| 00:02.35 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 00:02.58 | *** join/#asterisk JCMaxwell_ (~bradleyd@rrcs-24-73-194-54.se.biz.rr.com) |
| 00:03.57 | JCMaxwell_ | has anyone seen multiple dtmf packets per digit in a pcap capture? |
| 00:04.20 | JCMaxwell_ | I am getting about 20 packets per digit pressed going through Level3 |
| 00:04.36 | *** join/#asterisk CentroniX (~cent@cpe-72-179-37-219.austin.res.rr.com) |
| 00:13.28 | *** join/#asterisk wonderworld (~ww@port-92-201-27-39.dynamic.qsc.de) |
| 00:19.14 | *** join/#asterisk Praise (~Fat@unaffiliated/praise) |
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| 00:28.11 | *** join/#asterisk sskyles (~sskyles@60.105.91.184.cfl.res.rr.com) |
| 00:37.35 | paulc | JCMaxwell_: I have to run in a sec, but yes - I've seen similar. And although we use Level 3, I don't think it was with them - this was probably PSTN --> Asterisk --> Another Asterisk directly, as well as PSTN --> Asterisk --> Windows SIP recorder --> Another Asterisk. Specifically, a number of packets - I think it was something like "sending tone, sending tone, sending tone, tone done" kind of thing. We had an issue with DTMFs not flowing through the ca |
| 00:44.41 | p3nguin | paulc: You need to get a message split tool... you truncated at "flowing through the ca" |
| 00:46.14 | paulc | repost: We had an issue with DTMFs not flowing through the call recorder right because they didn't pass on all the packets. |
| 00:46.27 | paulc | irssi seemed happy enough with it - sorry about that |
| 00:46.37 | paulc | and on that note - my chariot awaits! I'm off home :-) |
| 00:53.48 | *** join/#asterisk vinhdizzo (~vinh@pool-173-51-129-161.lsanca.fios.verizon.net) |
| 00:57.47 | *** part/#asterisk Blue-Dragon (~asdf@dbeuchert.com) |
| 01:05.41 | *** join/#asterisk shapr (~shapr@nat/digium/x-dkhuddhgyuzxiakl) |
| 01:08.19 | shapr | Hody |
| 01:08.21 | shapr | Howdy |
| 01:08.36 | shapr | Is there some particular syntax to put multiple allow=codec lines on a single line? |
| 01:08.40 | shapr | Like, ampersand or something? |
| 01:13.55 | carrar | se a template? |
| 01:13.57 | carrar | use |
| 01:14.36 | carrar | the apply that template to your sip entry |
| 01:14.43 | *** join/#asterisk drivefast (~radu@adsl-99-92-126-154.dsl.lsan03.sbcglobal.net) |
| 01:14.55 | carrar | read the sip.conf example |
| 01:15.06 | carrar | there is a example of that in there |
| 01:15.10 | shapr | carrar: Oh excellent point, thanks! |
| 01:15.33 | carrar | search "my-codecs" |
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| 01:22.10 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
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| 02:26.04 | *** part/#asterisk sjobeck (~Adium@valdisere.sjobeck.com) |
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| 02:55.03 | BeeBuu | how can i disable the SIP multiline ? |
| 03:01.02 | ChannelZ | I assume you mean call waiting |
| 03:01.19 | ChannelZ | in which case that's a function of your phone, not Asterisk |
| 03:11.01 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 03:12.25 | *** join/#asterisk puzzled (~patrick@puzzled.xs4all.nl) |
| 03:14.54 | p3nguin | paulc: You need to use the splitlong.pl script to make it split the long posts so it doesn't truncate. |
| 03:16.37 | BeeBuu | ChannelZ: thanks. |
| 03:17.38 | BeeBuu | another question:i found the billsec is 0 even i answered the call in asterisk 1.6.2.5,what's problem? |
| 03:26.55 | *** join/#asterisk kerframil (~kerframil@gentoo/user/kerframil) |
| 03:34.55 | *** join/#asterisk xpot-mobile (~james@166-70-100-198.ip.xmission.com) |
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| 04:36.59 | *** part/#asterisk benngard (~mabe@h30n2-g-ml-a11.ias.bredband.telia.com) |
| 04:39.52 | *** join/#asterisk sawgood (~sawgood@173-13-158-27-sfba.hfc.comcastbusiness.net) |
| 04:48.53 | *** join/#asterisk JerJer (~JerJer@asterisk/original-h323-guy/JerJer) |
| 04:50.54 | sawgood | anyone skilled with Sangoma by chance? |
| 04:51.07 | sawgood | I have a A101 PRI (provisioned,active,up) |
| 04:51.15 | sawgood | I can make outbound calls, but I cannot receive incoming calls |
| 04:51.34 | sawgood | the incoming call reaches the * box, but an all circuits are busy message plays |
| 04:52.00 | *** join/#asterisk dobah (~dobah@cpe-98-150-177-51.hawaii.res.rr.com) |
| 04:52.43 | devdvd | sawgood, i dont know anything about sangoma but does your console say something like "unable to create channel of type" |
| 04:52.53 | devdvd | whats the debug and verbosity level on your console? |
| 04:53.13 | sawgood | verbose = 5 |
| 04:53.18 | sawgood | debug = 0 |
| 04:53.24 | *** join/#asterisk sjobeck (~sjobeck@valdisere.sjobeck.com) |
| 04:53.34 | devdvd | core set verbose 10 |
| 04:53.37 | devdvd | core set debug 10 |
| 04:53.40 | devdvd | then try a call |
| 04:53.58 | sawgood | doing it now |
| 04:54.18 | devdvd | and pastebin what you see, i probably can't help you but that output will help others who are smarter than I possibly be able to help you |
| 05:01.57 | *** join/#asterisk chilicuil (~chilicuil@unaffiliated/chilicuil) |
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| 05:18.43 | sawgood | besides udp on port 4569, are any other ports needed (firewall concerns)? |
| 05:18.49 | sawgood | for iax2 |
| 05:20.36 | *** join/#asterisk benngard (~mabe@213.88.138.230) |
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| 05:41.47 | kaldemar | sawgood: no. |
| 05:44.36 | sawgood | kaldemar: thank you! |
| 05:56.55 | *** join/#asterisk codefreeze-lap (~Steve_Mur@nv-69-68-103-77.sta.embarqhsd.net) |
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| 06:16.26 | *** join/#asterisk joobie (~joobie@unaffiliated/moo0o0ooo00o0o0o) |
| 06:16.35 | joobie | hey guys.. how can i see what version sip my asterisk supports? |
| 06:18.45 | shapr | ? |
| 06:18.52 | kaldemar | joobie: sip version as in 1.0/2.0? |
| 06:20.45 | joobie | kaldemar, ya |
| 06:20.58 | joobie | supposedly the ip 7000 polycom requires v3.0.2 |
| 06:21.03 | joobie | not sure if my asterisk box supports this |
| 06:23.11 | kaldemar | if you find SIP v3.0.2 somewhere, let us know. that must mean something else. |
| 06:24.58 | kaldemar | that's the polycom firmware version, not SIP protocol version. |
| 06:26.31 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 06:33.21 | joobie | ahh thanks |
| 06:40.06 | *** join/#asterisk luisfelice (~luisfelic@190.39.198.95) |
| 06:41.22 | luisfelice | Hi, I am having a weird noise on the FXS ports of a TDM400P card, I believe it is the power source, is it possible to install a filter? |
| 06:42.48 | shapr | If it's power, then your computer's power supply is the problem. |
| 06:42.57 | shapr | Is it cheap to put in a better PSU? |
| 06:43.55 | luisfelice | It is a DELL server |
| 06:44.33 | shapr | luisfelice: alternatively, you could put the card in a different computer and see if the noise is still there |
| 06:44.52 | luisfelice | ok thanks I will try |
| 06:59.39 | *** join/#asterisk Tim_Toady (~moi@188.4.36.223) |
| 07:01.53 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
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| 07:31.40 | *** join/#asterisk schmidts (~schmidts@Lo20-rt01.lm33.sil.at) |
| 07:31.42 | schmidts | good morning |
| 07:33.12 | shapr | howdy schmidts |
| 07:34.31 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:34.38 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
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| 07:54.23 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 07:55.37 | zkn | Hello, are there any australians here who could give me advice on local ITSPs ? |
| 07:56.19 | *** join/#asterisk hehol (~hehol@2001:1438:1009:200:512:8271:8b76:433) |
| 08:11.38 | *** join/#asterisk waschtl (~waschtl@3ed8a58a.d.d9tcloud.de) |
| 08:16.06 | Corydon76-home | zkn: email the -biz list |
| 08:16.27 | zkn | thanks, will try there |
| 08:18.08 | *** join/#asterisk jploh (~jploh@124.106.230.73) |
| 08:19.01 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 08:21.21 | *** part/#asterisk zkn (~zkn@195.222.14.202) |
| 08:21.45 | jploh | does anyone know if skype for asterisk works on asterisk 1.6.2 32-bit? |
| 08:22.06 | shapr | it should |
| 08:22.31 | shapr | Digium has a binary for 1.6.2 32-bit |
| 08:22.38 | *** join/#asterisk gr0mit (~tim@extrt.txrx.org.uk) |
| 08:24.21 | *** join/#asterisk tamiel (~tamiel@213.30.183.226) |
| 08:28.20 | *** join/#asterisk zkn (~zkn@195.222.14.202) |
| 08:28.42 | zkn | oh boy, the -biz list looks deserted |
| 08:33.14 | schmidts | zkn not really, there was 15 mails in the last week ;) |
| 08:41.05 | Corydon76-home | 32-bit is almost never the problem; it's whether vendors have binaries for 64-bit |
| 08:42.27 | *** join/#asterisk ketas (~ketas@ketas6-sixxs.si.pri.ee) |
| 08:47.53 | zkn | i'll try and keep my eyes peeled there |
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| 09:09.46 | *** join/#asterisk asterisk-learner (~chatzilla@77.42.241.114) |
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| 09:37.22 | asterisk-learner | hi, are the steps fin here : http://www.voip-info.org/wiki/view/Asterisk+debugging still accurate or outdated ? |
| 09:39.53 | *** join/#asterisk sgimeno (~chatzilla@163.117.206.10) |
| 09:40.40 | schmidts | asterisk-learner i just take a short look and most of them are still accurate but its outdated |
| 09:41.16 | schmidts | asterisk-learner maybe you should take a look here: https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information |
| 09:41.48 | kaldemar | the voip-info article is a mess by terminology also. |
| 09:42.54 | kaldemar | and contains information that has never applied. |
| 09:43.09 | asterisk-learner | the thing is tht i am doing the setps under : HowTo Debug a DeadLock in Asterisk |
| 09:43.20 | asterisk-learner | and reached point 6) Try to identify the first thread, that is dead locked. |
| 09:44.10 | asterisk-learner | they are sating that i should have a display like this : Thread 23 (Thread 3576854 (LWP 2910))" |
| 09:45.03 | asterisk-learner | but i am getting : Thread 2 (Thread 0x7fbe7940 (LWP 19443)) |
| 09:45.30 | asterisk-learner | and converting the hex to decimal will lead to a number in the range of 2 billions |
| 09:45.47 | asterisk-learner | which is surely not the thread id |
| 09:49.15 | schmidts | do you have a dump file where you use gdb or do you only see the core show locks output? |
| 09:52.10 | asterisk-learner | no i am only seeing the core show locks output (asterisk is still running , no crash ) |
| 09:53.19 | *** join/#asterisk DrDigital (~mmurphy@gallery/DrDigital) |
| 09:58.09 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 10:00.27 | *** join/#asterisk TimeRider (steve@188-220-33-123.zone11.bethere.co.uk) |
| 10:02.08 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 10:02.15 | jmls | hey |
| 10:03.01 | jmls | looking for a speech to text system that would take either a live call or recorded call and convert it into a text file |
| 10:03.15 | jmls | anyone know of something that can do that from an asterisk system ? |
| 10:05.34 | schmidts | asterisk-learner is this a production system or just testing? |
| 10:06.45 | asterisk-learner | schimdts: just testing, i have a lock somewhere in my application and i was just following the steps successfully until this point 6 |
| 10:06.56 | asterisk-learner | using asterisk 1.4.36 on a 64 bit machine |
| 10:20.43 | shapr | jmls: there's some sort of binding to cmu sphinx |
| 10:21.53 | jmls | ta |
| 10:28.29 | *** join/#asterisk coppice (~chatzilla@m121-202-106-16.smartone-vodafone.com) |
| 10:31.20 | asterisk-learner | ... |
| 10:33.59 | shapr | asterisk-learner: Do you have the same problem with 1.8 L |
| 10:34.00 | shapr | ? |
| 10:35.24 | asterisk-learner | i dont know i didn't try it, but i thought this is related to gdb more, no ? |
| 10:35.36 | schmidts | asterisk-learner sorry i am busy today, search on voip-info for the usage of gdb you can connect to a running asterisk process and get the output of all threads, there you will find the match to your locking thread |
| 10:35.36 | shapr | shrugs |
| 10:36.04 | shapr | Digium is soon to drop support for Asterisk 1.4 and 1.6, leaving 1.8 as the only Digium-supported version. |
| 10:36.12 | shapr | Thus it could be handy to try Asterisk 1.8. |
| 10:37.20 | asterisk-learner | schimdts: this is what i am doing already, but np if ur busy |
| 10:37.22 | asterisk-learner | thx anw |
| 10:37.53 | shapr | Does anyone know which versions of trixbox use which versions of Asterisk? |
| 10:38.23 | *** join/#asterisk mintos (~mvaliyav@nat/redhat/x-jtkxggogfvlpptnq) |
| 10:38.46 | schmidts | asterisk-learner do you allready have an output of gdb? if yes just pastebin and i will take a look |
