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| 00:40.03 | gregs | file: Do you have an ETA for 13.9.1? |
| 00:40.11 | file | soon? |
| 00:46.22 | Samot | Is 13.9.1 urgently needed? |
| 00:46.31 | Samot | I just got 13.9.0 today. |
| 01:03.34 | gregs | Thanks. |
| 01:03.53 | gregs | Samot: a |
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| 01:28.37 | gregs | Samot: A nasty PJSIP bug was introduced in 13.9.0. It was fixed today. |
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| 04:05.08 | Lope | what is a typical dial format for SIP? I know how to call softphones. Dial(SIP/foo) can you please give some other examples? |
| 04:06.37 | drmessano | core show application dial |
| 04:07.20 | drmessano | https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial |
| 04:07.34 | Lope | I did, it says Technology/Resource |
| 04:07.43 | drmessano | Yep |
| 04:07.49 | Lope | where <Technology> represents a particular |
| 04:07.49 | Lope | <PROTECTED> |
| 04:07.49 | Lope | <PROTECTED> |
| 04:07.57 | drmessano | Correct |
| 04:08.04 | Lope | But I want more info about the SIP channel driver's resource format? |
| 04:08.22 | drmessano | Its whatever you need to dial |
| 04:08.52 | Lope | so 888@foobar.com ? |
| 04:09.02 | drmessano | Could be |
| 04:09.18 | Lope | Where can I see what the possible formats are? |
| 04:10.10 | Lope | There is no @ sign in this doc: https://wiki.asterisk.org/wiki/display/AST/Asterisk+13+Application_Dial |
| 04:13.09 | Lope | for example SIP/extension[@host] |
| 04:23.14 | ChannelZ | Generally in the sample config for the particular channel driver |
| 04:23.28 | ChannelZ | chan_sip.conf in the SIP/... case |
| 04:24.12 | ChannelZ | err sorry.. sip.conf |
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| 05:23.07 | UbuntuDude | where can I read simple classification of the hardware cards to be installed within asterisk pbx with abbreviation of their functionality differences? |
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| 08:13.22 | Lope | ChannelZ: okay, thanks, will check it out. |
| 08:15.32 | Lope | What is supposed to be inbetween square brackets in sip.conf? From what I gather it looks like the username? One curious config is callcentric, who put the string "callcentric" inbetween the square brackets, which doesn't match anything. Surely it should rather be [1777MYCCID] ? http://www.callcentric.com/support/device/asterisk |
| 08:19.17 | Lope | ChannelZ thanks, busy reading thru the sample sip.conf. very enlightening :) |
| 08:20.15 | ChannelZ | Well [zzz] is a peer named [zzz] |
| 08:20.23 | Lope | so ot |
| 08:20.23 | ChannelZ | err named zzz |
| 08:20.32 | Lope | So it's just the device name inside the square brackets? |
| 08:20.46 | Lope | because I'm wondering how asterisk matches the register line with the context? |
| 08:20.47 | ChannelZ | Yes. Which is sort of be a username |
| 08:20.53 | Lope | Or if it needs to be matched at all? |
| 08:20.58 | ChannelZ | It doesn't |
| 08:21.05 | Lope | Doesn't need to be matched? |
| 08:21.22 | ChannelZ | No they are totally unrelated. Registration just informs a remote end what your IP is |
| 08:21.31 | Lope | I see. |
| 08:21.36 | Lope | Thanks!!! |
| 08:21.39 | ChannelZ | You still have to have a properly configured peer for an incoming call to match |
| 08:21.50 | Lope | yeah |
| 08:23.10 | Lope | The chan_sip driver offers a few different ways to dial, some specifically using a devicename. SIP/devicename SIP/devicename/extension SIP/devicename/extension/IPorHost |
| 08:23.20 | Lope | Are devicenames devices described in sip.conf? |
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| 08:24.17 | utrack | Hi! There's F option for the Dial() - however as the execution is started @ next priority - all the variables are lost. Is there any way to continue the flow without losing the vars? |
| 08:24.22 | Lope | So for example I can Dial() anything described in sip.conf by Dial()ing a devicename. But if I want to dial something NOT described in sip.conf, I'd use one of the other syntaxes that involve domains/hosts/IPs? |
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| 08:48.50 | utrack | Or is there a way to execute the h extension only for one channel of two? (exp. during Originate()d calls) |
| 08:51.02 | ChannelZ | lope, yes |
| 08:51.35 | ChannelZ | utrack, I think you'd need to use a hangup handler for that.. |
| 08:53.05 | ChannelZ | https://wiki.asterisk.org/wiki/display/AST/Hangup+Handlers |
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| 11:02.30 | utrack | ChannelZ: yep, thanks! Works :) |
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| 14:58.53 | dan_j | Hi. I've got some stuck channels where the actual call ended over 24 hours ago. I've tried ending the calls using channel request hangup but they are not ending. |
| 14:59.11 | dan_j | In the meantime, the CLI keeps repeating this |
| 14:59.36 | dan_j | chan_sip.c:4405 __sip_autodestruct: Autodestruct on dialog '2b7bf88f79c80acd3d5f43e66435da73@45.37.180.177:5060' with owner SIP/endpoint-0005010f in place (Method: BYE). Rescheduling destruction for 10000 ms |
| 14:59.51 | dan_j | Any ideas what might be causing this? |
| 15:03.47 | dan_j | When I do 'channel request hangup', the CLI says Requested Hangup on channel 'CHANNEL NAME', but nothing else happens. |
