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| 00:13.19 | PHunter | So besides a linux box WIMPy, is there a 'off the shelf' product that will work ok? |
| 00:14.16 | WIMPy | Most of them should work. |
| 00:14.37 | PHunter | I figured as much. Any you know with configurable NAT Timeout? |
| 00:14.37 | WIMPy | And, off course, most of them are just linux boxes. |
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| 00:14.58 | PHunter | Right, the Cisco's love to taunt me thought for whatever reason. |
| 00:15.00 | file | is a fan of the Ubiquiti Edgemax router |
| 00:15.07 | PHunter | looks.. |
| 00:15.23 | WIMPy | If you can get a shell, you can configure timeouts. |
| 00:15.45 | WIMPy | But maybe you should just enable keep-alives on the phones. |
| 00:15.57 | PHunter | Already doing that. |
| 00:16.03 | PHunter | Thats where Im lost. |
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| 01:13.05 | ChannelZ | mjordan: (from awhile ago) no doing something else entrely.. ffmpeg ish |
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| 02:14.07 | qakhan | i want to create an ext which can configure through alphabatic username (ext01) but it can be dialed as number (1001) |
| 02:17.14 | WIMPy | Sounds like your understanding of "extension" is not Asterisk compatible. |
| 02:17.26 | WIMPy | An extension is what you dial. |
| 02:19.14 | qakhan | ??? |
| 02:22.49 | qakhan | WIMPy i talking about ext registeration |
| 02:23.42 | qakhan | i want to register an ext using alpabatic username and pass but dial through number |
| 02:24.03 | qakhan | assigned to that ext |
| 02:24.29 | WIMPy | Extensions are "registered" by configuring them in extensions.conf. |
| 02:25.07 | qakhan | what about sip.conf |
| 02:25.17 | WIMPy | And that file is the only relation between extension and what they do, like e.g. dial some device. |
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| 02:29.53 | qakhan | what do we call those account which we create in sip.conf |
| 02:30.21 | WIMPy | "Account" is an appropriate term. So is "device" or "peer". |
| 02:34.52 | qakhan | ok so is there any way to register a device or peer using alphabatic username |
| 02:35.24 | WIMPy | Sure. It can be whatever you want. |
| 02:35.50 | qakhan | but assign a number to it |
| 02:36.12 | WIMPy | Purely numeric usernames are easy to guess. so they are a bad idea anyway. |
| 02:37.00 | qakhan | that is why i want to create alphabatic username and assign a number to it |
| 02:37.26 | WIMPy | That's the wrong way round. |
| 02:37.40 | qakhan | ok |
| 02:37.47 | qakhan | so how can we do it |
| 02:37.55 | WIMPy | You define what has to happen when an extension is called, not how a device is called. |
| 02:38.54 | qakhan | did not understand |
| 02:40.23 | WIMPy | extensions.conf. |
| 02:42.05 | qakhan | so i shold work in extesions.conf instead of sip.com |
| 02:42.52 | WIMPy | You define your peers in sip.conf (and/or possibly others) and your extensions in extensions.conf. |
| 02:44.27 | qakhan | extensions mean username and pass? |
| 02:45.26 | WIMPy | No. Scroll up. |
| 02:46.04 | WIMPy | Or read the |
| 02:46.09 | WIMPy | ~book |
| 02:46.09 | infobot | Asterisk: The Definitive Guide, 4th Edition (ISBN 1-4493-3242-0) available at http://oreilly.com/catalog/0636920025894 - Asterisk: The Definitive Guide is released under a Creative Commons License (http://creativecommons.org/licenses/by-nc-nd/3.0/us/) and a version is available for reading online at http://www.asteriskdocs.org/ or see ~buybook |
| 02:47.48 | qakhan | ok |