| 10:47.04 | *** join/#asterisk salimb (~chatzilla@83-244-177-2.cust-83.exponential-e.net) |
| 10:52.08 | asterisk-learner | schimdts: I can't for now, but thx again for your help |
| 10:53.48 | schmidts | ok ;) |
| 10:54.21 | *** join/#asterisk Sertys (~sertys@89.252.247.42) |
| 10:54.22 | gr0mit | hi - does anyone know how to set the k-break detection in dahdi on FXO ports? UK seems to have a short k-break of approx 100ms and i am not sure how check if this is too short or not |
| 10:54.25 | asterisk-learner | schimdts: :-) |
| 10:57.02 | coppice | is a k-break anything like a t-break? |
| 10:57.18 | *** join/#asterisk rdahlin_1 (~rdahlin_1@2001:16d8:cc97:1:21f:5bff:fe37:c2c9) |
| 11:01.25 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 11:08.42 | gr0mit | coppice, sigh. |
| 11:11.07 | coppice | yeah, i guess 100ms is a bit short for a uk t-break |
| 11:11.32 | gr0mit | :-) |
| 11:19.46 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 11:21.17 | coppice | gr0mit: if by k-break you mean power break, I think that is defined as 100ms in the UK |
| 11:22.54 | gr0mit | yes, exactly. |
| 11:23.05 | gr0mit | but where in dahdi does it detect it? |
| 11:23.12 | gr0mit | and will it detect 100ms? |
| 11:23.20 | gr0mit | thinks it won't |
| 11:25.31 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 11:28.06 | coppice | its pretty useless if it can't. 100ms is quite normal |
| 11:43.25 | *** join/#asterisk Poincare (~jefffnode@2001:5c0:150f:1704::2) |
| 11:48.24 | shapr | gr0mit: It's defined in zonedata.c |
| 11:48.54 | gr0mit | ok, let me take a peek |
| 11:50.36 | shapr | gr0mit: Different countries have different times for the various signals, so zonedata.c holds the various values. |
| 11:50.51 | benngard | when u execute a System() in the dial plan, shoudnt the dialplan continue? |
| 11:51.08 | shapr | gr0mit: In my vaguely recent version of the dahdi sources, UK settings are on line 128 |
| 11:51.15 | gr0mit | is this in dahdi or asterisk? |
| 11:51.17 | shapr | dahdi |
| 11:51.29 | kaldemar | benngard: when the application executed by System() returns, yes. |
| 11:51.41 | shapr | Are you asking about signalling rather than busy/hangup/etc detection? |
| 11:51.55 | benngard | yes, i thought so to |
| 11:51.56 | *** join/#asterisk lauris (~la@unaffiliated/lauris) |
| 11:51.58 | gr0mit | yes, signalling |
| 11:52.37 | lauris | hi, anyone from Finland ? i'm wondering are there any local SIP providers based in Helsinki ? |
| 11:53.42 | *** join/#asterisk jmkgreen (~chatzilla@wish-hq3.gotadsl.co.uk) |
| 11:54.12 | *** join/#asterisk fskrotzki_ (~fskrotzki@cpe-74-74-245-250.rochester.res.rr.com) |
| 11:54.29 | jmkgreen | anyone running asterisk within an esxi host? Got Ubuntu running with dahdi_dummy but am setting 99.5% accuracy which is baffling me |
| 11:55.03 | jmkgreen | the host has intel's virtualisation chip features which are bios-enabled |
| 11:55.26 | jmkgreen | I'm currently stuck and google doesn't help the answer that I've seen :-( |
| 11:55.29 | gr0mit | shapr, this file does not define the DC line state, only tones |
| 11:56.58 | kaldemar | lauris: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Residential#Finland |
| 11:58.22 | kaldemar | lauris: http://www.voip-info.org/wiki/view/VOIP+Service+Providers+Business+Europe#Finland |
| 11:58.30 | benngard | http://pastebin.com/rynjkXW0 <- can some1 explain for me why "tiff2pdf" is executed but nor "mutt"? |
| 11:58.37 | benngard | not* |
| 12:01.14 | lauris | kaldemar, thank you |
| 12:04.21 | *** join/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 12:08.57 | kaldemar | benngard: the first priority following the tiff2pdf is not executed either. does the command really exit properly? |
| 12:14.52 | *** join/#asterisk ariel_ (~chatzilla@63.214.236.169) |
| 12:14.52 | *** join/#asterisk volga629 (~slava@host7.pythian.com) |
| 12:16.13 | volga629 | <PROTECTED> |
| 12:16.23 | volga629 | what is mean ? |
| 12:16.31 | volga629 | thnak you in advance |
| 12:16.54 | benngard | kaldemar: tiff2pdf creates the pdf, how can i check that it is exciting properly? |
| 12:17.10 | kaldemar | volga629: it means to go to extension s-CONGESTION and priority 1 in context macro-callexception. |
| 12:18.29 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 12:18.52 | volga629 | so it should drop the call ? |
| 12:20.04 | kaldemar | volga629: no, it should continue dialplan execution at s-CONGESTION,1. it's up to your dialplan to decide what that extension does. |
| 12:20.45 | volga629 | can I put in paste bin some part of log I just looking for explanation for it |
| 12:21.04 | kaldemar | sure, go ahead. |
| 12:21.51 | volga629 | http://pastebin.com/cK7fWfZD |
| 12:25.43 | volga629 | I found this error in log this morning |
| 12:25.47 | volga629 | [Mar 16 08:18:12] WARNING[17199]: app_dial.c:1196 dial_exec_full: Unable to create channel of type 'SIP' (cause 3 - No route to destination) |
| 12:26.45 | kaldemar | volga629: your call to link2voip-sw2/001161416370127 fails for some reason, macro-sipfailover executes macro-callexception with arguments CONGESTION and 34, and macro-callexception goes to macro-callexception,s-CONGESTION,1. that's about it. |
| 12:27.40 | volga629 | it possible routing problem or just sip provider ? |
| 12:28.10 | kaldemar | volga629: "no route to destination" means that your asterisk does not know how to reach the peer. either they haven't registered to you (in case of host=dynamic), a host doesn't resolve to an ip address or your system has no route for the configured ip address. |
| 12:28.56 | *** join/#asterisk Mw3_ (mw3@mw3.hu) |
| 12:29.31 | volga629 | kaldemar: thank you on help |
| 12:29.47 | kaldemar | possibly a routing problem or a provider problem. providers don't usually register to clients, so most likely there's an issue in your configuration. see sip.conf for [link2voip-sw2]. |
| 12:30.02 | *** join/#asterisk Bidik (~bidik@li267-109.members.linode.com) |
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| 12:50.10 | moos3 | anyone have experience with queues and making so a button press drops you to voicemail |
| 12:50.24 | moos3 | some reason i can't get it work for my queue |
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| 12:57.47 | leifmadsen | moos3: ;context = qoutcon <-- queues.conf |
| 12:58.02 | leifmadsen | then put in the dtmf digit (single digit) into that context |
| 12:59.09 | moos3 | http://pastebin.com/S8mAak77 |
| 12:59.14 | moos3 | thats what i have |
| 12:59.30 | moos3 | works on all the other queues but doesn't work on this one for some reason |
| 12:59.41 | moos3 | so you can hit 1 until your blue in the face |
| 12:59.52 | moos3 | and never get to voicemail |
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| 13:03.00 | leifmadsen | moos3: is that the only queue with tT set? |
| 13:03.09 | *** join/#asterisk jaytee (~jforde051@unaffiliated/jaytee) |
| 13:03.40 | moos3 | nah they are all the same |
| 13:04.28 | leifmadsen | hmmm |
| 13:04.36 | leifmadsen | I was thinking maybe something was eating the DTMF |
| 13:05.07 | leifmadsen | probably have to see the queues.conf config and the dialplan referenced by 'context' as well then |
| 13:05.18 | leifmadsen | sorry, gotta run off as I'm finishing the last recipe for the Asterisk Cookbook |
| 13:05.25 | moos3 | k |
| 13:05.26 | *** join/#asterisk imox1234 (~imox1234@p4FC5C771.dip0.t-ipconnect.de) |
| 13:05.47 | moos3 | leifmadsen hit me up when your done |
| 13:05.54 | leifmadsen | yep will try -- crazy busy today |
| 13:06.16 | moos3 | cool thanks |
| 13:14.37 | *** join/#asterisk defswork (~andy@mx1.3gcomms.co.uk) |
| 13:14.49 | *** join/#asterisk The_Boy_Wonder (~vossel@asterisk/batman-developer/dvossel) |
| 13:15.14 | defswork | what are openvox pri/bri cards like ? worth using ? they appear cheap in comparison to others |
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| 13:37.42 | n3hxs | defswork, I have the analog version of their card and it works fine for me. Can't say about the PRI card. |
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| 13:57.16 | joelsolanki | Morning friends. have a quick question. if we have LLC in usa and traffic is originating from middle east and terminating in asian countries then do we need FCC 214 License ? |
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| 14:05.57 | *** join/#asterisk nwidger (~nwidger@steerpike.iol.unh.edu) |
| 14:06.32 | nwidger | does anyone here have experience using the 'outboundproxy' variable in sip.conf? |
| 14:07.25 | *** join/#asterisk Victor_Yure (~victor@unaffiliated/victoryure/x-837844) |
| 14:10.08 | nwidger | im running into the problem where outbound calls are correctly forwarded by asterisk to the server set in the 'outboundproxy' variable. however, when an inbound call comes in, the Dial(SIP/xxx) line in extensions.conf causes asterisk to forward the call _back out to my outbound proxy_ instead of to the user that registered to asterisk. |
| 14:11.55 | calhoun | hey having some troubles getting calls to pick up, got 2 sccp phones and when i dial stuff i see in the console the buttons pressed, i think i've got my extensions correct ( http://pastebin.com/QnBgYP9N ) anything look wrong there to you? I'm expecting if I dial 6004/6000 it should play the monkeys sound but just get a stop tone instead |
| 14:12.48 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 14:18.05 | bent_screwdriver | i have some Polycom ip650's that are in a ring group. Occasionally one or more of them continue to ring even after the call has been anwswerd by one of the other phone users. Phones and asterisk pbx are on same switch/lan. Any idea what causes this? thx |
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| 14:20.48 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 14:25.00 | asterisk-learner | is there a way to do the equivalent of ast_set_flag(chan, AST_FLAG_ZOMBIE) from the dialplan ? |
| 14:25.05 | asterisk-learner | (just for testing of course) |
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| 14:39.37 | *** join/#asterisk rcaskey (~rcaskey@dumbledore.athenshousing.org) |
| 14:46.23 | rcaskey | is the voice processing in asterisk typically done via specialized hardware? |
| 14:48.03 | *** join/#asterisk Tech_Travis (~Travis_El@cpe-76-168-191-127.socal.res.rr.com) |
| 14:48.23 | rcaskey | or is it basically agregated, digitized raw, and fed into the computer to be munched |
| 14:51.08 | *** join/#asterisk pallet (~pallet@remote.leaftechnology.co.uk) |
| 14:51.45 | pallet | Hi there, does anyone know if there's an IRC channel for 3CX ? |
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| 14:55.00 | jkprg | Hi. I have multicast audio stream in my network. How can I configure asterisk to allow people to call specific extension to listen that stream? Thx |
| 14:55.31 | jkprg | the stream is mpeg2... |
| 14:56.51 | *** join/#asterisk navis (~user@91.180.136.44) |
| 14:57.44 | navis | hello world |
| 14:57.58 | navis | I have an asterisk install connected to two BRI ISDN lines |
| 14:58.06 | navis | that means a total of four channels |
| 14:58.24 | navis | I'd like to be able to forward calls |
| 14:58.44 | navis | but if a call came from the BRI1, forward it through BRI2 |
| 14:59.23 | navis | and if a call came from BRI2, forward it through BRI1 |
| 14:59.24 | *** join/#asterisk codefreeze-lap (~Steve_Mur@wsip-24-234-181-18.lv.lv.cox.net) |
| 14:59.38 | navis | is this possible ? |
| 15:00.44 | Tozz_ | yes |
| 15:01.10 | navis | Tozz_: thanks but how ? :-) |
| 15:01.33 | Tozz_ | you can identify the BRI number by checking channel variables |
| 15:01.53 | Tozz_ | and based on that you can do something like Dial($BRI1}/1234567890) |
| 15:02.26 | Tozz_ | I think the variable you need is in ${CHANNEL} |
| 15:03.23 | navis | Tozz_: ok, just to be precise, the call has been answered by an inside phone, and I want it to use the other BRI channel if they forward the call |
| 15:03.53 | navis | Tozz_: are the variables still availlable in the new call ? |
| 15:04.39 | navis | Tozz_: so call comes through BRI1, I answer it, I decide to forward it, and asterisk should automatically choose the other channel |
| 15:05.01 | rcaskey | We currently have a proprietary phone system and it works great but I fear the day will come when our vendor will longer support our configuration and we will be forced to upgrade. Also in the forseeable future I can imagine we might want to step up past our bonded t1 to something a bit more zoomy. Can you cut out a bunch of the DSP muckity-muck if you get the data from your fiber provider in the right format? |
| 15:05.15 | tzafrir | navis, what channel do you use for ISDN? |
| 15:05.28 | tzafrir | DAHDI? mISDN? |
| 15:05.33 | tzafrir | CAPI? |
| 15:05.46 | navis | tzafrir: CAPI |
