterisk jarg (n=jarg@200.56.225.61) |
| 22:32.01 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 22:32.17 | CunningPike | asterisk-dud: Isn't the connection between asterisk and your channelbank a PRI connection? |
| 22:32.32 | CunningPike | ~TE405P |
| 22:33.46 | asterisk-dud | t1 |
| 22:34.02 | asterisk-dud | CunningPike: t1 |
| 22:34.08 | asterisk-dud | yup |
| 22:34.11 | asterisk-dud | that's correct |
| 22:34.16 | *** join/#asterisk ManxPower (n=ewieling@24-179-48-91.static.slid.la.charter.com) |
| 22:34.46 | CunningPike | asterisk-dud: Right - so signaling of fxo-ks is incorrect for a T1, right? |
| 22:35.03 | asterisk-dud | no |
| 22:35.04 | CunningPike | asterisk-dud: What does the documentation for your channelbank say? |
| 22:35.11 | asterisk-dud | it works perfectly |
| 22:35.24 | MikeJ[Laptop] | who likes slashdot! 866-387-9249 |
| 22:35.27 | asterisk-dud | it's just that I want to start with channel 11 instead of 1 |
| 22:35.35 | MikeJ[Laptop] | fun new toys |
| 22:35.44 | CunningPike | asterisk-dud: OK - we have a PRI direct tot the PSTN, so I'm not the right person to ask |
| 22:35.50 | *** join/#asterisk anthm (n=anthm@h460852d6.area4.spcsdns.net) |
| 22:35.50 | *** mode/#asterisk [+o anthm] by ChanServ |
| 22:35.53 | CunningPike | s/ask/answer!/ |
| 22:36.22 | MikeJ[Laptop] | or... sip 556@@208.64.200.42 |
| 22:36.25 | *** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it) |
| 22:36.31 | asterisk-dud | jbot: are you the right person? |
| 22:36.48 | asterisk-dud | i din't think thats the issue, the issue is with asterisk and the zap channels |
| 22:36.51 | Dr-Linux | asterisk-dud: yes he is |
| 22:36.55 | drray | jbot is righgt as rain |
| 22:37.08 | asterisk-dud | jbot: http://pastebin.com/768539 |
| 22:37.10 | MikeJ[Laptop] | ~slashdot |
| 22:37.12 | symmetre | is it PRI or PSTN ? are these two different ways to get a call out to the telephone system? |
| 22:37.12 | asterisk-dud | can u look at that? |
| 22:37.23 | asterisk-dud | PRI |
| 22:37.37 | CunningPike | ~pstn |
| 22:37.39 | jbot | i heard pstn is Pubic Switched Telephone Network, or "please stop the nonsense" |
| 22:37.42 | *** join/#asterisk JINDAL (n=root@220.226.36.2) |
| 22:37.44 | drray | ~pinkworld |
| 22:37.46 | CunningPike | ~pri |
| 22:37.48 | jbot | i heard pri is Primary Rate Interface, often called T1 or E1 (European Standard). E1 offers 30 ISDN B-Channels a 64kBit/s + 1 D-Channel with 64kBit/s. The T1 has 23 B-Channels + 1 D-Channel. Cards to use with *: T100P, E100P, TE410P, etc. |
| 22:37.48 | *** part/#asterisk JINDAL (n=root@220.226.36.2) |
| 22:38.03 | asterisk-dud | i am usingPRI |
| 22:38.10 | symmetre | yeah i read about pri on wiki, that info is pasted from there... |
| 22:38.17 | drray | heh |
| 22:38.29 | CunningPike | symmetre: So, a PRI is one of the ways to connect to the PSTN |
| 22:38.33 | symmetre | i just wonder, if this is a choice,... do you select pri or pstn.. or they 2 different tools for different things? |
| 22:38.51 | asterisk-dud | jbot: did u take a look? |
| 22:38.55 | *** join/#asterisk gambolputty (n=gambolpu@cblmdm72-240-85-84.buckeyecom.net) |
| 22:39.16 | Dr-Linux | lol |
| 22:39.32 | Dr-Linux | ~jbot |
| 22:39.34 | jbot | jbot is probably only marginally useful at best, He got a C- on his Turing Test, or a complete idiot |
| 22:39.51 | *** join/#asterisk kFuQ (n=somedude@c-67-185-114-199.hsd1.wa.comcast.net) |
| 22:39.55 | *** join/#asterisk gmaruzz (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 22:39.59 | asterisk-dud | whatever |
| 22:40.09 | *** join/#asterisk znoG (n=gs@153-129-89-200.fibertel.com.ar) |
| 22:40.11 | tzafrir | asterisk-dud, if you wantr channels to start from 11, you better have something to create channels 1-10 for you . Is there any other zaptel hardware on that system? |