| 15:03.54 | dan_j | Normally that does the job. |
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| 15:44.54 | darkdrgn2k | hi all, |
| 15:45.18 | darkdrgn2k | is tehre a way to have ASTERISK not touch DTFM tones on a call (dont try to read them ,re produce them etc just leave them inband as they come in?) |
| 15:46.00 | [TK]D-Fender | Set it to a mode that is not used by the call |
| 15:48.59 | darkdrgn2k | im not quite clear what you mean by mode |
| 15:49.29 | [TK]D-Fender | DTMT mode. |
| 15:49.40 | [TK]D-Fender | like the name of the very clear setting for this |
| 15:50.32 | darkdrgn2k | so set one side to INBAND and the other side to INFO? |
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| 15:51.08 | [TK]D-Fender | If you don't want * to touch it then make sure that each side does NOT match |
| 15:52.22 | darkdrgn2k | hmm didnt think of that one.. i should try it :) |
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| 16:32.51 | darkdrgn2k | ok really stupid question, when they say "domain" on the SPA its just @blablabla right? |
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| 18:17.38 | jennie | hi gusy, any idea whats going on? http://paste.ubuntu.com/16394886/ |
| 18:20.26 | [TK]D-Fender | Told you already ... manager.conf |
| 18:20.28 | [TK]D-Fender | LOOK at it |
| 18:20.33 | [TK]D-Fender | and keep this in #freepbx. |
| 18:20.36 | [TK]D-Fender | that GUI owns you now |
| 18:20.45 | [TK]D-Fender | and you have to do things its way |
| 18:24.34 | jennie | okay, I thought I got it and its asterisk problem now. I will looki |
| 18:26.09 | [TK]D-Fender | FreePBX owns your configs. You have given up the steering wheel to it and ahve to play by its rules. |
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| 18:41.07 | saint_ | hi all - Does anyone know by anychance what is the RFC defining what are the authorized characters for a User-Agent header in a sip registration ? |
| 18:48.48 | syadnom | anyone know of a build of Siren7/14 for Asterisk13? |
| 19:28.16 | newtonr | syadnom, I'm going to create a JIRA issue for it so we can make a build when possible. I don't think Digium has one at the moment. |
| 19:28.34 | syadnom | newtonr, nice |
| 19:28.48 | syadnom | newtonr, I'm probably going to drop to asterisk11 to run the existing codec builds for now. |
| 19:28.56 | syadnom | just don't like going backwards :? |
| 19:31.23 | *** part/#asterisk jennie (~chatzilla@unaffiliated/jennie) |
| 19:32.32 | newtonr | Sorry! This is the first time I've seen anyone ask about Siren codecs in the past few years so I imagine due to low demand it probably slipped the mind of whoever normally does it. |
| 19:33.01 | syadnom | newtonr, no surprise there. polycom specific... |
| 19:36.19 | newtonr | syadnom, do you have a JIRA account? I can change the reporter to you once I create the issue. You can PM me |
| 19:38.45 | syadnom | newtonr, I don't have a paid account. |
| 19:39.35 | syadnom | now sure if I need a 'proper' account or not. |
| 19:39.39 | newtonr | Atlassian JIRA is what we use for bug tracking Asterisk issues. There is no paid account |
| 19:39.53 | newtonr | https://signup.asterisk.org |
| 19:40.01 | newtonr | then login at issues.asterisk.org/jira |
| 19:40.44 | syadnom | im in |
| 19:42.28 | newtonr | Cool, got you on there. |
| 19:47.31 | syadnom | newtonr, awesome, thanks |
| 19:47.55 | syadnom | is there an opus build for asterisk13? or does it still need a patch to asterisk? |
| 19:50.39 | newtonr | Opus is currently only supported for passthrough only |
| 19:51.00 | syadnom | ok, that's what google tells me. or patch asterisk... |
| 19:51.34 | syadnom | do you know if it is a planned feature for a future release? |
| 19:53.28 | newtonr | I don't know of anyone working on it |
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| 19:59.01 | WIMPy | syadnom: Not happening due to legal issues. |
| 19:59.06 | WIMPy | http://lists.digium.com/pipermail/asterisk-dev/2013-May/060419.html |
| 19:59.46 | WIMPy | Siren would probably need a licensed module like G.729 as well. |
| 20:00.45 | syadnom | WIMPy, Siren is royalty free |
| 20:01.21 | syadnom | WIMPy, and you can get a license easily via form on polycom's website |
| 20:01.22 | WIMPy | News to me. |
| 20:01.56 | syadnom | https://en.wikipedia.org/wiki/Siren_(codec) |
| 20:01.59 | syadnom | see Licensing |
| 20:02.26 | syadnom | basically, if you run polycom endpoints you automatically meet all the requirements for the royalty free license. |
| 20:03.11 | syadnom | WIMPy, I think I have read that post, or one like it before, and just forgot. |
| 20:03.25 | WIMPy | Sounds like it at least couldn't be shipped with Asterisk. |
| 20:04.21 | syadnom | yeah, understandable |
| 20:04.49 | WIMPy | And all links to poycom are broken :-( |
| 20:05.09 | syadnom | yeah, polycom just redid a bunch on their website. |
| 20:06.01 | syadnom | http://www.polycom.com/company/about-us/technology/siren/siren-licensing-form.html |
| 20:06.06 | WIMPy | Nothing on their site map, either. |
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| 20:12.05 | syadnom | Polycom is actually supporting opus in current firmwares on their VVX500+ phones. |
| 20:12.25 | syadnom | wonder how they are confident that patents wont bite them... |
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