| 15:06.34 | navis | tzafrir: I don't necessarily need the forwarded call to be on the other BRI, but I need to be sure that it has a free channel |
| 15:07.07 | navis | tzafrir: so since I limit incoming calls to 2, the easiest way is to use the other BRI (I guess) |
| 15:08.01 | *** join/#asterisk Defraz (~Defraz@tim.spudnik.com) |
| 15:10.12 | JerJer | hmm... has anyone heard of a 'reverse enum lookup' (for authentication) ? |
| 15:11.50 | asterisk-learner | jerjer: i dont think it exists in C :-( |
| 15:12.03 | JerJer | ENUM DNS |
| 15:12.09 | JerJer | not lame ass java horseshit |
| 15:12.12 | JerJer | :) |
| 15:12.40 | JerJer | java crap is exactly what google is returning |
| 15:12.47 | *** join/#asterisk Freeaqingme| (~dolf@dsl-083-247-011-232.solcon.nl) |
| 15:13.47 | JerJer | i have found a motorolla patent, but hmm |
| 15:14.07 | Freeaqingme| | I have all my sip servers to register to and all phones that can log in (inc. credentials) in my db. What would be the neatest way to connect that to Asterisk? Looking at AMI, but that doesn't seem to support that? |
| 15:14.52 | JerJer | Freeaqingme: run kamailio in front of asterisk |
| 15:15.13 | JerJer | or deal with asterisk realtime |
| 15:15.33 | Freeaqingme| | asterisk realtime < how? |
| 15:15.51 | JerJer | ask google. I refuse to use it myself |
| 15:16.44 | Freeaqingme| | sorry, hadnt realized it was a module. why refuse using it? |
| 15:18.16 | JerJer | in my systems, asterisk has no need to depend on a database to process phone calls |
| 15:20.34 | JerJer | but also, i generally deal with very large systems |
| 15:21.12 | rcaskey | JerJer, are there any recent guides that might be of interest to a 50-100 phone setup? |
| 15:21.24 | Freeaqingme| | JerJer, how's that related to my q? |
| 15:22.03 | JerJer | Freeaqingme: i don't use realtime, since it becomes a serious bottle neck at my level of scale |
| 15:23.05 | Freeaqingme| | JerJer, ah, that makes sense. Does that also mean you have a limited number of changes in your configs (if not, how/where do you keep those up to date then?) |
| 15:23.45 | JerJer | its all in the configuration itself |
| 15:24.21 | JerJer | i never reload asterisk on production systems |
| 15:25.41 | JerJer | but back in the day, a simple reload asterisk could lead to a memory leak |
| 15:25.46 | JerJer | or worse, deadlock |
| 15:25.54 | JerJer | so i mitigated |
| 15:26.26 | *** join/#asterisk timholum (~timh@68-117-120-138.static.eucl.wi.charter.com) |
| 15:28.45 | JerJer | rcaskey: other than consuming 'thebook' - not really |
| 15:28.49 | JerJer | !thebook |
| 15:28.54 | JerJer | ~thebook |
| 15:28.55 | infobot | it has been said that thebook is Asterisk: The Future of Telephony 2nd Edition (ISBN 0-596-51048-9) --- Order yours at http://www.oreilly.com/catalog/9780596510480/ --- Free downloadable PDF http://downloads.oreilly.com/books/9780596510480.pdf --- HTML at http://astbook.asteriskdocs.org, or http://ofps.oreilly.com |
| 15:30.01 | rcaskey | JerJer, hey, as long as its in HTML, I don't mind reading |
| 15:31.09 | *** join/#asterisk Mhaddog__ (~Mhaddog@adsl-072-149-063-056.sip.bct.bellsouth.net) |
| 15:31.11 | JerJer | i spiral bound my own duplexed PDF print out (sorry guys) :P |
| 15:31.27 | leifmadsen | nice |
| 15:31.42 | rcaskey | JerJer, I'm now firmly in the binder camp |
| 15:31.44 | JerJer | heh - wondered if you were 'round :) |
| 15:31.52 | rcaskey | I got 3 proposals in, one bound, one spiraled, and one 3-ring |
| 15:31.54 | schmidts | leifmadsen congrats to the cookbook ;) |
| 15:32.03 | leifmadsen | 3rd edition will only be HTML unless you purchase a PDF copy :) |
| 15:32.14 | rcaskey | And even though it isn't the wining proposal, the 3 ring binder one was by far the most convenient :P |
| 15:32.21 | thehar | look at how green you are leifmadsen |
| 15:32.28 | JerJer | i'll gladly buy a PDF |
| 15:32.29 | rcaskey | why would you want a PDF copy over HTML :P |
| 15:32.38 | leifmadsen | thehar: I'm mostly white |
| 15:32.44 | thehar | Mostly. |
| 15:33.02 | ChannelZ | need Kindle version |
| 15:33.05 | rcaskey | are there two-or-three hardware brands? |
| 15:33.10 | rcaskey | err go-to hardware brands? |
| 15:33.17 | leifmadsen | ChannelZ: there should be a kindle version too since it's published by o'reilly |
| 15:33.27 | leifmadsen | schmidts: thanks :) |
| 15:33.32 | ChannelZ | cool |
| 15:33.45 | leifmadsen | all of that is outside of my control |
| 15:33.47 | timholum | hello, can someone point me to a schema for mysql on cel_odbc |
| 15:36.49 | schmidts | leifmadsen maybe i have some time soon then i will try to review it |
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| 15:38.15 | leifmadsen | schmidts: awesome -- it'll be a very short review because we're going to get it published likely before A:TDG even :) |
| 15:38.27 | leifmadsen | (it'll be electronic version only for now until it's large enough to justify a print version) |
| 15:38.59 | leifmadsen | we'll just keep adding chapters over a period of time -- I think people who purchase the electronic version get free updates for some period of time |
| 15:39.04 | leifmadsen | I'm not too sure how all that works |
| 15:39.23 | schmidts | :D |
| 15:39.37 | schmidts | lets call this chapters addons ;) |
| 15:41.44 | leifmadsen | :) |
| 15:41.59 | leifmadsen | we have another chapter already planned out so we'll be working on that next I imagine after this is done |
| 15:42.20 | leifmadsen | we only had 2 months to write the book, and is scheduled to be done mid-end of March, so we're stopping at this milestone |
| 15:43.10 | *** part/#asterisk salimb (~chatzilla@83-244-177-2.cust-83.exponential-e.net) |
| 15:43.26 | rcaskey | does Asterisk use the GPU to do DSP? |
| 15:43.30 | leifmadsen | no |
| 15:43.50 | leifmadsen | please search the mailing list archives for more information |
| 15:44.09 | *** join/#asterisk akraemer (~akraemer@HSI-KBW-109-192-155-127.hsi6.kabel-badenwuerttemberg.de) |
| 15:44.47 | akraemer | Hi, someone can tell me if asterisk is able to use G722.2 codec ? |
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| 15:51.32 | *** join/#asterisk bip (~bip@unaffiliated/bip) |
| 15:55.21 | bip | Do you know any vendor of asterisk based turn-key solutions ? I have a server running asteriskNow with 3 digium fxo modules, but I'm facing problems that go beyond my problem solving skills, I have been requested to explore a fully supported solution. |
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| 16:00.28 | bipolar | I'm looking for the method for asterisk to automatically know that a number is provided by google voice so it can route the outgoing call over SIP. I found a howto on it before, but now I can't re-find it. I should have bookmarked it. :( |
| 16:02.27 | *** join/#asterisk ssureshot (~digitolx@12.196.90.82) |
| 16:03.25 | madwill | anybody here knows about how to send a conference room sip (h323) to a web tech like flash using a media server and probably a transcoding gateway |
| 16:04.07 | Freeaqingme| | bipolar, since I'm from the eu I dont really know google voice, but maybe they just use 1number block? |
| 16:04.21 | paulc | bip: Are you looking for a company with a product? Or just a consultant that could manage a vanilla/custom Asterisk installation for you? |
| 16:04.46 | paulc | bipolar: I wonder if Google Voice are participating in enum? Be nice if they were.. but I read on Twitter that they were supporting inbound via SIP, then they weren't - mixed reports? |
| 16:04.55 | bip | the first you said paulc but i have sove very narrow requirements |
| 16:05.12 | bipolar | paulc: enum! that was the word I couldn't remember ;P Let me re-google |
| 16:05.39 | bipolar | paulc: inbound sip is working at the moment, and I have a pri as backup. So if it doesn't work, it goes out via pri. |
| 16:06.53 | paulc | bip: You could try Switchvox. Or describe your problems and maybe the second option is better - plenty of people in that arena (myself included) |
| 16:08.25 | bipolar | paulc: can't find google participating in enum. guess I was mistaken. |
| 16:08.43 | bipolar | I just put all our google voice numbers in an outgoing route. |
| 16:09.03 | bip | well i have installed myself AsteriskNow with frepbx and a digium analog card i got it up and running, but here peple keep complaining so my boss wants me to get a totally outsourced solution, not an expensive one that will use the harware we already have ... |
| 16:09.39 | bip | is that a description of my problem paulc ? |
| 16:09.46 | bip | or shall I say more ? |
| 16:09.53 | *** join/#asterisk salimb (~chatzilla@83-244-177-2.cust-83.exponential-e.net) |
| 16:12.34 | paulc | bip: sounds fine. One option is to install a different software package (like Switchvox), one option is to install Asterisk from scratch and customise it to do exactly what you want. |
| 16:12.47 | bipolar | bip: I don't suppose you're anywhere in east pennsylvania, US? :) |
| 16:12.47 | paulc | bip: I guess my question should be "What is it that people are complaining about?" |
| 16:13.21 | *** join/#asterisk Tim_Toady (~moi@188.4.36.223.dsl.dyn.forthnet.gr) |
| 16:13.26 | bip | they complain about calls dropping |
| 16:13.44 | bip | they ask supe smart features i do not know how to implement |
| 16:14.06 | bip | they want fancy ivr setups ... |
| 16:14.27 | bip | basically they want to outsource the support |
| 16:14.46 | bip | and they want to do that cheaply ;-) |
| 16:14.55 | bipolar | lol |
| 16:15.04 | bip | sorry to bother :( |
| 16:16.11 | paulc | bip: quality doesn't always come cheap, but you'll be happy with the results :) |
| 16:16.14 | bipolar | Not a bother... it's just that you get what you pay for. If they want someone to do the work for them, it's better to get it done right. |
| 16:16.47 | paulc | bip: I can do dialplan features, IVR menus, all that stuff. Not sure on your calls dropping problem though - those can be hard to troubleshoot |
| 16:16.53 | bipolar | bip: where in the world are you located? |
| 16:16.54 | *** part/#asterisk wesphillips (~wphillips@137-237-233-124.harris.com) |
| 16:17.00 | bip | paulc: i m just a jack all trade sysadmin i do not hold any budget :( |
| 16:17.14 | bip | italy i' m in Rome italy bipolar |
| 16:17.39 | bipolar | bip: if your company will fly me out there and put me up in a hotel for a while I'll fix it. :D |
| 16:18.06 | paulc | bip: I usually do my work remotely over SSH with no flights or hotels required ;-) hahaha |
| 16:18.13 | bipolar | hush you! |
| 16:18.33 | bipolar | I would suguest swtiching from analog lines to a SIP provider. |
| 16:18.53 | bipolar | your call dropping issues are probbly analog line related |
| 16:19.12 | bip | bipolar: they just signed a new analog contract a bought me a 3 module digium card ... |
| 16:19.16 | bipolar | And that can take a lot of debugging. |
| 16:19.22 | bipolar | ugh |
| 16:19.22 | bip | i think so to |
| 16:19.34 | bip | but there was no cal dropping before |
| 16:19.44 | bip | but we had a differnt pbx |
| 16:19.56 | bip | and a different telephony company |
| 16:20.09 | bip | but of course now the blame is on me :( |
| 16:20.12 | bipolar | yeah. the analog cards need to be setup to the conditions of the lines. |
| 16:20.32 | paulc | bip: ah, so multiple variables have changed.. always hard to point the finger.. but bipolar's right - with some tuning on the analog lines, your FXO card should work fine |
| 16:20.34 | bip | and different card an lines |
| 16:20.41 | paulc | bip: any commonality - is it inbound calls that drop? outbound? both? |
| 16:20.46 | bip | well they point fingers at me |
| 16:20.55 | bip | I m available ;-) |
| 16:21.12 | bip | it should |
| 16:21.16 | bipolar | I havn't used analog lines in a while... but I would check the docs on how to set the cards to the like voltage tolerances. |
| 16:22.00 | bip | before we had isdn card different pbx and different telephone company |
| 16:22.14 | bip | now everything is differnt and they want me to fix it :( |
| 16:22.21 | paulc | bip: really? why'd you switch from ISDN to analog? price? |
| 16:22.22 | bipolar | you switched from isdn? that sucks. |
| 16:22.51 | *** join/#asterisk cerberus_za (~coert@196-210-142-16.dynamic.isadsl.co.za) |
| 16:23.08 | bip | I have to tell ya |
| 16:23.12 | bip | they never ask me |
| 16:23.16 | *** join/#asterisk JonnyD_work (~Jon@173.226.80.154) |
| 16:23.47 | bip | lemme make an example i had al ol piece of harware i had turned into a file server |
| 16:23.59 | bip | they replaced it with a lil soho nas |
| 16:24.18 | bip | and now no one of the lovely lil scripts I had works :( |
| 16:24.33 | bip | they always do that ... |
| 16:24.37 | bip | management :( |
| 16:24.57 | paulc | bip: what did the scripts do and why don't they run on the new box? |
