| 00:05.04 | *** join/#asterisk fisted_ (~fisted@unaffiliated/fisted) |
| 00:16.29 | *** join/#asterisk afink (~afink@207.106.66.194) |
| 00:32.16 | afink | Hello everyone, I am having trouble with SIP registration. I'm beginning to think its PAETEC and that their sip server is just unreachable. I can ping the their router interface but, their vlan interface seems dead. Nmapped it got nothing. Asterisk just shows request sent on sip show registry. |
| 00:33.20 | ChannelZ | it could be not making it out of a local firewall too but I'm not sure what the question actually is |
| 00:36.24 | ChannelZ | (or the reply not making it back past a firewall too I should say) |
| 00:37.16 | afink | ChannelZ: I am trying to get a successful register to my itsp. I am using another ITSP behind the same firewall so all of the proper ports should be open. |
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| 02:42.23 | *** join/#asterisk chigambamukoko (~chatzilla@fl-76-3-18-120.dhcp.embarqhsd.net) |
| 02:42.32 | chigambamukoko | greetings everyone |
| 02:42.35 | chigambamukoko | I wanted to add multiple IPs to the host command, something like so: "host=199.255.42.35, 158.352.69.52, 58.52.44.25" is that allowed? |
| 02:42.51 | *** join/#asterisk ilk (~ilk@unaffiliated/ilk) |
| 02:43.12 | chigambamukoko | if not, what could be the solution? |
| 02:43.51 | pabelanger | chigambamukoko: no, 1 host= per line |
| 02:43.58 | pabelanger | host=1.1.1.1/24 |
| 02:44.04 | pabelanger | host=2.2.2.2/24 |
| 02:44.10 | chigambamukoko | ah |
| 02:44.30 | chigambamukoko | thank you pabelanger |
| 02:44.37 | pabelanger | chigambamukoko: oh, wait |
| 02:44.42 | *** join/#asterisk spotter (~spotter@user-12ld063.cable.mindspring.com) |
| 02:44.43 | pabelanger | you are talking about registrations? |
| 02:44.49 | chigambamukoko | no |
| 02:45.00 | chigambamukoko | I think you got it |
| 02:45.17 | chigambamukoko | ip authentication |
| 02:45.27 | pabelanger | well, each [section] will only allow 1 host, each new one will overwrite the previous |
| 02:45.45 | *** part/#asterisk drdru (~Adium@c-69-181-167-85.hsd1.ca.comcast.net) |
| 02:46.13 | chigambamukoko | hmmm, so I could not add 3 "host=" in one section, correct? |
| 02:46.38 | pabelanger | right, asterisk will only use the last one |
| 02:46.38 | chigambamukoko | like you mentioned |
| 02:46.49 | pabelanger | I thought you were asking about ACL |
| 02:47.20 | chigambamukoko | so whats the solution if a peer is to authenticate by several ips? |
| 02:47.35 | pabelanger | host=dynamic ? |
| 02:47.47 | pabelanger | or create new [sections] for each peer |
| 02:48.00 | pabelanger | asterisk only allows 1 to 1 registrations |
| 02:49.06 | chigambamukoko | I c so they would need to use a username and password, I wanted them to authenticate by ip, obviously I'm out of luck |
| 02:49.49 | pabelanger | yup |
| 02:50.11 | chigambamukoko | thanks buddy, thank you very much |
| 03:08.08 | *** join/#asterisk drdru (~Adium@c-69-181-167-85.hsd1.ca.comcast.net) |
| 03:12.05 | phix | chigambamukoko: Good day sir! |
| 03:12.30 | chigambamukoko | hi phix |
| 03:13.00 | phix | how's it going? |
| 03:13.20 | chigambamukoko | not bad my friend |
| 03:13.28 | chigambamukoko | whats cooking with you? |
| 03:24.19 | *** join/#asterisk file (~file@asterisk/developer-and-muffin-lover/file) |
| 03:24.19 | *** mode/#asterisk [+o file] by ChanServ |
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| 04:24.15 | phix | chigambamukoko: oh just playing some LOTRO and helping out some ppl in a few linux channels :) |
| 04:24.18 | phix | you? |
| 04:25.24 | *** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew) |
| 04:32.36 | chigambamukoko | phix: setting up test accounts for termination |
| 04:32.51 | chigambamukoko | and such |
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| 05:16.24 | drdru | hello |
| 05:16.44 | drdru | does asterisk 1.8 support messaging? |
| 05:26.13 | *** join/#asterisk kikohnl (~kotis@udp019436uds.hawaiiantel.net) |
| 05:35.03 | *** join/#asterisk PrasanthPrabalan (~root@218.248.25.99) |
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| 05:44.50 | Neptu | hej |
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| 05:52.45 | Prasanth | Hi all |
| 05:53.35 | Prasanth | Is there any asterisk add-on software with an user interface for getting call logs. |
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| 05:55.02 | WIMPy | Like ooCalc? |
| 05:57.22 | *** join/#asterisk moy (~moy@189.172.62.190) |
| 05:59.55 | Prasanth | @WIMPy: can ooCalc get integrated with asterisk |
| 06:00.33 | WIMPy | It can display your CDRs in a more pleasing form. |
| 06:02.07 | Prasanth | how can i integrate Asterisk with OOCalc |
| 06:03.09 | WIMPy | You click on the open file icon. |
| 06:03.43 | WIMPy | If you want more, you should tell us, what exactely you're looking for. |
| 06:04.53 | timgws | lol, best response ever :] |
| 06:05.17 | timgws | Prasanth: what are you trying to do? |
| 06:05.45 | Prasanth | ya.. I will explain |
| 06:07.00 | Prasanth | I have installed asterisk for a call centre application. That is working fine. Now i need a user interface which will pop up while a call lands and after the call hangsup i need to enter call datails into a data base through that interface. |
| 06:07.53 | Prasanth | Can u suggest any Asterisk add-on for that |
| 06:08.26 | WIMPy | You will obviousely have to do the integration into whatever you're using there, yourself. |
| 06:08.42 | WIMPy | But it's quite easy to send messages from an AGI. |
| 06:09.08 | WIMPy | Or if you're doing it on a larger scale, it might make sense to lsten on AMI instead. |
| 06:11.19 | Prasanth | Ya sure i will. But i need to know exactly which is the best software i can use for this application. |
| 06:11.46 | WIMPy | Whatever language you're comforatble with. |
| 06:13.57 | Prasanth | Ok thanks for the reply.. I havent tried AGI. I will try that and revert. |
| 06:16.51 | Neptu | if i use the option m on dial how can i define some music to be played?? |
| 06:18.21 | Neptu | how can i see if i support mp3 play? |
| 06:18.58 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 06:19.04 | WIMPy | 'core show application Dial' |
| 06:19.35 | *** join/#asterisk Yourname` (~Yourname@unaffiliated/yourname/x-837320) |
| 06:19.53 | WIMPy | mp3 support usually requires a licence and has to be added manually. |
| 06:20.35 | Neptu | mmm where i chan check the price of the licence? |
| 06:21.27 | WIMPy | They are granted by the Fraunhofer Institute. Don't know any details. |
| 06:21.47 | Neptu | can i just play a wav? |
| 06:22.02 | Neptu | or an ogg? |
| 06:22.05 | WIMPy | Yes |
| 06:22.10 | [TK]D-Fender | Neptu: No licence\ |
| 06:22.19 | WIMPy | But there are restrictions on the format. |
| 06:22.38 | [TK]D-Fender | Only on how it is bundled. In the real world there is no implication |
| 06:22.50 | Neptu | ok |
| 06:22.57 | WIMPy | There is. |
| 06:23.07 | WIMPy | mp3 playback is only free for private use. |
| 06:23.33 | Neptu | but then again can u use ogg instead? |
| 06:23.46 | Neptu | if the problem is the encoding use anotherone... |
| 06:24.38 | WIMPy | The best idea is to convert your sounds once and save them in the format youre calls use so that Asterisk doesn;t have to convert them each and every time. |
| 06:26.10 | Neptu | as far as I see you use application=/usr/bin/mpg123 -q -r 8000 -f 8192 -b 2048 --mono -s to reproduce so why not to use other app like mplayer and ogg files |
| 06:26.12 | Neptu | ... |
| 06:26.48 | [TK]D-Fender | conversion = load |
| 06:27.04 | [TK]D-Fender | No reason to leave them in a format that forces conversion for every channel using MoH |
| 06:27.29 | [TK]D-Fender | No call channel is in OGG any more than any channel is in MP3 |
| 06:28.11 | Neptu | mmm |
| 06:30.25 | *** part/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 06:30.31 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 06:30.40 | [TK]D-Fender | oops |
| 06:31.07 | WIMPy | "Do not press that button"? |
| 06:31.20 | [TK]D-Fender | It was so red and shiny!! |
| 06:35.09 | *** join/#asterisk mintos (mvaliyav@nat/redhat/x-pjbyzxtouajrrnwq) |
| 06:58.43 | Prasanth | Can u suggest any web interface for AstDB |
| 07:02.37 | [TK]D-Fender | Prasanth: None. AstDB is BDB which the world largely doesn't care about. To biuld a whole web interface? I doubt anyone would bother. |
| 07:03.06 | [TK]D-Fender | Prasanth: Also, not sure how that relates to your previous question of call logging at the start & end of calls. |
| 07:03.34 | *** join/#asterisk AmirBehzad (~behzad@93.110.1.195) |
| 07:05.33 | Prasanth | I mean i will store the call log details in AsteriskDB and can access it remotely through a web interface. |
| 07:06.26 | [TK]D-Fender | Prasanth: Store them somewhere better suited. |
| 07:08.05 | *** join/#asterisk Transisto (~TransistO@dsl-67-204-34-145.acanac.net) |