| 22:40.22 | fholmes | ~jbot |
| 22:40.24 | jbot | rumour has it, jbot is only marginally useful at best, He got a C- on his Turing Test, or a complete idiot |
| 22:40.24 | tzafrir | Could you pastebin your zaptel.conf? |
| 22:40.29 | znoG | does anyone use/run/own/whatever a RT31P2 unit? |
| 22:40.37 | asterisk-dud | tzafrir: http://pastebin.com/768539 |
| 22:40.44 | asterisk-dud | that is the porbole |
| 22:40.46 | asterisk-dud | problem |
| 22:40.59 | asterisk-dud | can i create 10 dummy channesl |
| 22:41.15 | tzafrir | that is /etc/asterisk/zapata.conf . I meant /etc/zaptel.conf |
| 22:41.31 | asterisk-dud | sorry bud |
| 22:41.45 | asterisk-dud | zaptel.conf is on the end |
| 22:41.46 | asterisk-dud | of that |
| 22:41.58 | tzafrir | Anyway, you can also create that offset in e.g. the dialplan |
| 22:42.26 | asterisk-dud | i know, but i plan to add channels that would go into 1-10, |
| 22:42.34 | asterisk-dud | so i want to leave them blank |
| 22:42.54 | ManxPower | asterisk-dud, you can't do that |
| 22:43.18 | asterisk-dud | so i can't create a dummy channel? |
| 22:43.32 | ManxPower | asterisk-dud, channel numbers are not "things you think up" they are tied to the hardware. |
| 22:43.41 | tzafrir | Or patch zaptel to begin from 10? :-( |
| 22:43.42 | MikeJ[Laptop] | asterisk-dud, what ManxPower said |
| 22:43.52 | drray | why do you want skip the first 10 anyway? |
| 22:43.52 | ManxPower | Channels start at 1, which is the first channel that is found. |
| 22:44.04 | tzafrir | They are not tied to hardware. They are just in the order of registration |
| 22:44.04 | MikeJ[Laptop] | there has been talk to change that.. but you can't right now without changing code |
| 22:44.06 | asterisk-dud | it's more convenient |
| 22:44.21 | asterisk-dud | for the system i'm setting up |
| 22:44.33 | ManxPower | asterisk-dud, I guess it sucks to be you. |
| 22:44.37 | asterisk-dud | but i'll just use the math function to |
| 22:44.43 | drray | my outbound PRI is 71-83 |
| 22:44.44 | asterisk-dud | yuop |
| 22:44.50 | drray | you just hard code it |
| 22:44.55 | tzafrir | So how can I create "dummy" channels? A span that will just register 10 channels and do nothing more? |
| 22:45.15 | tzafrir | It's not complicated to write |
| 22:45.22 | asterisk-dud | excellent question |
| 22:45.30 | *** join/#asterisk jeffik (n=Jeff@kns221.NetSurf.Net) |
| 22:45.47 | asterisk-dud | well, u're the expert, can u guide me in the right direction |
| 22:46.01 | ManxPower | asterisk-dud, If you fight Asterisk you will be unhappy. If you take a Zen-like approach and accept Asterisk's oddities, your life will be much better. |
| 22:46.10 | asterisk-dud | ok |
| 22:46.16 | asterisk-dud | thanks' for the advice, |
| 22:46.20 | symmetre | if you set up a 'farm' of asterisk boxes.. would you have failover security? |
| 22:46.22 | asterboy | bye |
| 22:46.28 | tzafrir | how about ztd-local? Can it be of some help? |
| 22:46.34 | *** join/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 22:46.37 | ManxPower | asterisk-dud, Well, first start out by learning C. Then, after learning C, study the zaptel code. |
| 22:46.39 | tzafrir | I just never used it |
| 22:46.43 | symmetre | asterisk-dud linux systems will hurt you |
| 22:46.44 | asterisk-dud | what is ztdummy? |
| 22:46.54 | *** join/#asterisk MikeJ__ (n=vircuser@d14-69-8-30.try.wideopenwest.com) |
| 22:47.05 | ManxPower | asterisk-dud, it's a driver to provide timing without having actual channels |
| 22:47.08 | tzafrir | ztdummy creates just a span with no channels. For timing from the system clock |
| 22:47.17 | asterisk-dud | ok |
| 22:48.27 | CunningPike | ~ztdummy |