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| 05:16.51 | zopsi | mjordan: We've been having issues with power lately and the d70s will go to the connection timed out screen where you can hit retry and it just sits there until someone hits retry. Is there any way to set a retry interval or increase the attempts to like 100? It really impedes the reliability of our system as they will often sit there and we won't know until we physically use or look at the phone (not in plain sight in all cases). |
| 05:20.36 | kafal1 | mjordan: is there any documentation that shows how to use database to store agents.conf setting in v12.x. In this version agents.conf schema has changed. Rather than specifying agents on a single line in comma delineated fashion, each agent is defined in a separate context. |
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| 05:24.37 | zopsi | sorry mjordan. I'll try and troubleshoot it myself. I have registration_max_retries=1000. I'll play around with the values. |
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| 06:03.00 | jameswf | http://lmgtfy.com/?q=asterisk+12+agents.conf |
| 06:04.56 | kafal1 | hi jameswf, i have googled a lot but i am not able to find any rsource which shows how to make static realtime work for agents.conf in 12.x |
| 06:05.36 | jameswf | kafal1: it is like the 3rd or 4th result |
| 06:05.39 | kafal1 | I have it working for asterisk 1.4 where the agent info is stored in ast_config in var_val column |
| 06:06.44 | kafal1 | Maybe I am missing something but those links just show how to use agents.conf to store the info |
| 06:06.56 | kafal1 | I wanted to use static realtime and store the info in db |
| 06:07.48 | kafal1 | earlier we could store the agent login detials in var_val column in ast_config table. |
| 06:08.19 | kafal1 | just wanted some guidance on how i can make static realtime work for agents.conf |
| 06:33.49 | kafal1 | if anyone knows how to do it, please provide some guidance. |
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| 12:30.08 | adium | Hello there |
| 12:30.25 | adium | I'm trying to use a DDoS defense on the SIP protocol |
| 12:31.28 | adium | But when I try to change the port to monitor the network the asterisk server doesn't allow the users to make a call |
| 12:31.30 | adium | Any help? |
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| 12:50.35 | adium | Hello? |
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| 14:34.34 | file | hi |
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| 15:45.42 | NirS | Good Morning/Afternoon all |
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| 21:08.11 | metakungfu | hey guys, is there a way to store records to s3 through punchblock ? |
| 21:08.28 | metakungfu | or directly asterisk? |
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| 21:11.16 | [TK]D-Fender | records of what? |
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| 21:14.51 | metakungfu | i'm using adhearsion, trying to have voice recording during a call |
| 21:14.52 | metakungfu | http://www.adhearsion.com/docs/call-controllers#voicemail-like-recording |
| 21:15.17 | metakungfu | i'm trying to get those recordings stored on s3 |
| 21:15.56 | metakungfu | not sure there is a way to do that natively with asterisk, or if anyone is familiar with adhearsion, if there is a way to do that... |
| 21:16.45 | metakungfu | my quickfix is to mount a s3 bucket and set adhearsion to tell asterisk to store the recordings in that folder |
| 21:16.54 | metakungfu | not sure if there is a better way |
| 21:19.46 | [TK]D-Fender | Don't really have 3rd party GUI support fort stuff like that here. |
| 21:19.59 | [TK]D-Fender | If you were doing doing this yourself it'd be easy |
| 21:20.19 | metakungfu | what do you mean do it myself? |
| 21:20.30 | metakungfu | you mean if i was doing that right from asterisk? |
| 21:22.10 | [TK]D-Fender | yes |