| 07:08.11 | Prasanth | My call centre application doesnt need a highend date base as if now. AstDB is enough. |
| 07:08.54 | Prasanth | Also i dont have to deal with the integration problem with other data bases. That is why i thought to choose AstDB |
| 07:09.11 | Prasanth | I that fine?? |
| 07:09.18 | Prasanth | *Is that fine |
| 07:10.16 | [TK]D-Fender | Prasanth: About as lame a choice as you can find, but it could work. |
| 07:14.30 | *** join/#asterisk irroot (~gregory@197.168.73.70) |
| 07:15.29 | Prasanth | Yes. But can i find any UI for AstDB |
| 07:15.49 | [TK]D-Fender | Prasanth: BDB <- |
| 07:17.26 | kaldemar | building something against the berkley version of astdb is a dead end. it already got changed to sqlite3 in asterisk 10. |
| 07:19.52 | kaldemar | besides there are quite a few native CDR modules in asterisk, ramming them into astdb with a DIY solution is probably the worst choice you can make. |
| 07:20.42 | [TK]D-Fender | kaldemar: He wants more than CDR which means he should be using some external SQL engine. Anything that isn't * related. |
| 07:21.59 | *** join/#asterisk singler (~singler@81-7-123-162.static.zebra.lt) |
| 07:23.15 | kaldemar | [TK]D-Fender: well, i fail to undestrand the point of circling the records through something that sounds like a user interface. |
| 07:24.09 | [TK]D-Fender | kaldemar: No he wants to be able to look at the records he'll apparently have to log himself in the dialplan |
| 07:27.22 | Prasanth | ok i understood. Plz suggest a better solution for my case. I want a data base for storing my call details and a web interface for accessing that remotely. Am a beginner in Asterisk. |
| 07:29.50 | [TK]D-Fender | Prasanth: You want information at the start of the call.. that is not CDR. You will have to log that yourself. CDR is posted only at the end of a call |
| 07:30.14 | [TK]D-Fender | Prasanth: Next CDR has a general format which is documented in your source tarball and on all of the WIKI's. |
| 07:30.30 | [TK]D-Fender | Prasanth: If you don't like that then you are again responsible for logging it your own way. |
| 07:31.00 | [TK]D-Fender | Prasanth: Each of these "manual" methods will be entirely up to you to web-ify |
| 07:31.31 | [TK]D-Fender | Prasanth: if you use standard CDR for that then there are web front-ends, most of which depend on your using a SQL engine to store * CDRs |
| 07:31.49 | [TK]D-Fender | Prasanth: SQLstorage methods are documented in THE BOOK. |
| 07:31.50 | [TK]D-Fender | ~book |
| 07:31.51 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 07:31.56 | [TK]D-Fender | Prasanth: Go read it. |
| 07:32.18 | [TK]D-Fender | Prasanth: For a sample front-end, lookup "Asterisk Stat". Google time. |
| 07:32.47 | [TK]D-Fender | Prasanth: You can lookup the GUI section of this older WIKI for some other options. |
| 07:32.48 | [TK]D-Fender | ~wikis |
| 07:32.48 | infobot | [~wikis] VoIP Wiki covering Asterisk, FreeSWITCH, TrixBox, SER, OpenSER, sipX, CallWeaver, and YATE. http://www.voip-info.org (c) Arte Marketing Inc / CommPartners |
| 07:32.54 | [TK]D-Fender | ^^^ |
| 07:33.07 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 07:34.07 | kaldemar | meh, still sounds like the requirement is more like an operator panel or just something to examine CDR afterwards. |
| 07:34.36 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:34.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 07:35.09 | [TK]D-Fender | kaldemar: He mentioned getting info at the start while it's still in progress. This is supplemental if not in place of CDR |
| 07:35.41 | [TK]D-Fender | kaldemar: Which means 100% DIY |
| 07:36.00 | Prasanth | Thanks for the reply. I will check these and revert. |
| 07:38.33 | kaldemar | [TK]D-Fender: i saw that and know it, but i got the impression that it's not necessarily the requirement. |
| 07:39.44 | [TK]D-Fender | kaldemar: Details are vague as always, though he did mention it. Whether it's a serious requirement of something he simply expects is magically easy and has ready to use tools surrounding is another matter. |
| 07:42.12 | *** join/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
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| 07:50.54 | rhce7320 | I hace a problem with Kiax on a Acer netbook (W7). It registers with asterisk, can be called, but can't place calls |
| 07:51.23 | rhce7320 | firewall & Mcafee AV off |
| 07:51.53 | bulkorok | some stuff on asterisk-cli when placing a call?! |
| 07:52.44 | rhce7320 | bulkorok: No, I think the only packets I see arrive on the iax box are the registration conversation. |
| 07:53.18 | bulkorok | then the connection seems to be ok... |
| 07:53.44 | kaldemar | rhce7320: pastebin iax debug of a registration and a call. |
| 07:59.37 | *** join/#asterisk olivier2831_ (~olivier28@gre92-3-81-56-51-177.fbx.proxad.net) |
| 08:01.33 | olivier2831_ | Has someone used function TESTTIME ? |
| 08:07.15 | rhce7320 | http://pastebin.com/F0d41HxN shows a sucessful call cimungin to Acer (2006), but no channel setup for outgoing |
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| 08:13.02 | rhce7320 | Apologies, I have a callout, I'll get back to this problem in a couple of hours. |
| 08:13.03 | kaldemar | rhce7320: that's not iax debug. iax debug is what you get on the CLI after "iax2 set debug on". give that command and "core set verbose 10", make a call and pastebin the output. |
| 08:13.38 | rhce7320 | kaldemar: tks, I'll tend to that when I return. |
| 08:14.53 | *** join/#asterisk IsUp (~nocturne@unaffiliated/isup) |
| 08:15.04 | IsUp | morning |
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| 08:15.56 | olivier2831_ | Hello |
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| 09:27.38 | dddh | hi |
| 09:28.12 | IsUp | hi |
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| 09:31.10 | phix | hi |
| 09:32.02 | ChannelZ | hi |
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| 09:40.00 | dddh | I have an asterisk with e1 card "Wildcard TE220 dual-span T1/E1/J1", what should I add to extensions.conf and dahdi-channels.conf, chan_dahdi.conf? |
| 09:40.53 | kaldemar | ~book |
| 09:40.53 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 09:42.00 | dddh | I'll read book, just not sure where to begin from |
| 09:42.35 | dddh | I never had e1 cards before, used SIP all the time |
| 09:44.21 | drdru | MESSAGE is supported in asterisk 10, yet I keep getting "Unsupported Media Type" when I try to send a message |
| 09:44.39 | drdru | what should a dialplan look like that will accept messages and calls? |
| 09:51.17 | IsUp | dddh: read book. also best way to start is installing DAHDI first. |
| 09:52.02 | IsUp | dddh: dahdi_genconf will create a config file for you. just basics. if you need specific service/type/framing you need to edit your conf file. |
| 09:56.31 | *** join/#asterisk mirelab (~mirko@212.200.146.253) |
| 09:57.05 | mirelab | heelo all |
| 09:57.41 | mirelab | does anyone know why ; membermacro=somemacro in queues.conf don't work on my 1.8.7 Ast version? |
| 10:02.05 | kaldemar | mirelab: based on that, no. |
| 10:02.27 | kaldemar | mirelab: you need to provide some more information. the configuration, CLI output of a call... |
| 10:02.55 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 10:03.33 | mirelab | kaldemar: macro is called macro-CCsimulateConf so I set ; membermacro=macro-CCsimulateConf is specific queue that i wanted |
| 10:04.11 | mirelab | kaldemar: but when member of a queue answers to caller , that macro is not invoked |
| 10:04.58 | *** join/#asterisk simplydrew_ (~simplydre@unaffiliated/simplydrew) |
| 10:05.17 | kaldemar | mirelab: you probably need to change membermacro=macro-CCsimulateConf to membermacro=CCsimulateConf for starters. |
| 10:05.55 | mirelab | kaldemar: I tried that at first but didn\t work |
| 10:06.42 | mirelab | kaldemar: but we have just figured that when a caller enters a queue one macro is inwoked |
| 10:07.27 | kaldemar | feel free to pastebin something to look at. queues.conf and CLI output of a call are a good start. |
| 10:07.34 | kaldemar | ~pb |
| 10:07.34 | 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. |
| 10:07.53 | mirelab | Queue(...,...,...,,,somemacro) which is overriding the membermacro :( |
| 10:10.49 | *** part/#asterisk mirelab (~mirko@212.200.146.253) |
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| 10:20.15 | *** join/#asterisk OneNarrowWay (~OneNarrow@ip4da1344b.direct-adsl.nl) |
| 10:20.55 | k-man | I have a cisco 7942 that I want to use with asterisk, how do I get the sip firmware onto it? |
| 10:22.13 | Dovid | k-man: http://h6315.com/pub/cisco/ |
| 10:23.12 | olivier2831_ | May I ask "has someone used function TESTTIME" ? |
| 10:23.33 | k-man | olivier2831_, yes, you may |
| 10:23.59 | kaldemar | and most likely someone has. |
| 10:24.04 | k-man | Dovid, how do I upload the firmware to it? |