| 16:25.06 | paulc | bip: and who chose the new PBX + analog lines? |
| 16:25.17 | paulc | bip: it half sounds like SEP - "Someone Else's Problem" ;-) |
| 16:25.26 | bip | because the new box is not a real linux system |
| 16:25.42 | bip | I' m the someone else paulc |
| 16:26.39 | bipolar | bip: try editting your zapata.conf file and increasing the busydetect line to a higher number. If that doesn't fix it, try setting callprogress to no. |
| 16:26.54 | bip | wait |
| 16:27.04 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 16:27.11 | bip | I don t think i have a zapata.conf file |
| 16:27.14 | paulc | bip: so if it was me, and my company, I'd go with a standard linux install, a standard Asterisk install, then configure it all myself so it worked exactly how I wanted it to. Forget the pretty web admin screens etc - administer it directly. |
| 16:27.26 | bip | but i have busydetect written somewhere |
| 16:27.34 | *** join/#asterisk jkroon (~jkroon@197.175.4.231) |
| 16:27.37 | bip | me too |
| 16:27.47 | bip | but "they" wanted a gui |
| 16:27.57 | bip | i m not supposed to think |
| 16:28.08 | bip | the ask gui, i give gui :( |
| 16:28.20 | bip | nothing works, my fault :( |
| 16:28.56 | paulc | bip: *shrugs* so go with a consultant who will help you with FreePBX (ask in #freepbx) or take a look at switchvox and pay the money to them |
| 16:30.01 | bip | switvox is a appliance or will run on my existying hardware ? |
| 16:30.06 | ssureshot | I'm having issues loading chan_dahdi.so.. with error Error loading module 'chan_dahdi.so': /usr/local/asterisk1.8.2/lib/asterisk/modules/chan_dahdi.so: undefined symbol: pri_sr_set_reversecharge |
| 16:30.17 | bipolar | bip: I thought all analog cards use zapata to config... maybe just mine does. what cards do you have? |
| 16:30.36 | bip | tdm410 |
| 16:31.35 | bipolar | bip: you must have a zapata.conf then. should be in /etc somewhere. |
| 16:31.45 | tzafrir | bip, zapata.conf? on a new installation? |
| 16:31.54 | tzafrir | What version of asterisk is it? |
| 16:32.02 | paulc | bip: They do both I think, but I'm not totally sure - you'd have to check www.switchvox.com |
| 16:32.22 | bipolar | tzafrir: not exactly sure what he has going on.... dropped calls on analog lines. |
| 16:32.33 | Qwell | bipolar: Zaptel was replaced by DAHDI quite a long time ago. |
| 16:32.43 | paulc | bip: there are plenty of GUI options but it's really down to why do they want one? how often do you add extensions etc - the real power is in building the dialplan yourself, generally |
| 16:32.48 | bipolar | Qwell: hah! shows how out of date I am :P |
| 16:33.04 | tzafrir | bipolar, well, what version of Asterisk is it? |
| 16:33.13 | bipolar | tzafrir: not me, bip. |
| 16:33.18 | tzafrir | oh |
| 16:33.24 | bip | never add extensions we are a very small company |
| 16:33.53 | paulc | bip: so what do they want the GUI for exactly? |
| 16:34.20 | bip | because they are aafraid of command line i me or my collegaue are not around |
| 16:34.32 | tzafrir | ssureshot, this is a symbol that should come from libpri |
| 16:34.40 | bip | beside they thonk gui == modern command line == old |
| 16:34.46 | bip | imho they are idiots |
| 16:35.00 | paulc | bip: but what would they need to change? that couldn't wait for someone who knew what they were doing? |
| 16:35.01 | tzafrir | Sounds like lasterisk was built with a certain version of libpri and is then installed with an older version |
| 16:35.20 | bip | nothing but resistance is futile ... |
| 16:35.32 | *** join/#asterisk piros (~piros@host9-162-static.15-79-b.business.telecomitalia.it) |
| 16:35.38 | bip | you ppl that think since they are the boss they know better :( |
| 16:35.46 | piros | hi everybody |
| 16:35.52 | bip | do you know ... |
| 16:36.16 | paulc | bip: maybe they just need reassurance from someone they feel they can trust (ie external) |
| 16:36.22 | tzafrir | still wonders what version of Asterisk bip is using that has zapata.conf |
| 16:36.41 | bip | tzafrir: i thik it s 1.6 ... |
| 16:36.47 | bip | i check |
| 16:36.48 | tzafrir | asterisk -V |
| 16:36.52 | paulc | bip: I did a conference call with a company that was quite remote.. talked to their CFO.. said what I'd do, how much it would cost, how we'd manage the work flow.. they were happy, I got a prepaid block of work.. |
| 16:37.06 | ssureshot | tzafrir: that makes a whole log of sense actually in my this attempt I installed libpri , dahdi and asterisk when it didn't work I installed the distributions packages for libpri and dahdi... so I need to clean that up I guess thank you |
| 16:37.19 | bip | Asterisk 1.6.2.7 |
| 16:37.28 | bipolar | bip: I would probably start with a fresh install of asterisknow or something. put just one line on it, leave the others on the old phone system until you get this stuff working. I'd pull an all nighter so they weren't around to bother me while I worked. :P |
| 16:37.51 | tzafrir | bip, for that version you need chan_dahdi.conf (and dahdi, rather than zaptel) |
| 16:38.05 | bip | bipolar: they do not pay me enuff for that ;-) |
| 16:38.07 | bipolar | tzafrir: thanks for catching that. I need to upgrade :P |
| 16:38.15 | bip | yeah one sec ... |
| 16:38.25 | bipolar | bip: they might pay you more in the future if you prove yourself. |
| 16:39.00 | bip | pastebin.com/9uiBEsCV |
| 16:39.09 | bip | they might |
| 16:39.14 | kaldemar | tzafrir: < bip> I don t think i have a zapata.conf file |
| 16:39.25 | bip | but i 'm old and got sick of proving myself |
| 16:39.34 | bip | my fault I guess :( |
| 16:40.15 | bip | http://pastebin.com/9uiBEsCV |
| 16:40.28 | bip | in that file there is a line about busydetect |
| 16:40.32 | piros | Can anyone help me with the configuration of a TDM400p card? |
| 16:41.07 | bip | somebody here or in #freepbx told me to add it because telehone kept ringing after hangup ... |
| 16:42.17 | paulc | bip: you mean on an inbound call? |
| 16:43.12 | bip | yes I think so |
| 16:43.47 | bip | if i rang myself and put down phone , here the phone on my desk kept ringing |
| 16:44.02 | bip | sorry my awful english :( |
| 16:44.45 | *** join/#asterisk kjs (~kjs@fedora/kjs) |
| 16:44.52 | paulc | bip: it's common, but should stop within a few seconds - it's down to the PBX not knowing the line has stopped ringing until the inter-ring delay has been exceeded |
| 16:44.52 | bip | anyhow wswitvox looks like a ggod solution, but is a one stop appliance form factor solution |
| 16:45.11 | paulc | bip: and no worries - your english is much better than my italian :-) |
| 16:45.18 | bip | well it lasted long but that was fixed |
| 16:45.21 | bip | anyhow |
| 16:45.31 | bip | we keep having dropped calls |
| 16:45.57 | bip | i suspect is the provider fault, but i m considered guilty until proven innocent :( |
| 16:46.22 | paulc | bip: the case to present is "A traditional PBX costs Exxxxx. Our hardware costs Exxxx. A consultant costs Exxxx. So long as (hardware + consultant) < PBX, we win" |
| 16:46.23 | *** join/#asterisk salimb (~3laz3r@83-244-177-2.cust-83.exponential-e.net) |
| 16:46.41 | paulc | bip: you can prove the provider by plugging your phone straight into the line.. making inbuond and outbound calls.. see if it fails the same |
| 16:46.54 | *** part/#asterisk salimb (~3laz3r@83-244-177-2.cust-83.exponential-e.net) |
| 16:46.55 | paulc | but it's proably some voltage sensitivity on your analog card |
| 16:47.04 | bip | so basically the request from my boss is: find a pbx that will use the card we have bought with full commerciual support and let em get da shit, and find em cheap too ;-) |
| 16:48.50 | benngard | is there some way to continue in the dialplan even if a dialplan application hangs upp the call? |
| 16:48.53 | volga629 | how can happend that female voices on conference triggering star button ? |
| 16:49.13 | bip | ok i will check the voltage thing ... |
| 16:49.14 | volga629 | DTMF probles ? |
| 16:49.31 | bip | and I agree building from scratch was the way to go ... |
| 16:49.57 | bip | but you know the joke about right wrong and military way ? |
| 16:50.12 | bip | we figured a fourth option: our way ;-) |
| 16:50.31 | bip | wich is even worse then military ;-) |
| 16:50.39 | benngard | :) |
| 16:51.38 | bip | my boss scream like adrill sargent, but is not even tryin to keep my ass alive :( |
| 16:52.20 | piros | can you help me with my pbx? |
| 16:52.26 | paulc | bip: time for a new job maybe? ;) |
| 16:52.35 | paulc | piros: what's your problem? |
| 16:54.04 | bip | you are right, but I do not know about where u are, but here there is a huge and longstanding market crisis, which means this job or nothing more or less ... |
| 16:54.31 | piros | hi paulc |
| 16:54.44 | piros | i have a problem with a tdm400p digium card |
| 16:54.59 | piros | it seems to work |
| 16:55.25 | piros | but i can't make inbound and outgoig call |
| 16:55.56 | piros | at cli i receive cause 34 |
| 16:56.52 | paulc | piros: "no circuit/channel available" |
| 16:56.55 | paulc | hmm |
| 16:57.02 | piros | this i s the message |
| 16:57.04 | piros | [Mar 16 17:56:54] WARNING[13503]: app_dial.c:1549 dial_exec_full: Unable to create channel of type 'DAHDI' (cause 34 - Circuit/channel congestion) |
| 16:57.04 | piros | <PROTECTED> |
| 16:57.30 | paulc | is the card up? and is the dahdi module loaded? |
| 16:57.33 | piros | when i make a call to line the cli remain mute |
| 16:57.38 | piros | yes |
| 16:57.40 | *** join/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 16:57.47 | piros | dahdi show channel 1 |
| 16:57.49 | piros | give this |
| 16:58.06 | piros | File Descriptor: 17 |
| 16:58.07 | piros | Span: 1 |
| 16:58.07 | piros | Extension: |
| 16:58.07 | piros | Dialing: no |
| 16:58.07 | piros | Context: DID_trunk_2 |
| 16:58.07 | piros | Caller ID: |
| 16:58.08 | piros | Calling TON: 0 |
| 16:58.08 | piros | Caller ID name: |
| 16:58.09 | piros | Mailbox: none |
| 16:58.09 | piros | Destroy: 0 |
| 16:58.13 | paulc | ~pastebin |
| 16:58.13 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 16:58.40 | piros | InAlarm: 0 |
| 16:58.41 | piros | Signalling Type: FXS Kewlstart |
| 16:58.41 | piros | Radio: 0 |
| 16:58.41 | piros | Owner: <None> |
| 16:58.41 | piros | Real: <None> |
| 16:58.41 | piros | Callwait: <None> |
| 16:58.42 | piros | Threeway: <None> |
| 16:58.42 | piros | Confno: -1 |
| 16:58.43 | piros | Propagated Conference: -1 |
| 16:58.43 | piros | Real in conference: 0 |
| 16:59.06 | paulc | piros: the only box I have with a PRI in is still using zaptel not dahdi.. but there's something like "dahdi show status" - what does that say? (use pastebin for the answer) |
| 16:59.23 | piros | DSP: no |
| 16:59.23 | piros | Busy Detection: no |
| 16:59.23 | piros | TDD: no |
| 16:59.23 | piros | Relax DTMF: no |
| 16:59.24 | piros | Dialing/CallwaitCAS: 0/0 |
| 16:59.24 | piros | Default law: alaw |
| 16:59.25 | piros | Fax Handled: no |
| 16:59.25 | piros | Pulse phone: no |
| 16:59.26 | piros | DND: no |
| 16:59.26 | piros | Echo Cancellation: |
| 16:59.58 | ChannelZ | piros: pastebin.com |
| 17:00.08 | piros | <PROTECTED> |
| 17:00.08 | piros | <PROTECTED> |
| 17:00.09 | piros | Actual Confinfo: Num/0, Mode/0x0000 |
| 17:00.09 | piros | Actual Confmute: No |
| 17:00.09 | piros | Hookstate (FXS only): Onhook |
| 17:00.15 | piros | that's all |
| 17:00.20 | ChannelZ | LEARN HOW TO READ |
| 17:00.39 | *** join/#asterisk garymc (~chatzilla@host81-148-15-59.in-addr.btopenworld.com) |
| 17:00.42 | piros | channelZ |
| 17:01.15 | piros | what's bin.com? |
| 17:01.55 | paulc | piros: go to pastebin.com to show us when you have lots of output to share |
| 17:02.53 | kaldemar | piros: move your eyes 25 lines up to see what infobot said. |
| 17:03.06 | paulc | for clarity: |
| 17:03.08 | paulc | ~pastebin |
| 17:03.08 | infobot | [~pastebin] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few : http://www.pastebin.com , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ , or install pastebinit with yum or aptitude. |
| 17:04.48 | piros | ok i post all my config |
| 17:04.53 | piros | in sterisk forum |
| 17:05.15 | benngard | here u have a pastebin from me: http://pastebin.com/hsD3zFHL |
| 17:05.28 | piros | http://forums.asterisk.org/viewtopic.php?f=36&t=77500 |
| 17:05.42 | piros | look at this there is all my config |
| 17:05.48 | piros | and status message |
| 17:05.51 | benngard | can some1 tell me if it is possible to continue in the dialplan even ehen call was hang up |
| 17:07.09 | paulc | benngard: take a look at the "h" extension |
| 17:08.07 | paulc | piros: I don't know what's going on with your card :-( |
| 17:08.56 | bip | does anybody know the pricing of the switchvox ently-level configuration ? |
| 17:08.57 | piros | NP, thank you paulc! |