| 22:48.28 | jbot | ztdummy is, like, a driver that interacts with zaptel to provide a timing source to Asterisk. On 2.4.x kernals, timing is obtained from a UHCI USB controller. It will not work with OHCI controllers. On 2.6.0 and later kernels, the timing is provided by the kernel, thus no hardware is required at all. |
| 22:48.42 | ManxPower | asterisk-dud, Oh, BTW, DON'T PUT QUOTES IN CALLERID |
| 22:51.25 | CunningPike | symmetre: What we have done is to have two identical servers, configured exactly the same. If the first one fails, our Polycoms use DNS SRV to fail over to the second server. We physically move the PRIs from one server to the other |
| 22:53.06 | Dr-Linux | CunningPike: what if DNS is working fine but something things fails? |
| 22:53.21 | CunningPike | Dr-Linux: Like? |
| 22:54.13 | Dr-Linux | CunningPike: asterisk service? email server or apache? |
| 22:54.21 | symmetre | ok let me ask this way: what are all the different ways to connect asterisk to the pstn (i.e. PRI is one such way)? |
| 22:55.02 | CunningPike | Dr-Linux: If the Polycoms can't register with the main server, they will fail over to the other server |
| 22:55.39 | CunningPike | symmetre: Usually, with a PRI, a channelbank, or a FXO interface |
| 22:55.48 | Dr-Linux | CunningPike: what tool you are using to identify this failover? |
| 22:56.08 | CunningPike | Dr-Linux: We're not - the Polycoms know if they've lost their registration |
| 22:56.15 | symmetre | channelbank and fxo are both pieces of hardware arent they? |
| 22:56.36 | CunningPike | Dr-Linux: Plus we use Intermapper with a modified version of the Nagios plugin for monitoring |
| 22:56.46 | Dr-Linux | CunningPike: only polycom does that or other softphones etc as well? |
| 22:57.03 | CunningPike | symmetre: Yes - you need some hardware to connect - or a service provider |
| 22:57.05 | Dr-Linux | we are using nagios as well |
| 22:57.51 | CunningPike | Dr-Linux: I think some of the other hardphones use DNS SRV too - I don't know of any softphones that do |
| 22:58.32 | CunningPike | Dr-Linux: Right now, we are just using an IAX ping, but I see there is a Nagios plugin for zaptel that I would like to try |
| 22:58.33 | h0 | hello guys I am new to Asterisk and want to start experimenting with it in a home environment. Would you guys recommend the SPA-3000 as a analog telephone adapter and is there any other hardware that I will need to get started other then a linux server |
| 22:59.19 | CunningPike | h0: The SPA-3000 is a good ATA and all you will need |
| 22:59.22 | *** join/#asterisk shmaltz (n=mybox@mail.dmaven.com) |
| 22:59.22 | Dr-Linux | CunningPike: what does that nagios zaptel plugin? |
| 22:59.34 | CunningPike | Just saw it on the list |
| 22:59.40 | CunningPike | ~thelist |
| 22:59.41 | jbot | somebody said thelist was the asterisk-users mailing list. Sign up or view archives at http://lists.digium.com/mailman/listinfo/asterisk-users |
| 22:59.44 | CunningPike | ;) |
| 22:59.45 | asterisk-dud | whats wrong with this: Set(${EXT}=${MATH($EXTEN}-2010)}) |
| 23:00.04 | shmaltz | anybody here heard of voipteck? |
| 23:02.04 | *** join/#asterisk anthm (n=anthm@adsl-68-248-236-217.dsl.milwwi.ameritech.net) |
| 23:02.04 | *** mode/#asterisk [+o anthm] by ChanServ |
| 23:02.13 | CunningPike | Dr-Linux: Look for a posting entitled 'asterisk nagios plugin' from yesterday |
| 23:03.24 | asterisk-dud | whats wrong with this: Set(${EXT}=${MATH($EXTEN}-2010)}) |
| 23:03.49 | *** part/#asterisk jarg (n=jarg@200.56.225.61) |
| 23:05.19 | CunningPike | asterisk-dud: Nothing, unless you're getting an error |
| 23:05.53 | anthm | omit $ in the ${EXT} |
| 23:06.44 | Shaun2222 | Jun 9 07:03:04 ERROR[31489]: pbx.c:1380 ast_func_read: Function AGENT not registered |
| 23:06.54 | Shaun2222 | anybody know what module i need loaded to use the AGENT function? |
| 23:07.02 | *** join/#asterisk Bullseye_Network (n=Kyle@216.143.192.69) |