| 10:24.08 | olivier2831_ | Fine, so "has someone used function TESTTIME" ? |
| 10:25.49 | olivier2831_ | This TESTTIME is said to change current time for current channel so that you can test an IVR, for example, on monday, as if you were running it on sunday. |
| 10:28.37 | olivier2831_ | I did try any possible combination I could think of, but could not make it work. |
| 10:29.05 | dddh | IsUp: I have dahdi installed, but "dahdi show channels" doesn't show them |
| 10:30.50 | IsUp | dddh: pastebin your dahdi_hardware output, also your dahdi/system.conf and chan_dahdi.conf |
| 10:37.04 | *** join/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:37.23 | dddh | IsUp: seems like the problem is that I do not know what should be in chan_dahdi.conf |
| 10:37.45 | *** part/#asterisk c0rnoTa (~c0rnoTa@78.24.154.190) |
| 10:37.59 | IsUp | dddh: run dahdi_hardware... |
| 10:38.05 | *** join/#asterisk kaushal (~kaushal@49.248.16.122) |
| 10:38.07 | dddh | IsUp: http://pastebin.com/mQieX0A4 |
| 10:38.13 | kaushal | Hi |
| 10:38.18 | IsUp | kaushal: hi |
| 10:38.22 | rhce7320 | I posted a Kiax 1-way problem abt 2 hrs ago. There is now a debug of the call test on thebottom of http://pastebin.com/AgGd3Bp6 |
| 10:38.32 | kaushal | IsUp: Hi |
| 10:38.52 | IsUp | dddh: okay so you are using PRI? |
| 10:41.41 | dddh | IsUp: if I understand what is PRI correctly then yes.. |
| 10:43.04 | kaushal | IsUp: as per http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 10:43.22 | kaushal | can cancel callstatus be configured in Asterisk ? |
| 10:43.34 | kaldemar | dddh: did you configure chan_dahdi.conf in asterisk? |
| 10:43.51 | IsUp | dddh: okay pastebin chan_dahdi.conf |
| 10:43.54 | kaushal | I mean when i check for cdr-csv i dont see "cancel" condition |
| 10:44.17 | dddh | kaldemar: no, I do not know what should be in there |
| 10:44.33 | dddh | IsUp: chan_dahdi.conf is from default debian installation |
| 10:44.52 | dddh | I tried samples from digium's pdf, but seems like it didn't help |
| 10:45.25 | kaldemar | dddh: you must configure the channels in asterisk too. look at a sample configuration file for example here: http://svn.digium.com/svn/asterisk/tags/1.8.8.0/configs/chan_dahdi.conf.sample |
| 10:45.55 | kaldemar | dddh: if it still does not work, show what you have done and ask for help. |
| 10:46.35 | dddh | I have done nothing to this sample, should it work? |
| 10:46.38 | IsUp | dddh: also another example, http://pastebin.com/idyR5K1q |
| 10:47.11 | kaushal | i dont get this cancel condition when "Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up." in /var/log/cdr-csv/Master.csv |
| 10:47.17 | kaushal | Any clue ? |
| 10:47.28 | kaushal | as per http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 10:49.42 | kaushal | whereas it says "NO ANSWER" instead of "CANCEL" |
| 10:50.36 | *** join/#asterisk netman (netman@50.229.76.188.dynamic.jazztel.es) |
| 10:51.40 | *** join/#asterisk bulkorok (~bulkorok@85.183.36.36) |
| 10:53.15 | kaushal | Please let me know if anyone needs more information |
| 10:54.27 | IsUp | kaushal: as far i know, you have to set DIALSTATUS in your CDR userfield. Because main concept is, DIAL is an application and DIALSTATUS returs from Dial application. So DIALSTATUS doesnt apply for a "call". It's just an info for call leg. |
| 10:54.52 | kaushal | ok |
| 10:55.15 | IsUp | kaushal: So basicly, when you run Dial(bla/bla), it doesnt affects your CDR. caller is already ANSWERED and executing Dial application. |
| 10:55.43 | kaushal | ok |
| 10:56.15 | IsUp | kaushal: in 1.4, it's SetCDRUserField(value), but its changed as far i know. so you can run SetCDRUserField(${DIALSTATUS}) and see how it goes |
| 10:56.31 | kaushal | IsUp: ok |
| 10:56.41 | IsUp | kaushal: ok, good luck |
| 10:59.54 | kaushal | IsUp: let me pastebin /etc/asterisk/cdr_custom.conf |
| 11:00.46 | kaushal | IsUp: http://pastebin.ubuntu.com/785566/ |
| 11:00.47 | *** join/#asterisk Cain (~Geek@unaffiliated/cain) |
| 11:00.59 | kaushal | so you said SetCDRUserField(${DIALSTATUS}) |
| 11:01.08 | kaldemar | dddh: no, a sample will not work. you need to configure your system. |
| 11:01.10 | kaushal | did not understand about it |
| 11:01.37 | IsUp | kaushal: SetCDRUserField is an application. you have to run it in your dialplan. |
| 11:01.41 | dddh | kaldemar ;( |
| 11:01.45 | dddh | IsUp: "PRI Error on span 0: We think we're the CPE, but they think they're the CPE too" |
| 11:01.49 | kaushal | ok |
| 11:02.04 | kaldemar | dddh: where are you connecting to= |
| 11:02.25 | dddh | kaldemar: some station in another room |
| 11:02.34 | IsUp | dddh: as kaldemar said, you have to configure your system. and please provide more info. whats your endpoint? whats that some station? |
| 11:02.53 | dddh | IsUp: what should I ask them about? |
| 11:03.13 | dddh | kaldemar: those guyz work in next room from here ;) |
| 11:03.36 | kaldemar | dddh: pretty vague. anyway, one side must be the network side and the other the customer premises side (CPE). change your signalling to pri_net. also, you should change span=1,1,0... to span=1,0,0... in system.conf. |
| 11:04.14 | dddh | "pri_find_dchan: No D-channels available! Using Primary channel 47 as D-channel anyway!" |
| 11:05.23 | dddh | kaldemar: changed span to 1,0.0 |
| 11:05.56 | IsUp | dddh: change your signalling too. in chan_dahdi.conf |
| 11:07.34 | dddh | IsUp: changed pri_cpe to pri_net |
| 11:07.56 | dddh | "We think we're the network, but they think they're the network, too." |
| 11:07.57 | dddh | ;( |
| 11:08.03 | IsUp | dddh: good, now stop Asterisk. run 'dahdi_cfg' and run Asterisk again |
| 11:08.42 | *** join/#asterisk irroot (~gregory@197.169.11.137) |
| 11:09.05 | *** part/#asterisk irroot (~gregory@197.169.11.137) |
| 11:11.02 | kaushal | IsUp: you around ? |
| 11:11.31 | kaushal | I still get "NO ANSWER" instead of "CANCEL" |
| 11:12.33 | kaushal | http://pastebin.ubuntu.com/785580/ |
| 11:12.37 | kaldemar | dddh: you have a loopback cable in the port, don't you? |
| 11:13.02 | dddh | kaldemar: I do not know, server is not here |
| 11:13.53 | kaldemar | dddh: you should know. find out. |
| 11:16.35 | kaushal | basically the caller hung up before the callee picked up. |
| 11:16.44 | kaushal | Any further clue ? |
| 11:18.00 | dddh | http://pastebin.com/n8zvT7DB |
| 11:18.03 | dddh | cries |
| 11:19.03 | kaldemar | kaushal: use Set(CDR(userfield)=${DIALSTATUS}) instead of app SetCDRUserField. |
| 11:19.26 | kaldemar | dddh: what is the system connected to now? |
| 11:22.45 | kaushal | kaldemar: still the same |
| 11:23.33 | *** join/#asterisk qakhan (~qakhan@182.185.166.208) |
| 11:23.37 | kaldemar | kaushal: what are you looking at? |
| 11:23.41 | qakhan | hi all |
| 11:23.52 | kaushal | http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS |
| 11:24.00 | kaushal | CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up. |
| 11:24.06 | kaushal | I tried that experiment |
| 11:24.29 | kaushal | it still does not log as CANCEL in the logs or CDR |
| 11:24.29 | qakhan | i need to setup an ivr which take caller name and address and save in database caller name and adderss |
| 11:24.33 | kaldemar | kaushal: what experiment? and what does your CDR entry look like? |
| 11:25.01 | kaldemar | kaushal: the disposition field will still be either ANSWERED, NO ANSWER, BUSY or FAILED whatever you do. |
| 11:25.29 | kaushal | are you specifying /etc/asterisk/cdr_custom.conf ? |
| 11:26.10 | kaushal | http://pastebin.ubuntu.com/785566/ |
| 11:27.57 | kaldemar | i'm not specifying anything, i was asking what your CDR entry looks like after a call that ends up in a CANCEL and sets the CDR userfield to that value. |
| 11:30.36 | aberrios | qakhan, so you're gonna use SRE for this? |
| 11:39.48 | kaldemar | kaushal: btw, that example on voip-info does not work since there is no channel that would execute dialplan after the caller hangs up. anything that is related to CANCEL should be done in a hangup extension. |
| 11:40.38 | kaushal | ok |
| 11:45.58 | *** join/#asterisk AmirBehzad (~behzad@93.110.1.195) |
| 11:47.08 | kaushal | kaldemar: Thanks |
| 11:48.02 | kaushal | kaldemar: so the CDR entry says "NO ANSWER" instead of "CANCEL" |
| 11:48.35 | kaushal | kaldemar: so do i need to refer http://www.voip-info.org/wiki/view/Asterisk+standard+extensions ? |
| 11:48.48 | kaldemar | kaushal: which field says that. pastebin! |
| 11:51.01 | kaushal | ok |
| 11:52.19 | dddh | yay |
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| 11:57.06 | kaushal | kaldemar: http://pastebin.ubuntu.com/785607/ |
| 11:59.35 | kaldemar | kaushal: that's not cdr_custom |
| 12:00.20 | kaldemar | kaushal: look in /var/log/asterisk/cdr-custom/callforward.csv if that's what you have configured. |
| 12:02.06 | kaushal | kaldemar: yeah |
| 12:02.30 | kaushal | line 16 in that pastebin |