| 17:09.36 | kaldemar | benngard: if you dial that extension using a phone, does it continue to the next priority? |
| 17:10.01 | piros | any one other can help me? |
| 17:11.52 | paulc | bip: Go to https://www.digium.com/en/forms/contact_swvx_sales.php and do live chat with someone in sales |
| 17:11.53 | kaldemar | bip: you'll find pricing information here: http://www.switchvox.com/ |
| 17:12.04 | bip | ok |
| 17:12.10 | benngard | need to read about "h" extension, and will try too calll from a phone |
| 17:12.17 | bip | i will |
| 17:12.20 | benngard | h = hangup? |
| 17:14.26 | piros | does anyone know tdm400p and dahdi? |
| 17:14.27 | kaldemar | benngard: https://wiki.asterisk.org/wiki/display/AST/Handling+Special+Extensions |
| 17:17.45 | benngard | kaldemar: Executing [h@fax.inputinterior.se:1] NoOp("Local/0317998975@fax.inputinterior.se-49ad;2", "") in new stack ;) thx!!!!!!!!!!!!!!!!!!!! |
| 17:18.12 | benngard | kaldemar: i can do whatever there i guess |
| 17:21.09 | kaldemar | benngard: sure, but still it is strange that the system app hangs up the call.. |
| 17:22.05 | *** join/#asterisk bn-7bc (bjarne@macbook-pro.lan.noare-1.holmedal.net) |
| 17:23.15 | pallet | I know I'm in the wrong channel, but does anyone here know 3CX ? |
| 17:24.00 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 17:25.29 | benngard | kaldemar: guess its the receivefax thats hangs up the call |
| 17:25.33 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 17:25.39 | benngard | kaldemar: -- Executing [h@fax.inputinterior.se:4] System("Local/0317998975@fax.inputinterior.se-b1a1;2", "mutt -s 'New fax from 0317998985' -a /var/spool/asterisk/fax/20110316-182902-0317998985-0317998975.pdf magnus.b@inputinterior.se < /dev/null") in new stack |
| 17:26.50 | benngard | kaldemar: u just made a lot of people happy, me most :) |
| 17:27.31 | benngard | runs for a drink |
| 17:27.33 | pabelanger | pallet: if you know you are in the wrong channel, why do you ask? |
| 17:28.18 | pallet | pabelanger, I was justing hoping that in a channel full of voip specialists, someone might know ;) |
| 17:30.06 | leifmadsen | we're asterisk specialists, and perhaps sip specialists, but not "every platform you can think of" specialists |
| 17:30.19 | leifmadsen | I doubt anyone here has even used as 3CX system |
| 17:30.26 | pallet | leifmadsen, ok, thanks dude |
| 17:30.29 | pallet | ;) |
| 17:31.09 | leifmadsen | flashes back to the 90s |
| 17:31.19 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 17:31.36 | pallet | leifmadsen, something I said ? |
| 17:31.52 | leifmadsen | nobody says, "dude" anymore :) |
| 17:31.56 | leifmadsen | (Other than Dude himself) |
| 17:32.07 | leifmadsen | ~dude |
| 17:32.07 | infobot | dude is, like, Be most excellent to each other! Also the moniker of Jim Dixon. |
| 17:32.10 | pallet | LOL, they do in my world ;) |
| 17:32.10 | *** join/#asterisk greg_logan (~greg@m1330.usask.ca) |
| 17:32.28 | leifmadsen | pictures a foreign company getting Seinfeld for the first time |
| 17:32.33 | leifmadsen | s/company/country/ |
| 17:32.41 | *** part/#asterisk greg_logan (~greg@m1330.usask.ca) |
| 17:33.26 | piros | anybody can help me with cause 34 in asterisk? |
| 17:33.28 | paulc | remembers contracting at a place where all the server names were Seinfeld characters |
| 17:33.40 | paulc | not being an avid fan, having their pictures as desktop wallpaper didn't really help me much |
| 17:34.02 | pabelanger | paulc: Cause No. 34 - no circuit/channel available. |
| 17:34.05 | pallet | All my servers are Ron, Baxter, Brick and Champ ;) |
| 17:34.11 | pabelanger | http://networking.ringofsaturn.com/RemoteAccess/isdncausecodes.php |
| 17:34.13 | pallet | which is only funny if you like Anchorman |
| 17:34.36 | paulc | pabelanger: I think it's piros that's asking ;-) |
| 17:34.47 | pabelanger | piros: ^ |
| 17:37.33 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 17:37.34 | paulc | piros: Can you receive inbound calls on that line ok, the problem is only with outbound? |
| 17:39.12 | leifmadsen | all my servers start with the letter 'S': scrappy, scooter, sleezy.. |
| 17:39.31 | *** join/#asterisk funxion (3fd6eca9@gateway/web/freenode/ip.63.214.236.169) |
| 17:40.47 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 17:40.47 | piros | pabelanger i'm here |
| 17:42.17 | piros | i have a problem with my tdm400p card |
| 17:42.36 | piros | you can see all my config at this address http://forums.asterisk.org/viewtopic.php?f=36&t=77500 |
| 17:43.22 | leifmadsen | tdm400p cards qualify for support from Digium directly |
| 17:43.31 | leifmadsen | you paid for it when you bought the card, so you should just use it |
| 17:44.09 | funxion | interesting concept |
| 17:44.17 | piros | yes i know |
| 17:44.55 | piros | but i have to stop a server to read the code |
| 17:45.25 | piros | would you please look at my config files? |
| 17:45.42 | piros | if there is something of wrong |
| 17:45.52 | funxion | whats is your problem with the tdm400? |
| 17:46.02 | piros | i can fix it without stop server |
| 17:46.06 | piros | thank you |
| 17:46.09 | piros | cause 34 |
| 17:46.18 | funxion | maybe I missed something but it seems that you had the channel assignment commented out |
| 17:46.32 | bip | sombody knows this poduct or something similar: http://www.gls-net.net/listaProdotti.php?cat=3 |
| 17:47.27 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 17:47.54 | funxion | I have an inbteresting issue with a tc400b combined with a te410p |
| 17:48.17 | *** join/#asterisk sahafeez (~sahafeez@65-119-47-100.dia.static.qwest.net) |
| 17:48.30 | *** join/#asterisk jaskew (~jdaskew@netblock-75-79-182-110.dslextreme.com) |
| 17:48.45 | funxion | call are working great except for one specific number that Im getting a weird one way audio clipped and delayed |
| 17:49.13 | funxion | all voip is g729:60 |
| 17:49.52 | piros | are you talking about users.conf? |
| 17:49.52 | *** join/#asterisk timahvo1 (~rogue@41.223.57.73) |
| 17:49.55 | funxion | I've had similar problem when trying to transcode and repacketize at the same time however in this case its TDM to voip |
| 17:50.05 | funxion | piros chan_dahdi.conf |
| 17:50.10 | *** part/#asterisk Joe_CoT (~joecot@pdpc/supporter/active/joe-cot) |
| 17:50.53 | piros | no it isn't commented |
| 17:50.59 | funxion | scrathc that you do have it piros |
| 17:51.58 | funxion | looks like an IRG problem |
| 17:52.01 | funxion | IRQ |
| 17:52.07 | funxion | have you already looked into that |
| 17:54.58 | *** join/#asterisk vinhdizzo (~vinh@dhcp-v009-212.mobile.uci.edu) |
| 17:55.23 | *** join/#asterisk b14ck (~b14ck@cpe-72-129-70-245.socal.res.rr.com) |
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| 18:06.33 | *** join/#asterisk NightMonkey (debian-tor@pdpc/supporter/professional/nightmonkey) |
| 18:07.28 | *** join/#asterisk dmz (~dmz@64.203.207.101.dyn-cm-pool-54.hargray.net) |
| 18:08.21 | piros | ithank you funxion |
| 18:08.32 | piros | i'll try with digium official support |
| 18:11.10 | *** part/#asterisk Olivier_54 (~Olivier_5@vaillant.famille-fontes.net) |
| 18:23.55 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
| 18:25.21 | *** join/#asterisk remnant13 (~Gray@unaffiliated/remnant13) |
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| 18:35.32 | *** join/#asterisk oraqol (~oraqol@67.221.68.250) |
| 18:37.20 | oraqol | hey guys, new user to asterisk, trying to setup my first pbx but i want to use google voice as my carrier. I hear asterisk 1.8 supports this but I'm having trouble finding a repo with all the required packages for gv. Do i have to compile from source? |
| 18:37.29 | oraqol | does the source code contain the gv packages? |
| 18:38.34 | Qwell | You need iksemel. |
| 18:39.02 | Freeaqingme| | "iksemel is an XML (eXtensible Markup Language) parser library designed for Jabber applications. " < Qwell that? |
| 18:39.24 | *** join/#asterisk lanning (~lanning@208.87.233.137) |
| 18:39.48 | Qwell | Yes. |
| 18:43.17 | oraqol | So...is there a repo for asterisk 1.8? |
| 18:43.24 | oraqol | im on ubuntu 10.10 |
| 18:43.35 | Qwell | looks at pabelanger |
| 18:44.20 | pabelanger | oraqol: expect an announcement in the next few days |
| 18:44.21 | Qwell | hrm, nevermind, guess there's no iksemel package there. |
| 18:44.42 | Qwell | maybe it's in ubuntu proper though |
| 18:45.21 | oraqol | hmm, looks like ill have to compile, oh well. I assume the source will have all packages required for gv? |
| 18:45.37 | Qwell | You'll need iksemel. |
| 18:46.55 | oraqol | ok so compile 1.8 and iksmel, or get iksmel on repo and it should work with gv? |
| 18:47.11 | Qwell | However you get it doesn't matter. |
| 18:47.18 | oraqol | cool, thanks guys |
| 18:47.30 | Qwell | That'll be $499.95. |
| 18:47.34 | oraqol | ill let you know how it turns out |
| 18:47.37 | oraqol | HA! |
| 18:47.43 | leifmadsen | he's not kidding |
| 18:47.45 | oraqol | ok, ill email you my credit card number |
| 18:48.01 | pabelanger | Qwell: yes, iksemel-dev should be enabled with the Debian packages |
| 18:48.06 | Qwell | pabelanger: ahh, cool |
| 18:48.10 | dmz | hey if i have a t1 card on an asterisk box with about 100 inbound DIDs on a hunt group; does a T1 card normally tell asterisk what # was being called? I've seen callerid but what about what DID was called and is coming in on the t1 channel? |
| 18:48.45 | Qwell | dmz: It should, but I think it's considered an extra feature on a plain T1. |
| 18:49.00 | dmz | something i need to get from the carrier? |
| 18:49.06 | Qwell | maybe |
| 18:49.10 | dmz | and what variable would asterisk see it with? any idea? |
| 18:49.27 | Qwell | ${CALLERID(num)} and/or ${CALLERID(name)} |
| 18:49.34 | dmz | that's the person calling |
| 18:49.43 | Qwell | umm |
| 18:49.44 | dmz | i want the # they called; have several hundred did's |
| 18:49.47 | Qwell | you're right |
| 18:49.50 | dmz | need to know where to send them to |
| 18:50.09 | Qwell | My brain is currently broken. I'll leave it to someone else. |
| 18:50.16 | dmz | :) |
| 18:50.49 | *** join/#asterisk manji (~manjiki@ppp-94-65-252-87.home.otenet.gr) |
| 18:51.22 | leifmadsen | doesn't that just enter as ${EXTEN} |
| 18:51.38 | dmz | hmm, maybe |
| 18:51.42 | leifmadsen | when the call comes into the context the T1 (using PRI signaling at least) requests a route |
| 18:51.57 | leifmadsen | exten => _NXXNXXXXXX,1,Verbose(2,Number dialed was ${EXTEN}) |
| 18:52.07 | dmz | cool |
| 18:52.28 | dmz | this will be my 1st t1 on asterisk |
| 18:52.34 | dmz | wanted to be sure before i started building system |
| 18:52.48 | leifmadsen | well T1 is just the trunk, but the signaling method used will be what provides the features you're looking for |
| 18:53.34 | dmz | it's actually routing to a fax in the end but just a standard 24 channel voice t1 pri |
| 18:53.47 | Qwell | Those words don't go together. |
| 18:53.50 | dmz | :) |
| 18:53.52 | Qwell | Not in that order, at least. |
| 18:54.06 | Qwell | PRI is 23 voice channels. PRI is a protocol over T1. |
| 18:54.12 | dmz | been a long time since i had to deal with t1's directly :) |
| 18:54.18 | dmz | hmm like 17 years |
| 18:54.24 | leifmadsen | T1 contains 24 time slots |
| 18:54.30 | dmz | that's what i remember |
| 18:54.37 | Qwell | leifmadsen: your face has 24 time slots. OH BURN. |
| 18:54.38 | leifmadsen | PRI uses 1 time slot for signalling and 23 time slots for voice |
| 18:54.43 | thehar | Qwell: you beat me! |
| 18:54.44 | dmz | heh |
| 18:54.46 | leifmadsen | Qwell: your mom seems to like them though |
| 18:54.48 | thehar | *shakes fist* |
| 18:54.54 | Qwell | leifmadsen: I am not surprised. |
| 18:55.32 | dmz | now if i could just find a carrier with an api for provisioning did's i'd be all set :) |
| 18:58.10 | leifmadsen | just uses SIP routes for that :) |
| 19:13.09 | *** join/#asterisk grEvenX (~even@cC0FD00C3.dhcp.bluecom.no) |
| 19:14.33 | *** join/#asterisk Buklov (~Buklov@mail.sapsun.su) |
| 19:24.04 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 19:33.31 | *** join/#asterisk Romeo- (~romi@unaffiliated/romeo/x-000000001) |
| 19:45.38 | *** join/#asterisk serafie (~erin@207.98.195.107) |
| 19:49.07 | *** join/#asterisk serafie (~erin@nat/digium/x-rlgxniodkrknigcj) |
| 19:59.50 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-gwzxzqtpgjmyxcgn) |
| 20:02.03 | *** join/#asterisk killown (~killown@unaffiliated/killown) |
| 20:02.34 | atan | I have a strange issue taking place on inbound calls |
| 20:02.49 | atan | I have Dial(SIP/4566,30) in my dialplan for the incoming call. SIP 4566 rings which is just fine... |
| 20:03.14 | atan | It used to ring for 30 seconds but now it only rings perhaps 3 times, maybe 10 seconds before skipping to the next item in the dialplan |