| 21:22.36 | metakungfu | sweet, could you help me out to store the recording on s3 from asterisk? |
| 21:24.16 | [TK]D-Fender | "core show application monitor" <- |
| 21:24.28 | [TK]D-Fender | you can have it run a script when it's finished |
| 21:25.07 | [TK]D-Fender | Set it up to send it over via FTP, SCP, ewhatever |
| 21:25.23 | metakungfu | awesome! i'll check that out |
| 21:25.32 | metakungfu | thx! |
| 21:27.51 | _0x5eb_ | hi, is it possible to have a SIP section only containing "fromdomain=some.domain"? |
| 21:29.44 | [TK]D-Fender | yes, though it won't work |
| 21:29.57 | [TK]D-Fender | because without a type * won't know to use it for anything |
| 21:31.25 | _0x5eb_ | sure, then would "type=peer" without specifying any host make more sense? |
| 21:32.07 | _0x5eb_ | (the idea is to force to use a specific from domain when connecting to fully qualified SIP URIs) |
| 21:34.05 | *** join/#asterisk BakaKuna (~Thunderbi@82-169-251-128.ip.telfort.nl) |
| 21:36.44 | *** join/#asterisk reconwireless (uid10170@gateway/web/irccloud.com/x-umitbpsdkaqyynlb) |
| 21:38.45 | [TK]D-Fender | And for the rest of the host settings? |
| 21:42.49 | _0x5eb_ | well I would like to use it for E.164 resolution, so precisely no specific host |
| 21:43.12 | _0x5eb_ | I would just like to present a specific host domain when performing E.164-resolved SIP calls |
| 21:43.32 | _0x5eb_ | s/host/from/ |
| 21:43.34 | [TK]D-Fender | Doesn't work that way. |
| 21:44.08 | [TK]D-Fender | You can't just pump a host in after that fact. That is a requires setting of a SIP entry |
| 21:44.22 | [TK]D-Fender | required* |
| 21:45.18 | *** join/#asterisk aboven (~aboven@204.14.77.11) |
| 21:46.27 | _0x5eb_ | so I could not use a SIP section other than [general] for setting the fromdomain for E.164 resolution? |
| 21:46.53 | [TK]D-Fender | vorrect. |
| 21:47.20 | [TK]D-Fender | you can't shove a URI to a peer. The peer is used to construct it. |
| 21:47.35 | [TK]D-Fender | Any raw dialed URI is getting [general] settings only |
| 21:48.36 | _0x5eb_ | and could I customize the @domain from the from using anything else than fromdomain=, ideally in the dialplan? |
| 21:50.11 | [TK]D-Fender | <PROTECTED> |
| 21:50.47 | _0x5eb_ | OK, thanks I will just stick with that then |
| 21:50.48 | [TK]D-Fender | You seem to be trying to insist on using chan_sip for things it was not designed for. This pretty much only ends in pain. |
| 21:51.06 | [TK]D-Fender | Asterisk is not a "SIP server" |
| 21:52.02 | [TK]D-Fender | chan_pjsip may have better odds. You've been advised to look at that route several times. Are you planning on getting off of the aged 1.8 you were on toward something current? |
| 21:52.08 | _0x5eb_ | yes, I'm really considering whether I should switch to asterisk 12 or Kamailio as a companion to Asterisk which would be then only used as IPBX |
| 21:53.09 | _0x5eb_ | is chan_pjsip a improved version of chan_sip or a SIP server with functionalities comparable to those of Kamailio? |
| 21:53.29 | [TK]D-Fender | it is a completely different stack from top to bottom based on pjproject |
| 21:54.00 | [TK]D-Fender | And no, it cannot be comparaable as kamailio is a full SIP router, registrar, etc |
| 21:54.37 | _0x5eb_ | so it would provide some (maybe a lot of) more functionalities, but probably not as much as using a SIP server in front of Asterisk? |
| 21:54.38 | [TK]D-Fender | Asterisk is not and will pretty much never be a full SIP server by any reasonable estimate. That was never its purpose or design. |
| 21:55.52 | [TK]D-Fender | pjsip may get you what you need though so don't waste time trying to beat chan_sip into compliance. This is not the future |
| 21:56.43 | _0x5eb_ | that definitively makes sense and that's one of the point that makes me stick (for now) to Asterisk 1.8 |