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| 12:05.17 | kaushal | kaldemar: is that wrong ? |
| 12:05.24 | kaldemar | kaushal: no. what you tail there is cdr-csv, which has no userfield in it. you configured cdr-custom to write cdr in /var/log/asterisk/cdr-custom/callforward.csv |
| 12:05.54 | kaldemar | kaushal: don't tail output from cdr-cvs when cdr-custom is what you need to read. |
| 12:06.35 | kaushal | kaldemar: i do not have anything in /var/log/asterisk/cdr-custom |
| 12:06.50 | kaldemar | then you don't even have cdr_custom.so loaded. |
| 12:06.54 | kaushal | ok |
| 12:07.03 | kaldemar | module load cdr_custom.so |
| 12:07.58 | kaldemar | if you don't have the module (which is quite likely), you need to compile/install it. |
| 12:08.53 | kaushal | http://pastebin.ubuntu.com/785618/ |
| 12:08.58 | kaushal | its there already |
| 12:09.49 | kaldemar | it doesn't necessarily mean that it is configured properly. module reload cdr_custom.so |
| 12:10.39 | kaushal | ok |
| 12:15.55 | kaushal | kaldemar: http://pastebin.ubuntu.com/785626/ |
| 12:16.44 | *** join/#asterisk fromol (~n1x@mail.orient-logic.com) |
| 12:17.10 | fromol | hi guys , i need fax on asterisk , how i can do it? i need hylafax unfortunately? |
| 12:17.55 | IsUp | kaushal: 'cdr status' in CLI. |
| 12:18.12 | kaldemar | kaushal: your CDR(userfield) is empty. |
| 12:19.44 | kaldemar | IsUp: it obviously is working since he already pasted a line by cdr_custom. |
| 12:20.05 | IsUp | kaldemar: ah okay ive missed |
| 12:20.18 | kaushal | ok |
| 12:20.39 | kaushal | kaldemar: do i need to set something ? |
| 12:20.49 | qakhan | aberrios what is SRE? |
| 12:20.55 | kaushal | < kaldemar> kaushal: your CDR(userfield) is empty. |
| 12:21.55 | kaushal | or am i missing anything ? |
| 12:22.58 | kaldemar | kaushal: you need to set a value to the field, DIALSTATUS value does not just magically appear in the field. |
| 12:23.34 | kaushal | ok |
| 12:30.48 | kaushal | kaldemar: http://pastebin.ubuntu.com/785633/ |
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| 12:33.56 | kaushal | kaldemar: so the CDR entry still says "NO ANSWER" instead of "CANCEL" |
| 12:35.35 | kaushal | Any clue ? |
| 12:37.25 | kaushal | also let me know if i need to share the asterisk logs ? |
| 12:37.39 | kaldemar | kaushal: how would i know what you did? |
| 12:38.02 | kaldemar | what did you do to set the value to the field? |
| 12:38.59 | kaldemar | and the disposition field that only seem to look at will not change. it will be "NO ANSWER". |
| 12:39.05 | kaushal | ${CSV_QUOTE(${CDR(userfield=${DIALSTATUS})})} |
| 12:39.29 | kaushal | kaldemar: ok |
| 12:40.29 | kaushal | so it will be either "ANSWER" or "BUSY" or "NO ANSWER" ? |
| 12:40.56 | kaushal | I was trying CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up. |
| 12:41.42 | kaldemar | that's wrong in many ways... |
| 12:42.29 | kaushal | kaldemar: so you said use hangup extensions ? |
| 12:42.57 | kaushal | kaldemar: basically i am trying to test for various test cases |
| 12:43.00 | kaldemar | you have two choices. 1: set a value in cdr_custom to be ${CSV_QUOTE(${CDR(userfield)})} and set CDR(userfield) in dialplan. 2: set a value in cdr_custom to be ${DIALSTATUS} |
| 12:43.33 | kaushal | ok |
| 12:44.01 | kaldemar | kaushal: ${CSV_QUOTE(${CDR(userfield=${DIALSTATUS})})} is setting the particular field to be what CDR(userfield=${DIALSTATUS}) has in it. what's inside () is the name of the field, which would be userfield=${DIALSTATUS} which is no what you want. |
| 12:47.57 | kaldemar | setting DIALSTATUS directly in cdr_custom.conf does not seem to work. use CDR(userfield)=${DIALSTATUS} |
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| 12:48.45 | kaushal | ok |
| 12:49.31 | kaldemar | exten => h,1,Set(CDR(userfield)=${DIALSTATUS}) in the context where hangup extension gets executed. |
| 12:49.40 | kaushal | ok |
| 12:49.56 | kaushal | kaldemar: so dont set anything in cdr_custom.conf ? |
| 12:50.22 | kaldemar | kaushal: did i say so? |
| 12:50.32 | kaushal | kaldemar: apologies |
| 12:50.44 | kaldemar | "set a value in cdr_custom to be ${CSV_QUOTE(${CDR(userfield)})}" |
| 12:50.48 | kaushal | if i got it wrong |
| 12:51.03 | kaushal | ok |
| 12:53.44 | kaushal | kaldemar: :( |
| 12:53.46 | kaushal | no luck |
| 12:54.02 | IsUp | ... |
| 12:54.38 | kaushal | IsUp: still "NO ANSWER" |
| 12:55.03 | kaushal | :( |
| 12:55.19 | kaushal | am i doing it wrong or understand it incorrectly ? |
| 12:56.58 | kaushal | IsUp: Please suggest further |
| 12:57.38 | kaldemar | kaushal: you're probably understanding it wrong and doing it incorrectly. show what you have. cdr_custom.conf, extensions.conf and CLI output of a call. |
| 12:57.53 | kaushal | sure |
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| 13:01.43 | kaushal | kaldemar: http://pastebin.ubuntu.com/785653/ |
| 13:02.47 | Dovid | anyone have this issue? https://issues.asterisk.org/jira/browse/ASTERISK-17998 |
| 13:03.33 | kaldemar | kaushal: still nothing sets the field when the caller hangs up. use hangup extension for that. also, fix Set(CALLERID(num)), it causes you a warning. |
| 13:05.49 | IsUp | kaushal: <kaldemar> exten => h,1,Set(CDR(userfield)=${DIALSTATUS}) in the context where hangup extension gets executed. |
| 13:05.51 | kaushal | kaldemar: are you referring to [Dec 28 18:30:28] WARNING[17438]: pbx.c:9677 pbx_builtin_setvar: Set requires an '=' to be a valid assignment. ? |
| 13:06.17 | kaushal | IsUp: ok |
| 13:06.17 | kaldemar | kaushal: yes |
| 13:06.45 | kaushal | kaldemar: is this exten => 6120,1,Set(CALLERID(number)=61816120) incorrect ? |
| 13:07.27 | kaldemar | Dovid: the AGI will get a SIGHUP when the channel is hung up unless you're using DeadAGI. trap that in the AGI app. |
| 13:08.15 | kaldemar | kaushal: yes, and so is Set(CALLERID(num)). |
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| 13:08.43 | kaldemar | kaushal: but they are incorrect in different ways. |
| 13:09.26 | kaushal | kaldemar: ok |
| 13:09.34 | kaldemar | Set(CALLERID(number)=61816120) has incorrect func CALLERID usage and Set(CALLERID(num)) has incorrect app Set usage. |
| 13:09.52 | kaushal | so any correct syntax ? |
| 13:10.23 | kaldemar | kaushal: Set(CALLERID(num)=61816120) |
| 13:11.48 | kaushal | ok |
| 13:12.02 | *** part/#asterisk descention (~scott@c-71-58-188-66.hsd1.pa.comcast.net) |
| 13:13.28 | kaushal | kaldemar: so is this the correct one -> http://pastebin.ubuntu.com/785668/ ? |
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| 13:15.03 | kaldemar | kaushal: i'm about to give up on you... priority 2 is correct but 4 is incorrect. |
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| 13:21.44 | kaushal | ok |
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| 13:26.49 | Keisuke | hi all, i have a trouble with my installation of Asterisk and Dahdi for my card TDM410P with 1FXO/1FXS |
| 13:27.32 | Keisuke | i believe this subject was already solve… but i don't found the solution online. |
| 13:28.17 | Keisuke | dahdi driver is installed |
| 13:29.35 | kaushal | kaldemar: please give me a moment |
| 13:29.37 | kaushal | brb |
| 13:29.45 | Keisuke | but i think the problem is about the configuration |
| 13:30.46 | Keisuke | port 1 : FXO / port 4: FXS -> for dahdi: channel 1: FXS / Channel 4: FXO…. |
| 13:31.49 | Keisuke | i have stated my problem in a french forum, but nobody answered me |
| 13:32.23 | Keisuke | what's the english forum, where i can stated the problem ? |
| 13:32.25 | IsUp | Keisuke: yeah, because your approach is wrong. whats your trouble? details, logs, etc |
| 13:33.47 | Keisuke | hum… IsUp : can i send you the message of french forum, because i have all paste on it ? |
| 13:34.12 | Keisuke | my file config/ log |
| 13:34.32 | IsUp | ok pm me |
| 13:34.36 | Keisuke | ok, thx |
| 13:34.36 | kaldemar | Keisuke: which driver do you have loaded? do you have 1 FXS and 1 FXO module? what order are they in the base card? what do you have in system.conf? have you run dahdi_cfg? what do you have in chan_dahdi.conf? what does not work and how? |
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| 13:59.11 | LordRain | hey guys... i have a quick question |
| 13:59.29 | beek | Quick answer: 42 |
| 13:59.44 | WIMPy | perfect |
| 13:59.46 | jaytee | :-) |
| 13:59.47 | LordRain | im converting from Asterisk 1.0.08 to 1.8.x and from zaptel to dahdi (hooray, what a huge upgrade, huh?) and im trying to make sure i understand the syntax of 1.8 |
| 13:59.56 | IsUp | haha |
| 14:00.00 | LordRain | where the lines in extensions.conf would say: |
| 14:00.01 | LordRain | exten => 5043550502,1,Dial,IAX2/pbx2:pbx2@206.41.40.8/5043550502 |
| 14:00.01 | LordRain | exten => 5048281068,1,Dial,Zap/g1/3375930059 |
| 14:00.08 | WIMPy | WOW! |
| 14:00.10 | IsUp | beeeeh |
| 14:00.11 | LordRain | my assumption is they should be: |