| 20:04.19 | atan | I assume "30" in this is 30 seconds... wonder where I went wrong on this and why it forwards? |
| 20:05.37 | *** join/#asterisk torgnyw (~torgny.wa@50-124-8.connect.netcom.no) |
| 20:06.55 | torgnyw | Hi, Anyone who can help me getting a PRI E1 up and running? I have configured it, but get an error message: Span 1: No D-channels are available! Using primary channel as D-channel anyway! |
| 20:07.02 | p3nguin | atan: core set verbose 4 |
| 20:07.15 | p3nguin | atan: Make a call to the extension that dials that phone. |
| 20:07.26 | p3nguin | atan: Let's see what it is doing. |
| 20:08.54 | atan | is there a way for me to clear my screen in asterisk? |
| 20:09.11 | p3nguin | hold down the Enter key for a while |
| 20:10.37 | atan | 1 ring = ~5 seconds? |
| 20:11.15 | p3nguin | US ring = 4 second on, 2 second off |
| 20:12.11 | *** join/#asterisk gerhard7 (~gerhard7@82-171-103-215.ip.telfort.nl) |
| 20:12.14 | torgnyw | Im using a Digium TE121B, and when trying to call out to DAHDI I get error: Unable to create channel of type DAHDI. (Cause 34 - Circuit/Channel Congestion) |
| 20:12.41 | torgnyw | I also have RED alarm when trying DAHDI SHOW STATUS in Asterisk Console |
| 20:13.02 | p3nguin | Wait, I said that backward... |
| 20:13.26 | p3nguin | US ring = 2 second on, 4 second off |
| 20:13.31 | p3nguin | sorry |
| 20:13.41 | atan | p3nguin, http://pastebin.com/YJfdC7Qr |
| 20:13.58 | atan | Seems it is running fine now. I'm slightly confused. The client said his phone skipped right to his voicemail. |
| 20:14.08 | atan | Sorry, to the next dial() in his dialplan. |
| 20:14.20 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 20:14.42 | atan | I suppose this is about 30 seconds now that I listen to the rings. |
| 20:15.01 | atan | goes back in to his hole in the ground |
| 20:15.05 | atan | Thank you p3nguin. |
| 20:15.10 | p3nguin | Nobody picked up in 30000 ms |
| 20:15.19 | p3nguin | That's 30 seconds. |
| 20:15.24 | *** join/#asterisk slidesinger (~slidesing@c-68-44-99-163.hsd1.nj.comcast.net) |
| 20:16.07 | p3nguin | 30 seconds would be 5 ring cycles. |
| 20:16.44 | *** join/#asterisk eject_ck (~eject_ck@83-218-246-246.dynamic.vega-ua.net) |
| 20:16.50 | eject_ck | Hi all |
| 20:16.53 | atan | The 30 seconds is accounted for right now. I can't explain why earlier it was only one or two rings before the dialplan moved on to the second Dial(). Perhaps the ATA adapter reset or dropped the connection for whatever off reason. |
| 20:17.12 | atan | p3nguin, thanks for the time though. Good to know each "ring" is ~6 seconds. |
| 20:18.24 | eject_ck | I got 5 SIP lines from my SIP provider, how should I use them is I have only one number? |
| 20:18.39 | eject_ck | I got 5 SIP lines from my SIP provider, how should I use them is I have only one number? |
| 20:18.42 | eject_ck | sorry, |
| 20:20.40 | p3nguin | You'll use one channel every time someone makes a call to you. |
| 20:21.09 | Sertys | u need em for DID or for dialout? |
| 20:21.16 | Sertys | specify that pls |
| 20:22.15 | Sertys | your provider may have meant your sip account is capable of 5 concurrent calls |
| 20:22.18 | eject_ck | for both |
| 20:22.54 | eject_ck | Sertys, p3nguin: yes, ISP stated that I can use 5 lines simultaneously |
| 20:23.00 | p3nguin | Regardless, you'll use one channel for every call you get inbound, and you'll use another channel for every call outbound. |
| 20:23.02 | eject_ck | SIP provider I mean |
| 20:23.08 | p3nguin | ITSP |
| 20:24.10 | eject_ck | my question is how should I configure asterisk for it. With 1 channel it's pretty easy - 1) register => 2) add peer to sip.conf 3) configure extensions.conf for both in and out calls |
| 20:24.28 | p3nguin | That's how it's done. |
| 20:24.33 | eject_ck | so, for inbound calls no changes needed ? |
| 20:24.35 | p3nguin | It's done the same way regardless. |
| 20:24.42 | rcaskey | just out of curiosity, are there any hot-swappable FXS card + backplanes/ |
| 20:24.54 | eject_ck | for outbound I need no changes ? |
| 20:24.58 | p3nguin | For inbound calls, you create the SIP peer and an extension to do something useful. |
| 20:25.08 | eject_ck | I have it already :) |
| 20:25.26 | p3nguin | For outbound calls, you'll use the same peer you've already created, and another extension to call out through that peer. |
| 20:25.51 | p3nguin | It doesn't matter how many channels they give you, the configuration is the same. |
| 20:26.06 | p3nguin | More channels just means more concurrent calls. |
| 20:26.20 | eject_ck | p3nguin: "another extension" what you exactly mean ? |
| 20:26.31 | p3nguin | If you have more calls than available channels, you'll get congestion. |
| 20:27.00 | p3nguin | I mean you create one extension for calls inbound, and you create one extension for calls outbound through the peer. |
| 20:27.05 | eject_ck | how can I detect congestion in context (wanna play to subscriber some custom message) |
| 20:27.21 | eject_ck | can I pastebin ? |
| 20:27.39 | p3nguin | I doubt you can control the congestion tones that your provider will give to callers. |
| 20:27.49 | p3nguin | Usually it is a fast busy. |
| 20:29.21 | torgnyw | Please, anyone who knows how to configure E1/T1 for Digium cards? |
| 20:30.51 | Kobaz | anyone have a good howto on using a 7940 with chan_sccp |
| 20:30.58 | Sertys | so many questions, so few answers |
| 20:31.28 | dwayne | torgnyw, pastebin your configs |
| 20:33.07 | p3nguin | kobaz: I don't have a howto, but I can probably answer any questions you have if you run into a problem. |
| 20:33.07 | Kobaz | nifty |
| 20:33.39 | Kobaz | should i upgrade the phone firmware? i have dsp 4.0 and boot 8.0 |
| 20:34.24 | torgnyw | /etc/dahdi/system.conf |
| 20:34.26 | Kobaz | p3nguin: i set the phone to factory defaults, it's looking for a server at 172.20.6.200 |
| 20:34.28 | p3nguin | Those numbers aren't familiar to me. You just need to have the SCCP firmware files on the tftpd. |
| 20:34.35 | Kobaz | k |
| 20:34.43 | Kobaz | do you know where to get them? |
| 20:35.26 | p3nguin | There are some public repositories of this proprietary software, but you're really supposed to pay Cisco to download them off Cisco's web site. |
| 20:35.33 | dwayne | ~pb |
| 20:35.33 | infobot | [~pb] A "pastebin" is a web-based service where you should paste anything over 3 lines so you don't flood the channel. Here are links to a few: http://www.pastebin.com , http://pastebin.us , http://pastebin.ca , http://channels.debian.net/paste , http://paste.lisp.org , http://bin.cakephp.org/ , http://asterisk.pastey.net/ ; or install pastebinit with yum or aptitude. |
| 20:35.33 | Kobaz | heh |
| 20:35.38 | dwayne | torgnyw, ^^ |
| 20:35.52 | p3nguin | Google will give you the public repository if you use the right key words. |
| 20:36.51 | Kobaz | sounds good |
| 20:37.00 | Severian | If I do a "iax2 show peers registered", does it show me the registrations I have made to other servers or the registrations other servers have made to this machine? |
| 20:37.37 | p3nguin | As far as the phone looking for the wrong IP address, you have two choices: configure your DHCP server to give out the proper address for TFTP, or wait on the phone to turn on and then go into the network configuration and override the tftp by using "alternate tftpd YES" and then set the tftpd address to your actual address. |
| 20:37.38 | torgnyw | Here is my pastbin: http://pastebin.com/FCdUKEFD |
| 20:39.32 | Kobaz | k |
| 20:39.43 | torgnyw | This is my chan_dhadi.conf file: http://pastebin.com/pdwWxzZP |
| 20:39.51 | Kobaz | what option should i set for the address... is it option 66? |
| 20:40.23 | Kobaz | from my server logs the phone is hitting the tftp |
| 20:40.33 | p3nguin | 66 should be enough, but I think I also use option 150. |
| 20:40.38 | Kobaz | it's looking for SEP001121D89A42.cnf.xml CTLSEP001121D89A42.tlv |
| 20:40.50 | p3nguin | You only need the SEP file, not the other. |
| 20:41.04 | Kobaz | k |
| 20:41.17 | Kobaz | where do i get a sep file? :) |
| 20:41.31 | p3nguin | Your SEP file will tell the phone where the Asterisk server is and it will tell it what firmware files to pull off the tftp. |
| 20:41.34 | p3nguin | You create it. |
| 20:41.45 | p3nguin | Let me give you a basic one. |
| 20:41.49 | Kobaz | yeah, do you have a sample.. |
| 20:41.52 | Kobaz | ah perfect, awesome |
| 20:42.04 | Kobaz | this is my first foray, so like... total cisco newb here |
| 20:42.35 | eject_ck | p3nguin: http://pastebin.com/dceNSGn8 |
| 20:43.27 | dwayne | torgnyw, what does 'dahdi_cfg -v' output? |
| 20:43.30 | eject_ck | p3nguin: should this work for 5 concurent callls ? |
| 20:43.45 | Kobaz | p3nguin: i have some notes from the last time we talked... you use chan_sccp-b is that in asterisk addons? |
| 20:44.01 | p3nguin | eject_ck: If it works for one, it should work for 5. I don't have time to look at it right this moment. |
| 20:44.11 | eject_ck | for example 1 inbound (to 300) and 4 outbound (from other internal extensions like 301, 302, 303, 304) |
| 20:44.33 | p3nguin | kobaz: chan_sccp-b is a 3rd party channel driver. You can find it on sourceforge. |
| 20:44.34 | eject_ck | p3nguin: thank you! I will check it tomorrow in real worls |
| 20:44.36 | eject_ck | world |
| 20:44.46 | Kobaz | ah okay |
| 20:44.56 | Kobaz | is that the recommended one to use? |
| 20:45.31 | p3nguin | It's the one I use. |
| 20:45.55 | p3nguin | chan_skinny (provided in Asterisk) used to be horrible. |
| 20:46.31 | Kobaz | ah |
| 20:46.33 | torgnyw | dwayne: Here is output from dhadi_cfg -v: http://pastebin.com/8PDNzisQ |
| 20:46.39 | Kobaz | what asterisk version are you running on |
| 20:49.30 | p3nguin | SEP<MAC>.cnf.xml : http://pastebin.com/cJGPggfi |
| 20:49.31 | p3nguin | XMLDefault.cnf.xml : http://pastebin.com/1KmBudvn |
| 20:49.48 | p3nguin | I use Asterisk 1.4.40 right now. |
| 20:50.27 | Kobaz | ah okay |
| 20:50.31 | Kobaz | so i should be okay with 1.6.0 |
| 20:50.34 | Kobaz | 1.6.0.26 |
| 20:50.34 | p3nguin | yes |
| 20:50.37 | Kobaz | perfect |
| 20:50.53 | p3nguin | chan_sccp-b v3 works with 1.4 and 1.6.x, but not 1.8. |
| 20:50.53 | Freeaqingme| | I currently have several register= entries in my sip config file. Now I'd like to store these in my db (mysql) instead. Any ideas on how I could do that? |
| 20:50.54 | *** join/#asterisk slim_ (~slim_@41.239.34.73) |
| 20:50.57 | *** join/#asterisk gray_ (~Gray@unaffiliated/remnant13) |
| 20:51.50 | p3nguin | The XMLDefault file is the one that I use to set the Asterisk IP address; the SEP file just reinforces the firmware version to load. |
| 20:52.26 | p3nguin | But I think you can put any of the settings from the default file into the SEP file if you want. |
| 20:54.19 | eject_ck | I'm running asterisk on xen domu Debian. How can I get timer working ? |
| 20:56.31 | *** join/#asterisk Deeewayne (~dwayne@75-150-14-49-Atlanta.hfc.comcastbusiness.net) |
| 20:56.31 | *** mode/#asterisk [+o Deeewayne] by ChanServ |
| 21:00.15 | eject_ck | I wanna get meetme working on Xen guest system |
| 21:07.16 | *** join/#asterisk bmg505 (~leon@196-209-7-254.dynamic.isadsl.co.za) |
| 21:09.13 | leifmadsen | eject_ck: you need res_timing_dahdi for MeetMe |
| 21:09.26 | leifmadsen | eject_ck: so compile and load DAHDI then ./configure in Asterisk and make install |
| 21:09.53 | eject_ck | leifmadsen: now way to get it working from packages ? |
| 21:10.00 | eject_ck | I have dahdi available |
| 21:10.03 | *** join/#asterisk jkroon (~jkroon@197.173.105.4) |
| 21:10.29 | Kobaz | p3nguin: k |
| 21:10.45 | leifmadsen | eject_ck: I don't use packages so I can't help you beyond that |
| 21:11.06 | dwayne | he doesn't even use his own package |
| 21:11.06 | *** part/#asterisk jmls (~Julian@host217-36-208-155.in-addr.btopenworld.com) |
| 21:11.20 | dwayne | leifmadsen, sorry |
| 21:11.44 | eject_ck | I have sh-3.2# /etc/init.d/dahdi restart /etc/dahdi/system.conf not found. Nothing to do. |
| 21:12.00 | eject_ck | sh-3.2# > /etc/dahdi/system.conf sh-3.2# /etc/init.d/dahdi restart Unloading DAHDI hardware modules: done Loading DAHDI hardware modules: FATAL: Module dahdi not found. |
| 21:16.18 | eject_ck | compiling dahdi |
| 21:16.29 | Kobaz | hmm |
| 21:16.36 | Kobaz | [2011-03-16 17:16:21] WARNING[13085]: loader.c:446 load_dynamic_module: Error loading module 'chan_sccp.so': /usr/lib/asterisk/modules/chan_sccp.so: undefined symbol: _ast_calloc |
| 21:16.39 | Kobaz | mismatch somewhere |
| 21:17.18 | p3nguin | Did you compile the svn on the same box where you're going to run it? |
| 21:17.28 | Kobaz | i didn't get the svn |
| 21:17.33 | p3nguin | You should. |
| 21:17.36 | Kobaz | i got the latest from the downloads at sourceforce |
| 21:18.10 | p3nguin | I personally like rev 2420. |