| 21:57.11 | _0x5eb_ | I'm afraid that Asterisk 12 is trying to incorporate (too) many functions and loosing its "simplicity" and efficiency at those tasks |
| 21:57.30 | WIMPy | It's what? |
| 21:57.45 | _0x5eb_ | I put the quotes ;) |
| 21:59.00 | _0x5eb_ | I would much prefer having a SIP server (doing only SIP, no media stream, nothing else) next to a IPBX than having some hybrid software trying to do both |
| 21:59.40 | WIMPy | Do it |
| 21:59.42 | [TK]D-Fender | 12 isn't seriously differnt for the basics |
| 21:59.56 | [TK]D-Fender | the SIP channel driver is the only thing that'll be rather new |
| 22:00.23 | WIMPy | There isn't ,uch difference if you don't use chan_pjsip :-) |
| 22:01.00 | _0x5eb_ | OK so if it's only the improvement of an obsolete channel (optionally) replaced by one with greater functionalites that sounds pretty safe |
| 22:01.32 | WIMPy | You can still use chan_sip. |
| 22:01.35 | [TK]D-Fender | Far from the only improvement. |
| 22:01.53 | [TK]D-Fender | but if you're doing general stuff.. you'll just have more |
| 22:02.27 | _0x5eb_ | just a very naive question: why is Debian still sticking to Asterisk 1.8? |
| 22:02.53 | WIMPy | Why does Debian never use anything less than 3 years old? |
| 22:04.14 | _0x5eb_ | I will try to compile some tests Asterisk 12 Debian 7 packages and test them |
| 22:05.57 | _0x5eb_ | (for Debian: I'm just surprised that given that Debian 7 was released in May 2013 they still choosed Asterisk 1.8) |
| 22:21.31 | *** join/#asterisk yokel (~yokel@unaffiliated/contempt) |
| 22:31.09 | *** join/#asterisk fstd_ (~fstd@unaffiliated/fisted) |
| 22:51.47 | mjordan | _0x53b_: I'm not sure I understand your statement about Asterisk 12 incorporating too many functions. |
| 22:55.04 | TazzNZ | _0x5eb_: Ubuntu followed a similair path - only with 14.04 did they switch to Ast11 |
| 23:06.23 | *** join/#asterisk MarcoZink (~marcozink@187-176-63-48.dynamic.axtel.net) |
| 23:12.02 | _0x5eb_ | mjordan: I was just wondering whether newer versions of asterisk were trying to incorporate functions out of its initial scope |
| 23:13.34 | _0x5eb_ | it seems that Asterisk 12 just offers the ability to use an updated version of a module, which is the right thing to do |
| 23:15.30 | _0x5eb_ | btw, is chan_jingle updated or removed in Asterisk 12? |
| 23:17.16 | [TK]D-Fender | replace long before with chan_motif |
| 23:17.37 | [TK]D-Fender | You're 3 entire branches behind |
| 23:22.50 | *** join/#asterisk jpoz (~jpoz@ec2-54-193-11-223.us-west-1.compute.amazonaws.com) |
| 23:24.27 | _0x5eb_ | OK I will experiment with Asterisk 12 starting tomorrow |
| 23:24.41 | *** join/#asterisk jpoz_ (~jpoz@c-24-22-98-12.hsd1.or.comcast.net) |
| 23:25.43 | _0x5eb_ | by reading http://blogs.digium.com/2013/11/20/asterisk-12-part-iv-sip-stack-future/ I am wondering if chan_pjsip is a stable replacement of chan_sip or is it a yet experimental module? |
| 23:30.38 | [TK]D-Fender | it works. |
| 23:30.52 | [TK]D-Fender | And it seems to be the way going forward. |
| 23:36.36 | *** join/#asterisk raspberrypifan (~textual@190.130.236.28) |
| 23:36.41 | _0x5eb_ | OK, thanks again for all that info, I will dig into Asterisk 12 docs to find out more about chan_motif and chan_pjsip and experiment with that |
| 23:38.46 | file | chan_sip for life! |
| 23:44.10 | mjordan | I'm not sure what "initial scope" means for Asterisk |
| 23:44.15 | mjordan | :-) |
| 23:44.53 | mjordan | looking at version 1.0.0, I'd say the scope of Asterisk has changed. |
| 23:46.23 | *** join/#asterisk petris (~petris@2607:5300:60:5475:a9d7:583:99b:a901) |
| 23:55.21 | *** join/#asterisk theron (~theron@173-29-74-236.client.mchsi.com) |