| 14:00.12 | LordRain | exten => 5043550502,1,Dial(IAX2/pbx2:pbx2@206.41.40.8/5043550502) |
| 14:00.12 | LordRain | exten => 5048281068,1,Dial(DAHDI/g1/3375930059) |
| 14:00.14 | IsUp | BEEEEEEEH |
| 14:00.18 | LordRain | for 1.8 syntax... |
| 14:00.20 | LordRain | correct? |
| 14:00.32 | WIMPy | Looking good. |
| 14:01.23 | LordRain | sweet |
| 14:01.25 | LordRain | thank you :) |
| 14:02.03 | WIMPy | But that iax string looks overcomplicated. Maybe you should create a peer for that. |
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| 14:31.23 | Kalidarn | can i see the version of an asterisk module from within asterisk? |
| 14:31.48 | Kalidarn | im aware of "module show" |
| 14:31.50 | WIMPy | What module? |
| 14:32.17 | Kalidarn | in this particular case chan_sccp |
| 14:32.37 | Kalidarn | chan_sccp.so Skinny Client Control Protocol (SCCP). R 0 |
| 14:32.48 | Kalidarn | i can see it there but i don't get the ability to see the version loaded |
| 14:32.57 | Kalidarn | not sure if this is exposed to asterisk |
| 14:33.25 | WIMPy | Only if it provides a command itself. |
| 14:51.10 | *** join/#asterisk zerohalo (~zerohalo@74.60.136.128) |
| 14:51.12 | *** join/#asterisk fisted (~fisted@unaffiliated/fisted) |
| 14:52.51 | *** join/#asterisk LordRain (~ItsRain@1800-lan.idsno.net) |
| 14:52.58 | LordRain | sorry, i got booted earlier lol |
| 14:53.10 | aberrios | So Digium are touting cheap trial/monthly subs for LumenVox Speech Recognition but theres no mention of this on LumenVox's site. Anyone else have dealings with LumenVox? |
| 14:54.35 | *** join/#asterisk caveat- (hoax@gateway/shell/bshellz.net/x-nwikggtqeuymkkme) |
| 14:54.52 | LordRain | i still can't get over this... |
| 14:54.55 | LordRain | [root@voicemail ~]# asterisk -V |
| 14:54.55 | LordRain | Asterisk CVS-v1-0-08/10/05-13:40:31 |
| 14:54.56 | LordRain | [root@voicemail ~]# |
| 14:54.56 | LordRain | lol |
| 14:54.59 | LordRain | priceless. |
| 14:56.43 | IsUp | :( |
| 14:56.57 | *** join/#asterisk brah (be88a59e@gateway/web/freenode/ip.190.136.165.158) |
| 14:57.46 | aberrios | =D |
| 15:03.14 | *** join/#asterisk _Corey_ (~chatzilla@64.121.4.75) |
| 15:03.41 | *** join/#asterisk putnopvut (~putnopvut@asterisk/master-of-queues/mmichelson) |
| 15:03.41 | *** mode/#asterisk [+o putnopvut] by ChanServ |
| 15:09.25 | IsUp | i have old Aculab cards (ISA and PCI), i am planning to sell them. any ideas? does anyone using Aculab around here? |
| 15:10.32 | *** join/#asterisk FiReSTaRT (~dlyh@unaffiliated/firestart) |
| 15:11.11 | WIMPy | CAPI only, I presume? |
| 15:12.41 | IsUp | well i dont know what is CAPI but, it's T1/E1 |
| 15:12.56 | IsUp | same item: http://www.ebay.com/itm/Aculab-AC2580-2T1-AC-2580-/380151409996 |
| 15:13.03 | WIMPy | Driver |
| 15:13.56 | IsUp | so what you think these cards are dead? :p |
| 15:14.02 | WIMPy | That's a big thing. |
| 15:14.49 | IsUp | i found a package in our IT room. i have about 15 cards |
| 15:14.51 | WIMPy | chan_capi is only a development version for Asterisk 1.8 so far. |
| 15:15.17 | IsUp | WIMPy: aculab uses own driver for Asterisk as far i remember. we were running SS7 with these cards. |
| 15:15.28 | WIMPy | And the next question would be what the latest Linux version is that's supported by the drivers. |
| 15:15.54 | WIMPy | So not even CAPI but somethign completely different? |
| 15:16.02 | IsUp | WIMPy: it's chan_aculab |
| 15:16.37 | WIMPy | Oh, yet another one :-( |
| 15:16.51 | WIMPy | And what is underneath? |
| 15:17.44 | IsUp | WIMPy: it was working fine. but yeah, its old. i'll talk to Aculan techs for new driver support. i dont think its supports 1.8 anyways |
| 15:18.14 | WIMPy | That's always teh issue with old hardware with proprietary drivers. |
| 15:18.54 | WIMPy | Do you know what kind of (hardware) drivers they have? |
| 15:19.11 | *** join/#asterisk seraphie (~erin@75.76.38.159) |
| 15:19.58 | IsUp | umm no but i have card front of me, i can tell what you need |
| 15:20.17 | *** join/#asterisk Cupidon (laci@c-4f661cfc-74736162.cust.telenor.se) |
| 15:20.26 | WIMPy | lsmod might be interesting. |
| 15:20.44 | IsUp | ah, card is not plugged in any PC |
| 15:20.49 | IsUp | lemme show you my storage haha |
| 15:20.50 | *** part/#asterisk asterisk-Tester (~RAMYT@193.227.170.247) |
| 15:20.59 | WIMPy | Ah, ok. |
| 15:22.04 | WIMPy | He, their stuff is from june 2011. Looks like they're still active. |
| 15:22.28 | IsUp | they are active yes. aculab is a great company actually. but i prefer Sangoma atm :p |
| 15:23.13 | WIMPy | The old cards were usually of very good quality, but drivers are usually an issue. |
| 15:25.20 | WIMPy | Oh, you need an account with Aculab to install the software. |
| 15:26.24 | *** join/#asterisk leifmadsen (~Leif@asterisk/documenteur-extraordinaire/blitzrage) |
| 15:26.24 | *** mode/#asterisk [+o leifmadsen] by ChanServ |
| 15:27.59 | IsUp | WIMPy: it doesnt matter i think. they are providing software |
| 15:28.32 | WIMPy | Yes, there seems to be a guest account that can be used. |
| 15:29.39 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 15:30.43 | IsUp | WIMPy: http://imageshack.us/photo/my-images/267/img7942b.jpg/ |
| 15:32.40 | WIMPy | Oh, where I see the one with the 8 LEDs... We used to have a PRI card that had a full 32 LEDs on the back. But I have NFI what kind of thing that was. |
| 15:32.54 | WIMPy | It had been used for fax2net. |
| 15:33.24 | IsUp | well all cards are different. some has a "bridge" cable. 3 cards were bridged together. like ATI Crossfire :p |
| 15:33.54 | WIMPy | Most cards have that option. |
| 15:34.09 | IsUp | WIMPy: http://imageshack.us/g/204/img7944a.jpg/ |
| 15:34.10 | *** join/#asterisk treborsux (~IceChat77@75-144-117-117-Jacksonville.hfc.comcastbusiness.net) |
| 15:34.25 | treborsux | How do i make dids for my analog card? |
| 15:34.30 | WIMPy | But I can't find much information about the Aculab software or what it supports. |
| 15:34.37 | treborsux | I need a channel to ring on an extension |
| 15:34.46 | IsUp | should i put them on eBay? what you think? |
| 15:34.50 | treborsux | How do i know what did is each channel |
| 15:34.54 | WIMPy | Alanlog doesn't have DID. |
| 15:35.12 | treborsux | so how do i make say channel 2 ring to an extension |
| 15:35.42 | WIMPy | treborsux: You put your channels in different contexts. |
| 15:35.43 | treborsux | the freepbx guys said to use the zap channel dids feuture but it has no effect on my system |
| 15:36.21 | treborsux | even with the context = from-zaptel in chan-dahdi.conf it did nothing |
| 15:36.30 | WIMPy | IsUp: That's a gamble. If someone needs such a thing you may get a very good price, but for Joe Average thay are probaly next to worthless. |
| 15:36.33 | IsUp | treborsux: you are running FreePBX right? i am sure they can help you better |
| 15:36.53 | treborsux | yes i did your first statement |
| 15:37.04 | WIMPy | Zaptel has been replaced by dahdi some years ago. |
| 15:37.04 | IsUp | WIMPy: okay :) |
| 15:37.09 | WIMPy | ~freepbx |
| 15:37.09 | infobot | [~freepbx] FreePBX is unable to be supported here. It is made up of complex dialplans and scripts which can't be easily supported by people who aren't deeply involved. Try joining #freepbx and asking there |
| 15:37.41 | *** join/#asterisk AmirBehzad (~behzad@79.127.53.248) |
| 15:38.43 | WIMPy | IsUp: If you can find information on how to use them, especially with Asterisk, that would certainly help. |
| 15:39.01 | treborsux | i came here to see if there was a place to manyually assign the channels a did since it does not work on free pbx |
| 15:39.12 | IsUp | WIMPy: yes but i think Aculab is in Christmas holiday |
| 15:39.47 | WIMPy | treborsux: Doing things manually with FreePBX installed is asking for trouble. |
| 15:39.58 | IsUp | WIMPy: anyways, ill get PN and SN for all cards. heavy work :) |
| 15:41.59 | WIMPy | Hmm. Do I get that right? Those cards disguise as ethernet? |
| 15:42.47 | WIMPy | Err, no. They have an ethernet port that you connect? |
| 15:43.16 | WIMPy | Are they like these Asterisk cards (can't remember the name) that only need the PC for power supply? |
| 15:44.16 | WIMPy | Certainly sounds interesting. |
| 15:45.38 | IsUp | well i cant really remember. i am taking photos now for all items |
| 15:45.44 | IsUp | i can send you details when i finish |
| 15:46.14 | WIMPy | I'm reading the installation guide. |
| 15:48.28 | IsUp | okay, another card: Prosody PCI-2 PM4, ACS2470 |
| 15:55.35 | WIMPy | They have an EOL notice. |
| 15:58.06 | LordRain | is VoiceMailMain still a valid dialplan application in asterisk 1.8 ? |
| 15:58.14 | LordRain | im having no luck with google |
| 15:58.19 | LordRain | google used to be my best friend :( |
| 15:58.23 | WIMPy | Sure |
| 15:58.53 | LordRain | so could i affectively call it via: |
| 15:58.56 | LordRain | exten => 6350635,1,VoicemailMain |
| 15:58.57 | LordRain | ? |