| 21:19.19 | Qwell | People still use chan_sccp? Why? |
| 21:19.29 | p3nguin | Did chan_skinny ever get better? |
| 21:19.41 | Qwell | 5 years ago. |
| 21:19.51 | p3nguin | It sucked a **** just a year ago. |
| 21:19.51 | Kobaz | utils/extconf.c has _ast_calloc |
| 21:23.43 | Kobaz | Qwell: response? |
| 21:28.30 | Kobaz | so i added a _ast_calloc, a copy of __ast_calloc |
| 21:28.36 | Kobaz | now it's complaining about manager_event |
| 21:28.40 | Kobaz | this might take a while |
| 21:28.51 | p3nguin | Compiling the svn isn't working well? |
| 21:28.57 | Kobaz | it builds |
| 21:29.01 | Kobaz | i can't load it just yet |
| 21:29.16 | Kobaz | probably need to make more functions static |
| 21:29.20 | p3nguin | That's too bad; it works _perfectly_ for me on 1.4. |
| 21:30.20 | Kobaz | main/manager.c there's __manager_event, probably need to add a wrapper |
| 21:30.55 | Qwell | It needs to not call internal crap directly. |
| 21:31.03 | Kobaz | i think it's not including the right header files |
| 21:31.14 | Kobaz | Qwell: that too |
| 21:31.40 | Kobaz | Qwell: anyways... what's your response for p3nguin saying skinny is crappy? |
| 21:31.52 | Qwell | <p3nguin> That's too bad; it works _perfectly_ for me on 1.4. |
| 21:31.58 | Kobaz | no no |
| 21:31.58 | Qwell | : <p3nguin> 1.4. |
| 21:31.59 | p3nguin | chan_sccp-b |
| 21:32.01 | Qwell | : 1.4. |
| 21:32.02 | Kobaz | well yeah |
| 21:32.04 | Kobaz | 1.4 |
| 21:32.10 | Kobaz | but what about 1.8 |
| 21:32.26 | Qwell | chan_skinny was practically rewritten 5 years ago. |
| 21:32.34 | p3nguin | 1.8 sure wasn't around back then. |
| 21:32.51 | Kobaz | but is skinny equivalent or better than sccp-b |
| 21:33.00 | p3nguin | If it was rewritten five years ago, why didn't the changes get into 1.4? |
| 21:33.13 | p3nguin | Just a year ago, it was terrible. |
| 21:33.13 | *** part/#asterisk slim_ (~slim_@41.239.34.73) |
| 21:33.17 | Qwell | because 1.4 was branched nearly 5 years ago? |
| 21:33.35 | Kobaz | but i don't care about 1.4 |
| 21:33.43 | Kobaz | is it functionally equivalent or better in 1.8 ? |
| 21:33.51 | Qwell | sccp-b doesn't even compile. |
| 21:33.55 | Qwell | So, you tell me. :) |
| 21:33.59 | Kobaz | killing me |
| 21:34.24 | Qwell | 2006-09-20 |
| 21:34.24 | Kobaz | say p3nguin is using sccp-b in 1.4... how does it compare to skinny in 1.8 :) |
| 21:34.29 | Qwell | That is when 1.4 was branched. |
| 21:34.55 | p3nguin | I dont' get it. 1.4 isn't in security only until next month. |
| 21:35.01 | *** join/#asterisk fauxalliance (~fauxallia@142.162.116.237) |
| 21:35.10 | Qwell | sccp-b has always been utter garbage. Extremely crashy, very few working features, dubious (at best) copyright/licensing |
| 21:35.23 | Qwell | Quite limited phone support |
| 21:36.55 | p3nguin | kobaz: All I can say is, try chan_skinny and see what happens. Judge for yourself and let me know. |
| 21:37.02 | Kobaz | so, my question is... does skinny have all the same features as sccp-b |
| 21:37.09 | Kobaz | yeah i'm going to probably compare the two |
| 21:37.54 | Kobaz | it might be a bit of a project just to get sccp to load |
| 21:37.59 | Qwell | Kobaz: even when chan_skinny was crap, sccp-b lagged behind on features. |
| 21:38.01 | Kobaz | who knows what other functions i have to monkey with |
| 21:38.27 | p3nguin | sccp-b v2 worked very well, but there wasn't much of it. v3 brought the features. |
| 21:38.36 | Kobaz | why it's trying to call manager_event instead of use the macro i don't know yet |
| 21:39.15 | Qwell | Kobaz: Because they are incompetent. :) |
| 21:40.02 | Kobaz | #ifdef CS_MANAGER_EVENTS #include <asterisk/manager.h> |
| 21:40.08 | Kobaz | mm |
| 21:41.08 | Qwell | "Let's ifdef manager events out instead of making it a config option!" "That's a great idea!" |
| 21:41.12 | Qwell | can hear the conversation now |
| 21:41.19 | Kobaz | hah |
| 21:41.28 | Qwell | I'm telling you.. Idiots. |
| 21:41.44 | Qwell | Nobody that has maintained that project in the last 6 years has known what they've been doing. |
| 21:42.01 | Qwell | It's been forked a half dozen times |
| 21:42.36 | Qwell | They don't even have separate branches for Asterisk versions. It's just ifdefs EVERYWHERE. |
| 21:42.49 | RypPn | I'd actually second that opinion, as an sccp-b user for 4 years, just waiting on my first poly arriving any day now |
| 21:43.24 | RypPn | Sick of the blowups since Frederico got ousted |
| 21:43.26 | p3nguin | Did you upgrade to v3, or do you still use v2? |
| 21:43.41 | Qwell | RypPn: It was bad even when Sergio was "maintaining" it. |
| 21:44.16 | *** join/#asterisk eugeneoden (~goden@conference/pycon/x-jeoumqvwzauhdope) |
| 21:44.18 | RypPn | p3nguin I've used both, but 3 aint ready for production imho |
| 21:44.32 | p3nguin | yeah, but at least it has features. |
| 21:44.56 | Kobaz | okay, it was using the wrong include files |
| 21:45.03 | Kobaz | now it's using the right include files and doesn't compile |
| 21:45.30 | RypPn | I'm not bashing their efforts, it's a huge task to take on. I'm just weary of it all |
| 21:45.39 | Qwell | RypPn: It's really not that much of an effort, heh. |
| 21:45.51 | Qwell | *fixing* it would be. adding features is not. |
| 21:46.11 | Kobaz | adding features is the easy part |
| 21:46.16 | Qwell | is probably the only person outside of Cisco that even understands the protocol. |
| 21:46.28 | RypPn | That would assume they had a stable base to bolt these things on in a logical modular way |
| 21:46.36 | Qwell | RypPn: they do. chan_skinny. ;) |
| 21:47.01 | Qwell | If people would stop forking the damn thing, and put their effort into that instead...we'd all be far better off. |
| 21:47.05 | Qwell | </rant> |
| 21:47.21 | RypPn | You've probably hit it right on the head there |
| 21:47.29 | Qwell | I hit that nail 6 years ago. |
| 21:47.30 | *** join/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 21:48.10 | Qwell | They wouldn't work with the Asterisk developers, so I said screw them, and fixed a good deal of chan_skinny by myself (and then had others join me after a while to improve things even more). |
| 21:48.23 | Kobaz | Qwell: basically i would like something that can: make calls, do blf/sla |
| 21:48.29 | Qwell | Kobaz: chan_skinny |
| 21:49.10 | Kobaz | and something that doesn't crash :) |
| 21:49.21 | Kobaz | so okay i'll give skinny a shot once i give up on this build |
| 21:49.58 | Qwell | RypPn: http://lists.digium.com/pipermail/asterisk-dev/2006-April/019678.html |
| 21:50.25 | Kobaz | In file included from chan_sccp.c:25: /usr/include/asterisk.h:42: error: conflicting types for 'ast_config_AST_CONFIG_DIR' |
| 21:50.30 | Qwell | RypPn: so, 5 years ago |
| 21:52.13 | p3nguin | I'm a bit unhappy it never made it down the pipe to 1.4. There's no reason it couldn't have, considering 1.4 is still maintained and doesn't go into security only until next month. |
| 21:52.18 | Kobaz | okay i give up, I don't want to fix other peoples code and makefiles right now |
| 21:52.19 | *** join/#asterisk fauxalliance (~fauxallia@142.162.116.237) |
| 21:52.34 | Qwell | p3nguin: because the work done was massive. |
| 21:52.35 | RypPn | Thats a shame, I'd have been up for a bit of that too, too much water under the bridge now, these things are hitting ebay, lol |
| 21:52.46 | Qwell | I rewrote a good portion of it. |
| 21:52.52 | *** join/#asterisk emora (~emora@213.37.32.74.static.user.ono.com) |
| 21:53.38 | Qwell | ANYWAYS, I'm done ranting. I just wish this crap would stop being spread. It was fixed *5 years ago*. |
| 21:54.00 | Freeaqingme| | What tz are most of the regulars in here in? |
| 21:55.26 | Qwell | Freeaqingme: timezones aren't important here. most of us don't sleep. |
| 21:56.18 | Kobaz | Qwell: yeah that's the other thing... if there's an issue it's nice to come in here and be like... hey, it's borken |
| 21:56.43 | *** join/#asterisk svm_invictvs (~patrick@unaffiliated/svminvictvs/x-938456) |
| 21:56.44 | svm_invictvs | Heya |
| 21:57.12 | svm_invictvs | I was curious if it was possible to program my PBX to call soembody, detect a voice somehow and then have it call another number and connect the two calls. |
| 21:57.27 | p3nguin | Sure it is. |
| 21:57.40 | svm_invictvs | Is there some examples of that? |
| 21:57.46 | svm_invictvs | (No i'm not a telemarketer) |
| 21:58.15 | p3nguin | I don't know about examples, but you can look at all of your applications on Asterisk and get some ideas of what you have to work with. |
| 21:58.29 | Kobaz | AMD() |
| 21:58.41 | p3nguin | and WaitForSilence() |
| 22:00.08 | Kobaz | oh. that's why chrome crashed |
| 22:00.12 | Kobaz | i dont have swap turned on |
| 22:05.31 | Kobaz | p3nguin: those two pastes you wrote before with cisco phone config examples... they aren't accessable |
| 22:05.51 | *** join/#asterisk WIMPy (~wimpy@e183095026.adsl.alicedsl.de) |
| 22:08.40 | p3nguin | kobaz: hmm |
| 22:08.53 | p3nguin | I'll repaste. |
| 22:09.20 | p3nguin | Looks like they expired. |
| 22:09.24 | Kobaz | dankee |
| 22:09.24 | Kobaz | ah |
| 22:09.50 | *** join/#asterisk quintana (~sylvain@aghnar.doowan.net) |
| 22:10.38 | p3nguin | http://pastebin.com/yx8BABEA |
| 22:11.33 | p3nguin | http://pastebin.com/KfGDSgz5 |
| 22:12.36 | Kobaz | allrightey |
| 22:12.39 | Kobaz | so i have that stuff in |
| 22:13.04 | eject_ck | I've build lastest 1.8.2 *. I'm able to register on server on SIP, but I'm getting strange warning |
| 22:13.05 | eject_ck | [2011-03-17 00:11:45] WARNING[23350]: chan_sip.c:13843 register_verify: Failed to parse contact info |
| 22:13.11 | Kobaz | other than changing the nodename ip address? what should i edit? |
| 22:13.12 | Qwell | eject_ck: upgrade |
| 22:13.13 | eject_ck | should I be afraid ? |
| 22:13.36 | eject_ck | Qwell: what should I upgrade ? |
| 22:13.43 | Qwell | Asterisk |
| 22:13.52 | eject_ck | :) |
| 22:13.58 | eject_ck | Just compiled from sources |
| 22:14.05 | Qwell | 1.8.2 is not the latest version |
| 22:14.48 | eject_ck | 1.8.3 sorry |
| 22:14.55 | p3nguin | kobaz: Make sure your sccp version number matches the version referenced in the file. The files tell the phone which version to load from the tftpd. |
| 22:15.01 | Kobaz | Qwell: you know what would be cool |
| 22:15.09 | Qwell | eject_ck: https://issues.asterisk.org/view.php?id=18982 |
| 22:15.11 | Kobaz | Qwell: finger @asterisk.org for version status |
| 22:15.17 | Qwell | umm |
| 22:15.21 | Qwell | umm |
| 22:15.25 | Qwell | umm |
| 22:15.30 | Qwell | that would be cool |
| 22:15.36 | Kobaz | like finger @kernel.org |
| 22:15.41 | Qwell | we don't control that box though |
| 22:15.45 | Qwell | russellb: ^^^! |
| 22:16.25 | Kobaz | sure would beat having to load up a web page to see the latest version so i can switch the svn branch i work off of |
| 22:16.30 | Qwell | yeah |
| 22:16.37 | p3nguin | lynx -dump http://www.kernel.org/kdist/finger_banner |
| 22:16.42 | Kobaz | The latest mainline 2.6 version of the Linux kernel is: 2.6.38 |
| 22:16.46 | Kobaz | mm, sexy |
| 22:16.55 | Qwell | p3nguin: something like that might work |
| 22:17.21 | p3nguin | That banner sure has grown. |
| 22:17.24 | Kobaz | haha |
| 22:17.28 | Kobaz | a little |
| 22:17.31 | p3nguin | It used to be like six lines. |
| 22:17.40 | p3nguin | Now it's 15. |
| 22:18.03 | Kobaz | p3nguin: okay. what's the spot that has the version number that needs to change |
| 22:18.17 | p3nguin | The loadInformation lines. |
| 22:18.34 | Kobaz | k |
| 22:18.38 | p3nguin | P00308010200 would be sccp 8.1.2 |
| 22:18.41 | Kobaz | so i have a 7940 |
| 22:18.47 | p3nguin | or 8.12, can't remember |
| 22:18.51 | Kobaz | how do i know what sccp the phone is running? |
| 22:19.31 | Kobaz | i found app load id on the phone config screen |
| 22:19.41 | Kobaz | P00308000400 |
| 22:19.54 | p3nguin | It doesn't really matter what the phone is currently using because you're probably going to want to put the files on the tftpd anyway. Match the file version to the loadInformation. |
| 22:20.01 | Kobaz | k |
| 22:20.05 | Kobaz | well i dont have any firmware yet |
| 22:20.37 | p3nguin | If you don't have the files on the tftpd, it could take a while to load up, and it may never set your Asterisk address into call manager 1 slot. |
| 22:20.46 | Kobaz | k |
| 22:21.14 | eject_ck | Qwell: should I try trunk ? |
| 22:21.19 | p3nguin | If you don't have any firmware, try taking out the version number completely. |
| 22:21.27 | Qwell | eject_ck: no, look at the change in the bug report I mentioned |
| 22:21.49 | p3nguin | So where I have P00308010200, just remove P00308010200 totally and let the phone pull the file with no loadInformation. |
| 22:22.08 | p3nguin | It could load quickly and just use what it already on the phone. |
| 22:22.48 | Kobaz | k, took out the whole line for loadInformation |