| 23:07.21 | Bullseye_Network | Quick question if I didnt install SOX would that cause voicemails to not record? |
| 23:09.12 | asterisk-dud | the prob was i forgot the { before EXTEN |
| 23:09.17 | asterisk-dud | iam a dumbass |
| 23:11.05 | anthm | unless they changed it Set(${EXTEN}=something) will make a new var that has the name of what ${EXTEN} evals to |
| 23:13.13 | *** part/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 23:14.13 | r_evolution | CHYA!!! |
| 23:14.24 | anthm | Bullseye, no |
| 23:14.36 | anthm | aww he'll never know now |
| 23:14.44 | r_evolution | :( |
| 23:14.53 | r_evolution | i kinda figured it'd be a lil one-way audio issue |
| 23:18.09 | *** join/#asterisk generalhan (i=general_@ip67-90-64-2.z64-90-67.customer.algx.net) |
| 23:18.13 | generalhan | Whats up all |
| 23:19.02 | generalhan | has anyone ever had any issues with a 7960 and the message light staying on even when there are no messages ??? i need some possible causes for this cause its starting to make my boss angry ... lol |
| 23:19.19 | generalhan | he sees red and it makes him mad ! hahaha ! |
| 23:20.23 | *** join/#asterisk gmaruz1 (n=gmaruzz@217-133-80-112.b2b.tiscali.it) |
| 23:20.31 | anthm | write a cron to send him a vmail saying "you have no new messages just letting you know" constantly |
| 23:20.32 | *** part/#asterisk h0 (n=h0@ool-44c69453.dyn.optonline.net) |
| 23:22.09 | asterisk-dud | is there a way to cancel absolute timout after it's been activated |
| 23:23.39 | asterisk-dud | yes, reset it to zero :D |
| 23:23.56 | luke-jr_ | ... |
| 23:24.55 | *** join/#asterisk chaoscon (n=ph33r@smartserv/ceo/chaoscon) |
| 23:25.34 | generalhan | anthm: thats good .. but still doesnt get rid of the red light ! lol |
| 23:27.10 | *** join/#asterisk litage (n=nick@203.220.55.70) |
| 23:28.44 | *** join/#asterisk justinu|laptop (n=Justin@dsl081-083-093.lax1.dsl.speakeasy.net) |
| 23:29.33 | asterisk-dud | whats a good soundcard to use with asterisk |
| 23:30.16 | tzafrir | Once again: anybody here uses res_zeroconf? |
| 23:30.18 | De_Mon | huh? |
| 23:30.35 | De_Mon | asterisk-dud huh? |
| 23:30.47 | CunningPike | asterisk-dud: asterisk doesn't need a soundcard, per se..... |
| 23:30.52 | tzafrir | asterisk-dud, chan_alsa |
| 23:31.02 | De_Mon | chan_alsa isn't a sound card! |
| 23:31.09 | tzafrir | Not really important, though |
| 23:31.33 | tzafrir | Ah, what make? |
| 23:31.51 | tzafrir | I figure that any full-duplex card will do. How can you tell that? |
| 23:32.19 | tzafrir | my laptop's sound card isn't :-( |
| 23:32.30 | De_Mon | why do you need full duplex? |
| 23:32.51 | CunningPike | Why do you need a soundcard? |
| 23:33.17 | De_Mon | CunningPike console audio |
| 23:33.44 | CunningPike | De_Mon: Oh |
| 23:33.57 | asterisk-dud | CunningPike: Public Address System |
| 23:34.07 | CunningPike | asterisk-dud: Ah |
| 23:34.38 | De_Mon | hey.. thats not a bad idea |
| 23:35.14 | asterisk-dud | i'm going to try SB Live |
| 23:37.08 | *** join/#asterisk mogorman (i=ejabberd@68.62.237.103) |
| 23:41.05 | *** join/#asterisk SplasPood (n=jwb@ool-18b93b84.dyn.optonline.net) |
| 23:43.25 | *** join/#asterisk Dr-Linux (n=Linux@202.59.73.131) |
| 23:44.09 | *** join/#asterisk ryguillian (n=ryanhaye@c-24-12-98-36.hsd1.il.comcast.net) |
| 23:44.14 | ryguillian | clear |
| 23:44.29 | *** join/#asterisk JINDAL (n=root@220.226.36.2) |
| 23:46.22 | *** join/#asterisk Snake-Eyes (n=blog@203.220.55.70) |
| 23:49.24 | *** join/#asterisk hads (n=hads@mail.nice.net.nz) |
| 23:49.38 | *** join/#asterisk ToTo (n=ToTo@host68-166.pool879.interbusiness.it) |
| 23:53.37 | Shaun2222 | is there a way to reset the ${EXTEN} ? |
| 23:53.55 | Shaun2222 | i tryed Set(EXTEN=1949${EXTEN}) |
| 23:54.01 | Shaun2222 | but it doesnt appear to be working |