| 15:59.16 | WIMPy | 'core show application VoiceMailMain' |
| 16:02.11 | LordRain | ty |
| 16:05.44 | LordRain | when i call an application via a dialplan, i.e. VoiceMailMain, do I have to follow it with a () - i.e. exten => 6350635,1,VoicemailMain() or just VoiceMailMain ? |
| 16:06.01 | LordRain | (ast 1.8) |
| 16:06.14 | WIMPy | I haven't tried without. |
| 16:07.30 | LordRain | k, i'll just put them in for good measure then lol |
| 16:07.31 | *** join/#asterisk tzafrir (~tzafrir@local.xorcom.com) |
| 16:09.39 | ChannelZ | If there's no argument, you don't have to |
| 16:11.23 | LordRain | Now what about commands as apposed to applications, i.e. Wait ? In 1.0, it appears to be Wait,<duration> - has that changed to Wait(<duration>) i.e. Wait(15) ? |
| 16:11.31 | LordRain | or would Wait,15 still apply? |
| 16:12.39 | WIMPy | All parameters in () now. |
| 16:12.41 | ChannelZ | Wait is an application so Wait(seconds) |
| 16:13.17 | LordRain | okay, so that's the major difference in syntax is that parems for all commands and applications are now expected to be in the parenthesis? |
| 16:13.28 | LordRain | thank you much for that clarification. life just got easier lol |
| 16:13.35 | *** join/#asterisk jshriver (~jshriver@72.240.39.37) |
| 16:13.40 | jshriver | Greetings |
| 16:13.42 | ChannelZ | Yes, and use , not | |
| 16:14.12 | jshriver | can someone point me to documentation on setting up asterisk with an email account for sending voicemails? i have the appropriate settings in voicemail.conf per extension but don't see where I put the email server info |
| 16:14.44 | WIMPy | You don;t it will use your local MTA. |
| 16:14.44 | ChannelZ | jshriver: you don't. It calls 'sendmail' or some other smart delivery agent who actually routes the mail somewhere interesting |
| 16:14.46 | LordRain | jshriver: im using the local sendmail for that... |
| 16:14.53 | ChannelZ | see mailcmd |
| 16:15.08 | jshriver | ug so I have to run my own email server? what if you don't have a domain name |
| 16:15.08 | LordRain | as long as your sendmail is configured correctly, it should send the message out. |
| 16:15.14 | ChannelZ | If you don't have a mailer on your server, get something like msmtp |
| 16:15.17 | WIMPy | Otherwise you have to specify a full command to send by other means. |
| 16:16.06 | jshriver | hrm no idea |
| 16:16.26 | jshriver | does it use sendmail by default? |
| 16:16.33 | ChannelZ | With MSMTP installed, I just use mailcmd=/usr/bin/msmtp -t -f asteriskvm@mydomain.com |
| 16:16.34 | jshriver | or where do I specify what email agent to use. |
| 16:16.41 | ChannelZ | it's in /etc/asterisk/voicemail.conf |
| 16:17.11 | ChannelZ | (and then msmtp is configured via its config file on an actual SMTP server to contact, etc.) |
| 16:18.27 | jshriver | appreciate the help off to read and tinker |
| 16:19.56 | ChannelZ | enjoy |
| 16:39.32 | jshriver | hrm looks like I set sendmail up and use gmail as a SmartAuth mail relay |
| 16:39.35 | jshriver | fun stuff :) |
| 16:41.13 | LordRain | it makes me sad that google voice won't let me set the CID :( |
| 16:41.18 | LordRain | lol :) |
| 16:52.16 | *** join/#asterisk kresp0 (~kresp0@112.200.217.87.dynamic.jazztel.es) |
| 16:53.45 | *** join/#asterisk [TK]D-Fender (~chatzilla@64.235.218.194) |
| 16:56.26 | *** join/#asterisk grapsus (~grapsus@rai21-4-88-179-127-17.fbx.proxad.net) |
| 16:56.30 | grapsus | hi there |
| 16:56.51 | grapsus | I have a really weird problem with call transfer |
| 16:57.19 | WIMPy | What channeltype? |
| 16:57.30 | grapsus | I have tT options, DTMF signals recognized by asterisk (logger with DTMF) |
| 16:57.43 | grapsus | the call comes from SIP |
| 16:57.57 | grapsus | and goes to a DAHDI channel |
| 16:58.08 | grapsus | and I want to park that call |
| 16:58.19 | WIMPy | Park or transfer? |
| 16:58.32 | grapsus | I tried both, but nothing works |
| 16:58.51 | grapsus | #700, #, #somenumber |
| 16:58.51 | WIMPy | Does your DTMF work at all? |
| 16:58.54 | grapsus | yes |
| 16:59.04 | grapsus | DTMF[3952] channel.c: DTMF end '#' received on SIP/user0-00000001, duration 100 ms |
| 16:59.09 | grapsus | I'm positive about that |
| 16:59.34 | WIMPy | But why do you use DTMF transfers? |
| 17:00.06 | grapsus | I have this c450IP phone |
| 17:00.15 | grapsus | it doesn't have any 'transfer' button |
| 17:00.34 | WIMPy | :-( |
| 17:01.59 | grapsus | WIMPy: #700 should park the call right ? |
| 17:02.34 | grapsus | it works with a Cisco SPA504G, so my call parking is working |
| 17:02.35 | WIMPy | See features.conf |
| 17:04.48 | [TK]D-Fender | grapsus: pastebin the complete failed call from beginning to end along with your dialplan. |
| 17:05.10 | [TK]D-Fender | ~pb |
| 17:05.10 | 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. |
| 17:08.32 | grapsus | [TK]D-Fender: I know what is pastebin thanks |
| 17:08.50 | grapsus | there's nothing special, when I press #700 just nothing happens |
| 17:08.58 | grapsus | excepted for the [DTMF] log |
| 17:10.47 | Dovid | hi all. I am using 10.0. I have an issue where every so often asterisk does not respond to a BYE |
| 17:11.34 | Dovid | are there any "known" issues? |
| 17:12.23 | grapsus | WIMPy: IT WORKS !!! blindxfer => #, did the trick |
| 17:43.25 | *** join/#asterisk Naikrovek (~mbluth@unaffiliated/naikrovek) |
| 17:47.30 | *** join/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 17:47.40 | ujjain | What is the difference between a PAP2T and a PAP2? |
| 17:55.28 | ujjain | New model it seems. |
| 17:55.42 | ujjain | Are 2nd hand wireless VOIP/SIP phones available for <$100? |
| 17:59.13 | WIMPy | Yes, but you probably don't want them. |
| 17:59.29 | *** join/#asterisk dhananjay (~dhananjay@117.213.7.47) |
| 18:00.54 | dhananjay | Im a newbie at Asterisk, Im stuck in a problem that I cannot register any SIP phones to my server. And I dont have a hunch what to search. |
| 18:01.03 | *** join/#asterisk anthm (~anthm@freeswitch/developer/anthm) |
| 18:02.41 | *** join/#asterisk tekoholic (~tekoholic@97-118-219-224.hlrn.qwest.net) |
| 18:03.53 | ujjain | WIMPy: I see :) |
| 18:07.51 | *** join/#asterisk kikohnl (~kotis@72.253.138.39) |
| 18:12.15 | *** part/#asterisk nny (~Scott@cpe-174-107-223-014.sc.res.rr.com) |
| 18:20.12 | *** join/#asterisk Defraz (~Defraz@65.101.67.10) |
| 18:22.07 | *** join/#asterisk tekoholic (~tekoholic@97-118-209-47.hlrn.qwest.net) |
| 18:31.21 | *** join/#asterisk coppice (~chatzilla@183178203098.ctinets.com) |
| 18:32.42 | *** join/#asterisk [sr] (~Unknowned@pal-213-228-181-48.netvisao.pt) |
| 18:32.48 | *** part/#asterisk [sr] (~Unknowned@pal-213-228-181-48.netvisao.pt) |
| 18:38.19 | LordRain | hey guysd |
| 18:38.20 | LordRain | odd one... |
| 18:38.28 | LordRain | im almost done moving all my configs from 1.0 to 1.8 |
| 18:38.32 | LordRain | but, i run in to this one issue: |
| 18:38.35 | LordRain | exten => 5042870008,1,Voicemail(u5863000) |
| 18:38.37 | LordRain | results in: |
| 18:38.47 | Kobaz | 1.0, oh wow |
| 18:38.50 | LordRain | WARNING[19136]: app_voicemail.c:5644 leave_voicemail: No entry in voicemail config file for 'u5863000' |
| 18:38.57 | LordRain | yeah, 1.0.08CVS lol |
| 18:38.57 | Kobaz | read the help on voicemail |
| 18:39.01 | Kobaz | core show application voicemail |
| 18:39.08 | LordRain | is it not allowing me to put the flag u anymore? |
| 18:39.26 | WIMPy | It is a flag now, yes. |
| 18:39.26 | Kobaz | Hint: Voicemail(5863000,u) |
| 18:39.43 | LordRain | oh... |
| 18:39.46 | LordRain | i see it now |
| 18:39.53 | LordRain | VoiceMail(mailbox[@context][&mailbox[@context][&...]][,options]) |
| 18:40.02 | LordRain | so u would be the option |
| 18:40.13 | Kobaz | yeap, it's an option |
| 18:40.13 | LordRain | hence, ,u ... |
| 18:40.17 | LordRain | yeah, a LOT has changed, aye? ;x |
| 18:40.20 | Kobaz | you'll run into a number of those types of changes |
| 18:40.48 | LordRain | well, this should be the last of it that i need to get done :) |
| 18:40.57 | LordRain | and it'll be off of 1.0 and on to the latest... lol |
| 18:41.04 | LordRain | what a headache... |
| 18:41.38 | ChannelZ | welcome to 2010 |
| 18:41.48 | Kobaz | 2012 ? |
| 18:41.58 | ChannelZ | He said 1.8... |
| 18:42.06 | Kobaz | mm, right |
| 18:42.18 | ChannelZ | :) |
| 18:42.39 | *** join/#asterisk simNIX (~simNIX@80.187.148.115) |
| 18:44.14 | ChannelZ | I don't even know when 1.0 came out |
| 18:44.21 | Kobaz | hmm |
| 18:44.28 | Kobaz | i haven't tried it yet but |
| 18:44.34 | Kobaz | if you have autodelete turned on for voicemail |
| 18:44.39 | Kobaz | and you don't have an email set up |
| 18:44.44 | Kobaz | i think you lose the voicemail |
| 18:44.52 | LordRain | [root@voicemail /etc/asterisk]# asterisk -V |
| 18:44.54 | LordRain | Asterisk CVS-v1-0-08/10/05-13:40:31 |
| 18:44.54 | LordRain | [root@voicemail /etc/asterisk]# |
| 18:44.57 | LordRain | apparently 05.... ? lol |
| 18:44.59 | Kobaz | like the email field is blank |