| 22:22.58 | eject_ck | Qwell: not fixed yet ? |
| 22:23.02 | Kobaz | it's still trying to hit 172.20.6.200 |
| 22:23.10 | p3nguin | ... or you culod try that. |
| 22:23.13 | p3nguin | could |
| 22:23.28 | Kobaz | i'll add option 150 to my dhcp too |
| 22:23.30 | Kobaz | i never did that |
| 22:28.25 | Kobaz | should 150 be type text? |
| 22:30.39 | p3nguin | no clue |
| 22:31.29 | p3nguin | I apparently don't have 150 in my dhcpd.conf after all. |
| 22:31.33 | p3nguin | option tftp-server-name "asterisk.local"; |
| 22:31.41 | p3nguin | That's all I have for it. |
| 22:32.10 | Qwell | That is 150. |
| 22:32.17 | p3nguin | That should be 66. |
| 22:32.34 | RypPn | my entry in dhcpcd.conf is in this format http://pastebin.com/9hJLZE6f |
| 22:32.43 | RypPn | -c |
| 22:33.07 | *** join/#asterisk DanFromUK (DanFromUK@2.27.7.192) |
| 22:33.14 | p3nguin | tftp-server-name should be 66, and 150 is, well, 150. |
| 22:33.41 | Qwell | 150 is address |
| 22:33.53 | DanFromUK | Hi All, does anyone know if theres a way to detect if all members of a queue are unavailable, before adding a call to the queue. I dont want to use autopause because i need the member to unpause automatically when they come back online. |
| 22:34.07 | Kobaz | i have to convert the ip to hex |
| 22:34.08 | Kobaz | mm |
| 22:34.21 | Kobaz | /etc/dhcpd.conf line 17: expecting string or hexadecimal data. |
| 22:34.26 | Qwell | quote it |
| 22:34.35 | Kobaz | but then it's string and not address |
| 22:34.39 | p3nguin | Make it a string by putting double quotes. |
| 22:34.41 | RypPn | Or pastebin the file |
| 22:35.09 | Kobaz | i have an old stupid version of isc dhcp |
| 22:35.16 | p3nguin | me too |
| 22:35.26 | Kobaz | most of the options you have to find the name for |
| 22:35.27 | p3nguin | I don't have 150 and my 7940 pulls it just fine. |
| 22:35.40 | Kobaz | like in order to use option 66, you need to use: option tftp-server-name |
| 22:35.48 | p3nguin | that's right. |
| 22:35.55 | Qwell | option numbers should always work |
| 22:35.55 | Kobaz | yeah like what you have |
| 22:35.57 | Qwell | option option-66 |
| 22:36.15 | Kobaz | i've had problems with the config file parser with i try and use option numbers |
| 22:36.26 | p3nguin | I just thought I used option 150, but I really don't. |
| 22:36.36 | Kobaz | anyway, so |
| 22:36.47 | Kobaz | updated firmware in place, option 66 and 150 set |
| 22:37.04 | Kobaz | phone: requesting configuration |
| 22:37.10 | p3nguin | If the phone is set to use an alternate tftpd, you'll never force it to use anything else. |
| 22:37.44 | Kobaz | it's set to whatever factory defaults are |
| 22:37.49 | p3nguin | The on-phone setting for an alternate overrides the dhcpd. |
| 22:38.06 | Kobaz | settings network |
| 22:38.11 | Kobaz | dhcp server 192.168.51.1 |
| 22:38.17 | Kobaz | which is what it's supposed to be |
| 22:38.21 | Kobaz | bootp server: no |
| 22:38.34 | Kobaz | tftp server 1: 105.112.45.97 |
| 22:38.37 | Kobaz | oh i guess i should nuke that |
| 22:38.54 | Kobaz | i can't change it |
| 22:38.55 | p3nguin | If you aren't having luck still, watch the dhcp requests and make sure it is sending out the right information. |
| 22:39.21 | Kobaz | i have save and cancel, no edit |
| 22:39.27 | p3nguin | Go down to option 32. |
| 22:39.36 | p3nguin | Does it say yes or no? |
| 22:39.41 | Kobaz | it says no |
| 22:39.50 | *** join/#asterisk vfabi (~fabi@h2-92.faust.net.ua) |
| 22:39.52 | p3nguin | Then the dhcpd has control of it. |
| 22:39.55 | Kobaz | kj |
| 22:40.04 | Kobaz | as soon as i put in option 150 the phone stopped booting |
| 22:40.19 | p3nguin | That's probably why I don't use 150. |
| 22:40.34 | p3nguin | I couldn't remember. It's been a long time since I configured this stuff. |
| 22:41.49 | Kobaz | okay took out 150, it's getting configs from tftp now |
| 22:42.00 | Kobaz | Mar 16 18:41:34 acs-cin tftpd[19114]: tftpd: trying to get file: CTLSEP0014A9239820.tlv |
| 22:42.06 | Kobaz | Mar 16 18:41:34 acs-cin tftpd[19116]: tftpd: trying to get file: SEP0014A9239820.cnf.xml |
| 22:42.09 | p3nguin | That's good. |
| 22:43.02 | Kobaz | and it's just sitting there wanting something on 172.20.6.200 |
| 22:43.15 | p3nguin | Where does that number come from? |
| 22:43.25 | Kobaz | from the phone config |
| 22:43.31 | Kobaz | i see it in the network settings on the phone |
| 22:43.40 | Kobaz | i can't change any of those addresses though |
| 22:43.46 | Kobaz | it's call director #1 |
| 22:45.13 | Kobaz | how can i edit those options? |
| 22:48.28 | p3nguin | You can try to press **# when you're in that menu. |
| 22:48.40 | p3nguin | It may ask for a password. By default it is cisco. |
| 22:49.45 | Kobaz | **# didnt do anything |
| 22:50.00 | Kobaz | oooh |
| 22:50.03 | Kobaz | wait it did |
| 22:50.06 | Kobaz | it poped up the edit button |
| 22:50.08 | *** join/#asterisk FlaPer87 (~FlaPer87@unaffiliated/flaper87) |
| 22:50.08 | Kobaz | nice |
| 22:51.04 | *** join/#asterisk mzb (~mzb@ppp108-88.static.internode.on.net) |
| 22:51.15 | Kobaz | doh, but when i go to to the call manager settings the edit button is gone |
| 22:51.26 | FlaPer87 | hi guys, I'm trying to build an asterisk module that requires an external lib. I'd like to use the samke configure/make files of asterisk but I can't find out how to specify the -l options for the lib I need |
| 22:52.10 | Kobaz | okay |
| 22:52.22 | Kobaz | i went down to erase configuration |
| 22:52.28 | Kobaz | the old call manager settings are finally gone |
| 22:53.20 | Kobaz | ack |
| 22:53.25 | Kobaz | i already have stuff running on port 2000 |
| 22:53.47 | *** topic/#asterisk by leifmadsen -> Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 1.8.3.1 (2011/03/16), 1.6.2.17.1 (2011/03/16), 1.4.40 (2011/02/28), *-Addons 1.6.2.3, 1.4.13 (2010/01/14), dahdi-linux 2.4.0 (2010/09/01), dahdi-tools 2.4.0 (2010/09/01), libpri 1.4.11.5 (2010/11/22) -=- Visit the new official Asterisk wiki: wiki.asterisk.org -=- http://xkcd.com/806/ |
| 22:54.24 | Kobaz | k, changed the port |
| 22:54.33 | Kobaz | leifmadsen: what about a asterisk.org finger server? |
| 22:55.26 | Qwell | Kobaz: poke russellb about it. He might have more influence to get it done. |
| 22:55.29 | *** join/#asterisk l2cache (~l2cache@c-98-213-116-85.hsd1.il.comcast.net) |
| 22:55.34 | Qwell | (tomorrow) |
| 22:55.34 | Kobaz | nifty |
| 22:55.48 | Kobaz | what's the proper way to reload skinny configs |
| 22:56.02 | Qwell | skinny reload? |
| 22:56.05 | leifmadsen | doesn't even know what a finger server does |
| 22:56.12 | leifmadsen | loads up gopher |
| 22:56.12 | Qwell | leifmadsen: finger @kernel.org |
| 22:56.13 | Kobaz | i have skinny reset/set/show |
| 22:56.15 | Qwell | leifmadsen: CLI that |
| 22:56.24 | Qwell | Kobaz: umm, I dunno. module reload chan_skinny.so? |
| 22:56.29 | leifmadsen | Qwell: don't tell me what to do |
| 22:56.30 | Kobaz | yeah |
| 22:56.34 | Qwell | leifmadsen: DO EET |
| 22:56.40 | Kobaz | i thought there might have been a less module reload type of way |
| 22:56.44 | leifmadsen | Qwell: neato |
| 22:56.50 | Qwell | Kobaz: most modules are getting away from that |
| 22:56.58 | leifmadsen | Qwell: if it requires any more steps to releases though I won't be a fan |
| 22:57.19 | Qwell | leifmadsen: it would, but probably scriptable enough |
| 22:57.54 | Kobaz | leifmadsen: is there a current script you have that pulls down what is the current version of everything? |
| 22:58.08 | Qwell | I think even just a script on the downloads site that checks the latest versions in that dir would be enough |
| 22:58.08 | leifmadsen | Asterisk 1.6.1.23, 1.6.1.17.1 and 1.8.3.1 are now available to address the security issues described in AST-2011-003 and AST-2011-004. For more information please see the release announcement at http://www.asterisk.org/node/51595 |
| 22:58.11 | *** join/#asterisk serafie (~erin@207.98.195.107) |
| 22:58.48 | leifmadsen | Kobaz: yep -- http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-1.8-current.tar.gz |
| 22:59.21 | Qwell | leifmadsen: maybe your script to do that could create a file like latest-version, which is just a small list |
| 22:59.39 | p3nguin | kobaz: You put something else on the sccp port? |
| 22:59.56 | Kobaz | weird |
| 23:00.08 | Kobaz | p3nguin: i already have an in house service on port 2000 |
| 23:00.13 | Kobaz | i moved skinny to 3000 |
| 23:00.19 | Kobaz | anyways |
| 23:00.21 | Kobaz | this is weird |
| 23:00.33 | Kobaz | acs-TEST*CLI> module load chan_skinny.so Command 'module load chan_skinny.so' failed. |
| 23:00.41 | Kobaz | yet from the output it looks like skinny loaded just fine |
| 23:00.41 | Qwell | already loaded? |
| 23:00.49 | Kobaz | == Skinny listening on 0.0.0.0:3000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) |
| 23:00.54 | Kobaz | nope... i did an unload. and then load |
| 23:01.03 | Qwell | some modules don't like that |
| 23:03.01 | Kobaz | in my startup log. i did a restart |
| 23:03.05 | Kobaz | skinny seems to load okay |
| 23:03.24 | Kobaz | but then i can't do a module reload on it |
| 23:03.32 | Kobaz | 2011-03-16 19:03:18] NOTICE[20632]: loader.c:697 ast_module_reload: The module 'chan_skinny.so' was not properly initialized. |
| 23:03.51 | Qwell | weird. noload it, restart, load it manually |
| 23:03.55 | Kobaz | yeah |
| 23:04.00 | Kobaz | if i do that, it says failed to load |
| 23:04.15 | Kobaz | but prints happy info messages saying everything worked |
| 23:04.22 | Qwell | funky.. |
| 23:04.24 | Kobaz | probably have to look at the code |
| 23:04.32 | Qwell | poke me tomorrow. I've got to go pick a kid up in a second |
| 23:04.36 | Kobaz | k |
| 23:04.45 | Qwell | (just some random kid. not too important which.) |
| 23:05.04 | Kobaz | hah |
| 23:05.04 | Qwell | ((Dear FBI, I'm joking. -Qwell)) |
| 23:05.44 | Kobaz | it must be a bug in the 1.6.0.26 skinny |
| 23:05.51 | Qwell | 1.6.0? |
| 23:05.56 | Kobaz | i loaded up 1.8.2.3 |
| 23:06.03 | Kobaz | and it's more happy |
| 23:06.09 | Kobaz | hah sorry, yeah 1.6.0 for reels |
| 23:06.13 | Qwell | Color me surprised. |
| 23:06.26 | p3nguin | Current is 1.8.3.1. |
| 23:06.33 | Kobaz | i know i know |
| 23:06.55 | Kobaz | tell that to the 5000 line patch of inhouse changes to 1.6.0.26 |
| 23:07.13 | p3nguin | Even though I stay in the 1.4 branch, I'm usually a couple weeks behind on an upgrade. |
| 23:07.27 | Kobaz | i've ported everything to 1.8 actually |
| 23:07.32 | Kobaz | i just need to test the hell out of it |
| 23:07.42 | Qwell | cd 1.8/; svn merge /path/to/1.6.0/ /path/to/1.6.0-yourchanges/ .; svn ci -m "Merge changes from 1.6.0 branch." |
| 23:07.46 | Qwell | Thank me later. |
| 23:07.50 | Kobaz | haha |
| 23:07.54 | Kobaz | yeaaaah |
| 23:07.59 | *** join/#asterisk Grnd_Wire (~GroundWir@173.160.170.254) |
| 23:08.37 | Kobaz | tell that to the changes that aren't automatically mergable :P |
| 23:09.13 | Qwell | don't use non-interactive mode :p |
| 23:09.22 | Kobaz | true |
| 23:10.44 | *** join/#asterisk upp (upp@N1205.neckar.wh.tu-darmstadt.de) |
| 23:11.13 | Grnd_Wire | Can someone help me understand something? I'm trying to use "SetTransferCapability" and I see that is now deprecated. I am also seeing I should be using the CHANNEL function - but there is not a lot about it on voip-info.org. |
| 23:17.54 | serafie | Grnd_Wire: try https://wiki.asterisk.org/wiki/display/AST/Function_CHANNEL |
| 23:18.29 | Kobaz | oh |
| 23:18.39 | Kobaz | i moved the port back to 2000 and it registered |
| 23:19.10 | Kobaz | i wonder what the setting is to change the port that the phone connects to... i thought it was ethernetPhonePort but that didn't do anything |
| 23:20.27 | p3nguin | I couldn't say. I wouldn't change my sccp port for anything. |
| 23:20.43 | p3nguin | Just like I won't change an ssh port nor a sip port. |
| 23:21.04 | Kobaz | i've had to change sip ports when doing some local proxying |
| 23:21.14 | Kobaz | you just need to make sure everything matches |
| 23:22.17 | Grnd_Wire | serafie |
| 23:22.23 | Grnd_Wire | serafie: cool! going there now.. THank you |
| 23:24.13 | Grnd_Wire | bye george.. I think we got it.. |
| 23:29.28 | Grnd_Wire | Thank you all. Have a good day. |
| 23:29.30 | *** part/#asterisk Grnd_Wire (~GroundWir@173.160.170.254) |
| 23:37.13 | Kobaz | this skinny stuff is working good so far |
| 23:37.14 | Kobaz | [2011-03-16 19:32:14] WARNING[22993]: chan_skinny.c:1673 find_subchannel_by_instance_reference: Could not find subchannel with reference '13' on 'SEP0014A9239820' |
| 23:37.32 | Kobaz | calls still work but i get those |
| 23:44.36 | *** part/#asterisk sizzers (~k_killah@c-68-41-174-94.hsd1.mi.comcast.net) |
| 23:46.49 | *** join/#asterisk codefreeze-lap (~Steve_Mur@nv-69-68-103-77.sta.embarqhsd.net) |
| 23:55.20 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |