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| 09:50.33 | AL13N | i have an asterisk at home, and i have a sip account somewhere... i would like to connect my asterisk to the sip account, instead of having 2 sip accounts on my client |
| 09:50.50 | AL13N | how can i connect my asterisk to the remote asterisk with a sip registration? |
| 09:51.36 | AL13N | huh, wait, i can just do as if it's a remote trunk |
| 10:13.56 | pchero_travel | AL13N: Do you have remote trunk? And do you want to register it Asterisk to Trunk provider? |
| 10:15.06 | pchero_travel | May this link can help you. http://www.voip-info.org/wiki/view/Asterisk+SIP+register |
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| 10:32.15 | AL13N | thx! i'll go read |
| 10:39.13 | pchero_travel | AL13N: Once you done config, you can check it using "sip show registry". :) |
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| 12:12.23 | ldc | hello! since I've changed my router I have one single phone that doesn't work, all the others work fine |
| 12:12.29 | ldc | that single phone seems to register then it becomes unreachable |
| 12:12.45 | ldc | it's the only phone different from the others (grandstream vs cisco) |
| 12:12.53 | ldc | any hints? |
| 12:13.09 | ldc | phones are behind nat with pbx outside |
| 12:24.09 | pchero_travel | NAT? |
| 12:24.47 | pchero_travel | Have you checked STUN config? |
| 12:26.44 | ldc | hmm I've never used STUN on the other phones |
| 12:33.03 | pchero_travel | ldc: Using SIP protocol with NAT, it needs STUN config. |
| 12:38.06 | ldc | pchero_travel: adding a stun server seems to have fixed it, thanks :) |
| 12:38.16 | ldc | odd, since all the other phones work fine without one |
| 12:39.17 | pchero_travel | Well, magic? It supposed working with STUN, if it worked inside of NAT.. |
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| 15:31.03 | volga629 | Hello Everyone, having issue with where asterisk not save record route from proxy when received SUBSCRIBE and send NOTIFY |
| 15:31.05 | volga629 | https://issues.asterisk.org/jira/browse/ASTERISK-13803 |
| 15:32.14 | volga629 | based on loose route it save and send NOTIFY with record routes based from SUBSCRIBE |
| 15:42.21 | AL13N | does asterisk have starttls support on SIP? |
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| 19:36.05 | roswell | hello everyone. is there a way of setting a custom sip header in a peer definition, like with setvar? |
| 19:38.35 | [TK]D-Fender | You should be able to set the function using that. |
| 19:42.49 | roswell | thought so, but all i've found related to this, was SIP_HEADER , which seems not to let setting, only retrieving. just decided to ask |
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| 19:48.10 | ChannelZ | are you talking about SipAddHeader() ? |
| 19:53.04 | [TK]D-Fender | Yup, looks like its only the application that adds them. |
| 19:55.58 | roswell | right, i figured that prior to asking here, just wanted to know whether someone had some workaround for a peer definition |
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| 21:40.56 | marlinc | What's the highest quality codec supported by Asterisk for media files |
| 21:41.14 | marlinc | Music on hold, sounds, etc.. |
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| 21:42.09 | WIMPy | signed linear 192Ks/s |
| 21:42.13 | roswell | adpcm perharps |
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| 21:49.16 | marlinc | Okay.. lets find out more about those |
| 21:49.41 | roswell | WIMPy was more correct however. slin192 |
| 21:50.53 | marlinc | So if I wanted to for example use 'slin192', would it be possible to convert files using for example avconv or what would be the usual way of doing so |
| 21:52.07 | WIMPy | I doubt you will find any sources for that quality. |
| 21:52.52 | WIMPy | Or any phone that can make use of it. As far as phones go, G.722 is usually as far as it gets. |
| 21:54.29 | marlinc | Okay I've got no idea what to use and I don't have any knowledge about any of the codecs so I'm just going with what you're saying for now |
| 21:54.38 | marlinc | Now I just have to find out how to create G.722 sound files right? |
| 21:55.45 | pchero | Hm... |
| 21:55.45 | WIMPy | You don't have to. Just use slin16 (or even more). Asterisk will transcode on demand. |
| 21:56.14 | WIMPy | If you want to save resources, you should have the files in the format that is most likely used by your phones. |
| 21:56.30 | marlinc | Okay that's nice, so what format would you suggest (that has the highest quality) for music on hold files |
| 21:56.34 | marlinc | Resources are no problem |
| 21:57.28 | WIMPy | Usually you wil have your music in 44.1 or 48 Ks/s, so just keep it that way. But convert them to mono. |
| 21:57.47 | panamark | Sorry to cut in but am not a fan of transcoding. So i would sugget to convert your sound files to your prefered codec. You can read more at: http://wiki.innovaphone.com/index.php?title=Howto:Convert_wave_files_in_to_G722_coder_files |
| 21:58.47 | marlinc | What's your reasoning for disliking transcoding, because it has to transcode on the fly? |
| 21:59.19 | WIMPy | It's repeating the work for each single call instead of once for all. |
| 21:59.25 | panamark | Yes, and resources might not be a problem as we speak but it can be in the near future |
| 21:59.38 | panamark | +1 for WIMPy |
| 21:59.49 | panamark | its a repeated process you are wasting resources |
| 22:00.38 | WIMPy | But the advantage is that you don't have to know what CODECs will be used on the calls and always get optimum quality. |
| 22:01.04 | panamark | moreover g722 is low complexity codec |
| 22:01.09 | WIMPy | But you can also save your files in multiple formats. |
| 22:01.59 | marlinc | Is there a way to see what codec is being used on a call? In older Asterisk version it was possible to see using 'sip show channels' |
| 22:02.13 | marlinc | But now I'm using pjsip and the same command for pjsip doesn't show the codec |
| 22:03.09 | panamark | you dont have access to asterisk cli? |
| 22:03.15 | marlinc | I do |
| 22:03.39 | marlinc | I do have acces |
| 22:06.25 | ChannelZ | The channel knows. 'core show channels concise' and then 'core show channel XXXX' |
| 22:07.32 | marlinc | WriteTranscode: Yes (slin@8000)->(ulaw@8000) |
| 22:07.42 | marlinc | Those are the two codecs |
| 22:11.35 | panamark | i thought 'core show channels concise' was deprecated |
| 22:13.00 | ChannelZ | It is but stupidly 'verbose' cuts off the channel name in most instances too, so I'm not sure what you're supposed to do once 'concise' goes away |
| 22:13.28 | WIMPy | Use the Tab key. |
| 22:13.30 | ChannelZ | If you have verbose console on you'll see the channel name a bunch of times as the call goes through the dialplan, but still.. |
| 22:13.46 | marlinc | I just tabbed and it autocompleted |
| 22:14.23 | ChannelZ | True. |
| 22:14.59 | panamark | concise is overated. when it goes away we will sniff packets :D |
| 22:15.01 | ChannelZ | It's kind of a pain for 'trunks' like incoming/outgoing ITSP if you have a lot of calls happening simultaneously, to find the one you actually want, but.. |
| 22:16.07 | panamark | Does anyone use grandstream phones? |
| 22:16.12 | panamark | *uses |
| 22:16.16 | ChannelZ | I don't know why the columns are so artificially small. Should probably be re-worked to try and find the longest channel name first and then create the columns accordingly, but oh well. There's other things to worry about :) |
| 22:17.16 | WIMPy | Oh yes |
| 22:28.09 | pchero | Does anybody know why "module reload chan_sip.so" doesn't work? |
| 22:28.15 | pchero | in cli? |
| 22:28.36 | pchero | Actually, it's working, but it returns error. |
| 22:28.58 | pchero | No such module 'chan_sip.so' |
| 22:29.18 | ChannelZ | then it probably wasn't working in the first place |
| 22:29.42 | pchero | ah.. |
| 22:29.45 | ChannelZ | (or maybe it's statically built? Rare but not impossible) |
| 22:30.06 | pchero | I just looked chan_sip.c file |
| 22:30.24 | pchero | But, I couldn't find reload_module() |
| 22:31.02 | pchero | Not implemented? or wrong usage? Hm.. |
| 22:31.40 | pchero | but "sip reload" working fine. |
| 22:32.16 | pchero | but problem is when do the "sip reload", it doesn't send Event: Reload AMI.. |
| 22:32.17 | ChannelZ | What does 'module show' say? |
| 22:32.48 | pchero | module show like chan_sip.so |
| 22:32.48 | pchero | <PROTECTED> |
| 22:33.07 | ChannelZ | hmm.. well module reload works here. What version of *? |
| 22:33.21 | pchero | I just downloaded from svn |
| 22:33.30 | pchero | to test for this. |
| 22:33.43 | pchero | svn checkout http://svn.asterisk.org/svn/asterisk/trunk asterisk |
| 22:34.00 | pchero | Asterisk SVN--r432405 built |
| 22:35.11 | pchero | technically, working fine. But the message is problem. It says no such module. Even AMI message code set to "2" which is "no such module".. |
| 22:37.30 | marlinc | If I get the following |
| 22:37.30 | marlinc | No joint capabilities for 'audio' media stream between our configuration((g722)) and incoming SDP((ulaw|gsm|alaw|g729|speex|g726|g726aal2)) |
| 22:37.45 | marlinc | Are the codecs listed in the last part the ones supported by my provider? |
| 22:37.55 | [TK]D-Fender | us VS them |
| 22:38.07 | WIMPy | Whatever you're talking to. |
| 22:38.08 | [TK]D-Fender | they don't allow g.722 |
| 22:38.18 | [TK]D-Fender | And that's all you allow |
| 22:38.24 | ChannelZ | Huh. Never seen a provider do speex |
| 22:38.27 | marlinc | They do advertise with that's what they have |
| 22:38.39 | marlinc | They so say they allow g722 |
| 22:38.53 | [TK]D-Fender | Perhaps you need your account set up especially for it |
| 22:39.06 | [TK]D-Fender | Or it could allow that only between account on the same service, etc |
| 22:41.12 | marlinc | Okay so which one of the ones they provide would be the highest quality? Because g722 appears to be the 'HD' codec |
| 22:41.33 | [TK]D-Fender | ulaw |
| 22:43.01 | marlinc | Okay, I think I'll contact them |
| 22:43.17 | ChannelZ | 722 is nice for internal phones or perhaps through a provider where you're calling another number who uses them as a provider as well, assuming both of you even have wideband phones |
| 22:44.07 | ChannelZ | But for the majority you're never going to get a wideband call from a regular person |
| 22:45.15 | ChannelZ | Speaking of which, anyone notice with Vitelity if you are calling Vitelity-to-Vitelity, the Caller ID is destroyed? |
| 22:47.05 | ChannelZ | Ooh, maybe they fixed it.. it's working now |
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