| 18:45.05 | LordRain | oh, it just deletes it? |
| 18:45.14 | Kobaz | i think so |
| 18:45.21 | LordRain | 0311 => 1234,Ambassador Hotel,warrenreuther@aol.com,,|delete=yes |
| 18:45.24 | ChannelZ | I guess Whackapedia says Sept 2004 |
| 18:45.33 | LordRain | majority of the lines have |delete=yes as the option |
| 18:45.40 | LordRain | do i need the | without having another option defined? |
| 18:45.53 | Kobaz | look at the voicemail.conf example file |
| 18:45.54 | LordRain | or could i essentially do 0311 => 1234,Ambassador Hotel,warrenreuther@aol.com,,delete=yes |
| 18:46.02 | LordRain | i dont have an example file ;[ |
| 18:46.06 | Kobaz | yeah you do |
| 18:46.07 | LordRain | id have to make samples |
| 18:46.13 | LordRain | nah, i got this handed to me as is lol |
| 18:46.14 | Kobaz | it's in the source tree |
| 18:46.20 | Kobaz | oh |
| 18:46.24 | Kobaz | well download a 1.8 tarball |
| 18:46.27 | LordRain | and it was a disaster |
| 18:46.30 | LordRain | :P |
| 18:47.38 | LordRain | well, the examples ive seen only show the | in use with multiple options |
| 18:47.51 | LordRain | but will it cause a problem if its the only option? |
| 18:48.53 | Kobaz | probably not |
| 18:51.11 | LordRain | these guys actually didnt wanna upgrade from this version |
| 18:52.37 | LordRain | but it keeps magically and randomly seg faulting |
| 18:53.01 | LordRain | with a bunch of nifty 41's (A) in the ANI buffer |
| 18:53.32 | LordRain | (buffer overflow, anyone?) |
| 19:09.46 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 19:12.54 | *** join/#asterisk flebel (~flebel@dsl-67-204-52-89.acanac.net) |
| 19:17.21 | *** part/#asterisk flebel (~flebel@dsl-67-204-52-89.acanac.net) |
| 19:18.17 | *** join/#asterisk tekoholic (~tekoholic@97-118-249-39.hlrn.qwest.net) |
| 19:19.11 | woleium | Does anyone have experience of getting busy lamps to work on Polycom handsets? |
| 19:19.25 | Naikrovek | yes, it's easy. |
| 19:19.45 | Naikrovek | asterisk needs dialplan hints set up. this is straightforward. |
| 19:19.56 | *** join/#asterisk hardwire (~spencersr@cl-36.anc-01.us.sixxs.net) |
| 19:20.10 | hardwire | k so.. runing CURL() inside exten h = you have people yell at you. |
| 19:20.24 | Naikrovek | then you need to add whomever you want a busy lamp for to the phone's local directory, give them a speed dial number, then set buddywatch to '1'. |
| 19:20.46 | woleium | I've done the buddywatdh bit |
| 19:20.47 | Naikrovek | and it's that simple |
| 19:20.57 | woleium | and the directory |
| 19:21.02 | Naikrovek | if you're on freepbx or something like it then the dialplan hints are already done probably |
| 19:21.13 | woleium | I guess I must be missing the dialplan hints |
| 19:21.25 | woleium | yeah, using piaf purple |
| 19:21.50 | Naikrovek | check documentation on that distro to work out how to set up the hints. should be pretty straightforward |
| 19:22.18 | woleium | wanders off to google |
| 19:22.22 | woleium | thanks Naikrovek |
| 19:22.24 | woleium | :-) |
| 19:23.25 | *** join/#asterisk flebel (~flebel@dsl-67-204-52-89.acanac.net) |
| 19:26.18 | *** join/#asterisk Dovid (~Dovid@static-173-63-105-18.nwrknj.fios.verizon.net) |
| 19:27.39 | *** join/#asterisk flebel (~flebel@dsl-67-204-52-89.acanac.net) |
| 19:27.41 | Dovid | hi all. when asterisk does DEBUG[19212] and VERBOSE[19212] the 19212 is unique per call? |
| 19:29.41 | Naikrovek | woleium: welcome |
| 19:31.19 | *** join/#asterisk iulius (~iulius@adsl-74-160-98-148.asm.bellsouth.net) |
| 19:33.29 | *** part/#asterisk flebel (~flebel@dsl-67-204-52-89.acanac.net) |
| 19:36.32 | *** part/#asterisk ujjain (~ujjain@unaffiliated/ujjain) |
| 19:39.49 | Kobaz | Dovid: it's the lwp process id, an it's not guaranteed unique |
| 19:44.27 | Dovid | Kobaz: Whats the LWP process? |
| 19:44.43 | [TK]D-Fender | PID <- |
| 19:44.51 | Kobaz | light weight pid |
| 19:45.09 | Kobaz | thread id basically |
| 19:46.00 | Kobaz | and i saw your chat in -dev. you *can* use it to trace debug/verbose messages as long as a call is alive, like kp says |
| 19:46.30 | Kobaz | but i wouldn't use it for anything serious, you're better off tracking uniqueid |
| 19:46.50 | Dovid | Kobaz: What I found interesting that in the same second there are two calls using the same identifuer |
| 19:47.19 | Kobaz | if calls get masq'd and etc then channels will move around to different threads |
| 19:47.46 | Dovid | kobaz: it had the same as my opensips that sends INFO packets every 40 seconds |
| 19:48.01 | *** join/#asterisk afink (~afink@207.106.66.194) |
| 19:48.28 | Kobaz | it depends on where the msg is being generated also |
| 19:49.01 | Kobaz | if there is a general purpose handler that handles things for multiple channels, like say chan_sip |
| 19:49.06 | Kobaz | then you'll always get the same thread it |
| 19:49.08 | Kobaz | id |
| 19:49.55 | Kobaz | in other words you can trace some things but not everything |
| 19:50.03 | Kobaz | and the best approach is to use a different approach |
| 19:51.24 | Dovid | Kobaz: What would you reccomend ? |
| 19:51.36 | Kobaz | look in -dev, start with sip tracing |
| 19:52.14 | Kobaz | make sure asterisk is getting what you think it's getting, removing firewalling/networking issues from the equation |
| 19:56.17 | *** join/#asterisk tekoholic (~tekoholic@97-118-211-57.hlrn.qwest.net) |
| 19:58.17 | Dovid | Kobaz: I am doing sip tracing. i was filtering the log file based on the id of the call. wondering whats the best way to filter to find a specific call? |
| 20:03.36 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:04.44 | *** join/#asterisk dre (~dre@69.38.200.246) |
| 20:08.38 | dre | i am having difficulty understanding how the rtpkeepalive setting in asterisk sip settings is interpretted. is this value the frequency between keepalives sent or a timer that dictates how long in seconds keepalives will be sent |
| 20:09.44 | WIMPy | No it's the time after the last activity until the call is considered dead. |
| 20:09.49 | WIMPy | +1 |
| 20:10.01 | *** join/#asterisk gerhard7 (~gerhard7@82-168-115-40.ip.telfort.nl) |
| 20:10.05 | *** join/#asterisk TimeRider (steve@188-220-34-144.dsl.cnl.uk.net) |
| 20:11.15 | dre | so setting it low will terminate calls faster, not send keepalives faster |
| 20:11.18 | Kobaz | Dovid: you can do sip debug with a specific ip and narrow it down if you have a lot of traffic |
| 20:11.38 | WIMPy | dre: yes |
| 20:12.48 | Dovid | Kobaz: all the traffic is from the same IP |
| 20:32.29 | *** join/#asterisk neurosys_ (~neurosys@216-106-180-226.ds1-static.mia1.net.ststelecom.com) |
| 20:34.01 | neurosys_ | Strange issue. Hope someone has seen this: voicemail() only plays the greeting and the jumps to next line. Anyone seen this? |
| 20:36.46 | Kobaz | Dovid: then you'll have to make do with that, you can parse the sip messages and track it by username or something like that |
| 20:38.45 | [TK]D-Fender | <PROTECTED> |
| 20:42.01 | neurosys_ | [TK]D-Fender: there's nothing in the debug or verbose. Its strange. |
| 20:42.14 | *** part/#asterisk tekoholic (~tekoholic@97-118-211-57.hlrn.qwest.net) |
| 20:42.33 | [TK]D-Fender | I doubt there is nothing... |
| 20:48.42 | neurosys_ | [TK]D-Fender: Ill set core verbose and debug on 9 |
| 20:50.21 | neurosys_ | [TK]D-Fender: http://pastebin.com/jbQdDCwm |
| 21:01.19 | neurosys_ | [TK]D-Fender: nm. i got it |
| 21:01.31 | neurosys_ | [TK]D-Fender: thx brutha |
| 21:04.51 | *** join/#asterisk drdru (~Adium@c-69-181-167-85.hsd1.ca.comcast.net) |
| 21:05.41 | *** join/#asterisk evil_gordita (robert@ip70-188-50-186.rn.hr.cox.net) |
| 21:10.59 | *** part/#asterisk beek (~klinebl@pdpc/supporter/bronze/beek) |
| 21:14.01 | *** part/#asterisk drdru (~Adium@c-69-181-167-85.hsd1.ca.comcast.net) |
| 21:25.29 | hardwire | hmmphm |
| 21:25.50 | hardwire | Anybody know what's so bleeding special about the h exten that anything that lasts more than a split second or spawns a thread causes severe problems? |
| 21:26.24 | hardwire | trying to issue a CURL request in a NoOp in h and it is teh fail. |
| 21:26.34 | hardwire | hello high cpu load |
| 21:32.40 | woleium | I worked out why my busy lamps weren't working on my polycoms - I needed <feature.presence feature.presence.enabled="1"></feature.presence> in the phone\s config file |
| 21:33.48 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 21:40.39 | *** join/#asterisk kaushal (~kaushal@115.118.146.94) |
| 21:50.25 | *** join/#asterisk irroot (~gregory@41.51.190.97) |
| 21:50.47 | *** part/#asterisk irroot (~gregory@41.51.190.97) |
| 21:58.54 | kaushal | Hi |
| 21:59.27 | *** part/#asterisk mjordan (~mjordan@nat/digium/x-vqqefxetgzvdywmy) |
| 21:59.46 | *** join/#asterisk vinhdizzo (~vinh@pool-173-60-112-55.lsanca.fios.verizon.net) |
| 22:06.52 | kaushal | As per http://www.voip-info.org/wiki/view/Asterisk+variable+DIALSTATUS, I am trying CANCEL: Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up. |
| 22:07.07 | kaushal | but i get "NO ANSWER" instead of "CANCEL" |
| 22:08.03 | kaushal | http://pastebin.ubuntu.com/786256/ |
| 22:08.07 | kaushal | Any clue ? |
| 22:10.09 | *** join/#asterisk s[X] (~mark@eth589.qld.adsl.internode.on.net) |
| 22:14.45 | *** join/#asterisk mantequilla_ (be0bbc86@gateway/web/freenode/ip.190.11.188.134) |
| 22:16.02 | kaushal | so as per http://www.asteriskguru.com/tutorials/cdr_custom_conf.html |
| 22:16.10 | kaushal | it is ${CDR(disposition)} = status of the call (ANSWERED, BUSY, NO ANSWER) |
| 22:16.30 | kaushal | so there is no CANCEL status ? |
| 22:18.00 | mantequilla_ | Hi, I've a trouble with Originate AMI Command, my manager.conf here http://pastebin.com/aa9JdPRZ : I can originate from telnet but not from AMI over HTTP, any idea ? |
| 22:18.19 | mantequilla_ | ah, asterisk 1.8 on Debian |
| 22:19.22 | kaushal | Please help me understand |
| 22:19.58 | kaushal | Are there any status condition apart from (ANSWERED, BUSY, NO ANSWER) ? |
| 22:23.46 | *** join/#asterisk infobot (~infobot@rikers.org) |
| 22:23.46 | *** topic/#asterisk is Asterisk: The Open Source PBX and Telephony Platform (asterisk.org) -=- Asterisk 10.0.0 (2011/12/15), 1.8.8.0 (2011/12/15), dahdi-linux 2.5.0.2 (2011/10/21), dahdi-tools 2.5.0.2 (2011/10/21), libpri 1.4.12 (2011/07/06) -=- Visit the official Asterisk wiki: wiki.asterisk.org |
| 22:23.47 | kaushal | Any one here ? |
| 22:27.47 | *** join/#asterisk neurosys_ (~neurosys@69.198.214.198) |
| 22:28.00 | pabelanger | kaushal: *CLI> core show application Dial |
| 22:28.21 | kaushal | pabelanger: please give me a moment |
| 22:28.32 | pabelanger | kaushal: *CLI> core show function CDR |
| 22:29.03 | kaushal | ok |
| 22:29.41 | kaushal | disposition - ANSWERED, NO ANSWER, BUSY, FAILED. |
| 22:30.14 | kaushal | pabelanger: so what would be the scenario in case of FAILED state ? |
| 22:30.23 | kaushal | How do i reproduce it ? |
| 22:31.22 | pabelanger | Congestion or some other error when dialing |
| 22:31.39 | *** join/#asterisk s[X]_ (~mark@eth589.qld.adsl.internode.on.net) |
| 22:31.39 | kaushal | ok |
| 22:31.51 | kaushal | pabelanger: so as per core show application Dial |
| 22:32.03 | kaushal | i see various state also |
| 22:32.47 | kaushal | CANCEL DONTCALL CONGESTION CHANUNAVAIL |
| 22:33.05 | kaushal | pabelanger: i have set it in dialplan |
| 22:33.38 | kaushal | am i missing something else ? |
| 22:34.58 | pabelanger | nope, sounds about right |
| 22:35.51 | kaushal | pabelanger: shall i pastebin my configs ? |
| 22:37.59 | pabelanger | IIRC: CANCEL is returned when you reach your Dial timeout |
| 22:38.31 | pabelanger | from the looks of it, you are not setting one |
| 22:38.55 | pabelanger | EG: Dial(DAHDI/g1/foo,30) |
| 22:39.10 | kaushal | ok |
| 22:39.23 | kaushal | so 30 means 30 secs ? |
| 22:39.27 | pabelanger | yes |
| 22:39.46 | kaushal | so by default i have only ANSWERED, NO ANSWER, BUSY, FAILED. ? |
| 22:39.56 | pabelanger | yes |
| 22:40.08 | pabelanger | it does not mean you cannot create your own |
| 22:40.08 | kaushal | pabelanger: ok |
| 22:40.55 | kaushal | pabelanger: not sure how do i set this state CANCEL DONTCALL CONGESTION CHANUNAVAIL also ? |
| 22:41.12 | pabelanger | checkout ${HANGUPCAUSE) |
| 22:41.20 | kaushal | ok |
| 22:41.22 | pabelanger | time for some food |
| 22:41.25 | pabelanger | & |
| 22:43.36 | *** join/#asterisk rhce7320 (~rhce7320@59.167.200.141) |
| 22:46.08 | kaushal | pabelanger: ok |
| 22:50.58 | kaushal | pabelanger: no luck :( |
| 22:52.47 | kaushal | http://pastebin.ubuntu.com/786298/ |
| 22:53.36 | *** join/#asterisk F2Knight (~ben@c-98-246-210-98.hsd1.or.comcast.net) |
| 22:54.23 | F2Knight | Q: Has anyone come across the accountcode not being populated when using realtime? specficly using sipfriends from a mysql DB for accounts. |
| 23:07.15 | twodogs | Odd issue. Apps that have been compatible with my device all along suddenly show as not in the market app... but they are, and market knows it at the web interface. Doesn't seem to matter what rom I'm using. |
| 23:07.45 | twodogs | But several games, in particular, can't be installed or updated through the app, only the web interface. |
| 23:07.53 | twodogs | msifire |
| 23:07.55 | twodogs | sorry |
| 23:10.16 | kaushal | pabelanger: Any clue ? |
| 23:16.38 | *** join/#asterisk EugeneKay (eugene@hosted.by.khresear.ch) |
| 23:20.04 | *** join/#asterisk Hanumaan (~Hanumaan@dslb-092-075-153-246.pools.arcor-ip.net) |
| 23:21.58 | *** join/#asterisk TimeRider (~steve@82.132.139.248) |
| 23:31.01 | kaushal | Also if i look at the console |
| 23:31.12 | kaushal | Any clue about [Dec 29 04:58:28] WARNING[12223]: app_dial.c:1353 wait_for_answer: Unable to write frametype: 2 ? |
| 23:31.21 | kaushal | I see this message at CLI > |
| 23:33.50 | *** join/#asterisk r33dtard (~r33dtard@gateway/tor-sasl/r33dtard) |
| 23:36.19 | kaushal | is there a way to see detailed description about ${HANGUPCAUSE) ? |
| 23:36.28 | kaushal | in CLI |
| 23:36.51 | F2Knight | kaushal, check your codecs |
| 23:37.20 | kaushal | F2Knight: are you referring to [Dec 29 04:58:28] WARNING[12223]: app_dial.c:1353 wait_for_answer: Unable to write frametype: 2 ? |
| 23:37.29 | F2Knight | yes |
| 23:37.36 | kaushal | ok |
| 23:37.44 | kaushal | F2Knight: any document ? |
| 23:37.54 | F2Knight | I suggest testing with both codes set to something like ulaw. |
| 23:38.32 | F2Knight | in sip.conf on the account your testing you can add disallow=all followed by allow=ulaw |
| 23:38.43 | F2Knight | then set your phone to use ulaw as its first choice |
| 23:39.11 | kaushal | F2Knight: i am using dahdi |
| 23:39.35 | F2Knight | umm... |
| 23:39.48 | F2Knight | US? |
| 23:39.54 | kaushal | nope |
| 23:40.02 | kaushal | India zone |
| 23:40.03 | F2Knight | do you use ulaw or alaw? |
| 23:40.08 | kaushal | alaw |
| 23:40.18 | F2Knight | okay set your phone and account to alaw to start. |
| 23:40.27 | kaushal | ok |
| 23:40.32 | F2Knight | thats the same codec your 'dahdi/zap' should be using. |
| 23:40.40 | kaushal | ok |
| 23:40.46 | kaushal | any config file ? |
| 23:40.55 | kaushal | under /etc/asterisk/ ? |
| 23:40.57 | F2Knight | the next thing is an assumption that the dahadi is actually configured correctly its self |
| 23:41.04 | F2Knight | nano /etc/astersisk/sip.conf |
| 23:41.09 | kaushal | ok |
| 23:41.26 | F2Knight | is your dahdi connected to a T1 or to a FXS/FXO card? |
| 23:41.42 | kaushal | E1 Sangoma PRI Card |
| 23:42.18 | kaushal | 06:04.0 Network controller: Sangoma Technologies Corp. A104d QUAD T1/E1 AFT card |
| 23:43.29 | F2Knight | okay you will also have to be assured that your E1 is acually configured correctly for your provider. |
| 23:44.16 | F2Knight | if your card is not configured correctly for yoru provider then you will obviously be having issues at that point as well. |
| 23:45.14 | F2Knight | ulaw/alaw is the 'native' codec for telephones... so setting your device to that means asterisk does no transcoding before sending it to the dahdi device. thus assuring us there is not transcoding issues. |
| 23:45.36 | F2Knight | asterisk more or less talks to dahadi as an external device so to speak. |
| 23:45.59 | kaushal | F2Knight: not sure where do i set it |
| 23:46.26 | kaushal | as you said you can add disallow=all followed by allow=ulaw |
| 23:46.57 | F2Knight | that is unlike sip/iax where asterisk configures the properties of the sip/iax channel. with dahadi asterisk more or less communicates to an external dahadi device that is configured out side of asterisk. |
| 23:47.08 | F2Knight | under your sip account you are testing with. |
| 23:47.16 | *** join/#asterisk Defraz (~Defraz@70.36.76.167) |
| 23:47.20 | F2Knight | are you using freepbx? |
| 23:47.25 | kaushal | Asterisk |
| 23:47.45 | F2Knight | did you set this box up? |
| 23:47.54 | kaushal | Asterisk 1.8.5.0 |
| 23:47.58 | kaushal | yes |
| 23:48.21 | F2Knight | you know how to set up sip? |
| 23:48.25 | *** join/#asterisk aidinb (~aidin@unaffiliated/aidinb) |
| 23:48.27 | kaushal | nope |
| 23:48.45 | F2Knight | then how did you ever get a phone to register |
| 23:49.05 | kaushal | F2Knight: just a sec |
| 23:50.33 | F2Knight | read thourgh your sip.conf.samples file usually located in the asterisk source directory, configs/sip.conf.sample |
| 23:50.39 | *** join/#asterisk EugeneKay (eugene@itvends.com) |
| 23:50.56 | kaushal | F2Knight: sure |
| 23:51.02 | kaushal | F2Knight: Thanks |
| 23:51.07 | F2Knight | it will give you some understanding of what to configure for sip. |
| 23:51.20 | carrar | Also read: |
| 23:51.22 | carrar | ~book |
| 23:51.23 | infobot | Asterisk: The Definitive Guide, 3rd Edition (ISBN 0-596-51734-3) available at http://oreilly.com/catalog/9780596517342 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 23:51.59 | kaushal | carrar: Thanks |
