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Alembic – Asterisk 11

I’ve had years of experience using ODBC for CDR, SIP, and extensions with Asterisk. One thing that has been problematic in the past is with documentation as far as database tables changing between versions (even within minor releases, though that was back in the 1.4 days). I was excited to see there is a plan for better managing that on Asterisk 12 via Alembic. All that being said, are there any plans to implement that with Asterisk 11, since that is the current LTS release? Or are we pretty sure the table structure won’t be changing within that version through the rest of its lifespan, making such an effort a waste?



Webrtc And Adventures With Asterisk 11


I spent the past week experimenting with webrtc + asterisk 11.9.0-rc1 +
opus/vb8 codec patch. This is interesting technology and I try to find out how to connect all the moving parts.

Neither sipml5 or jssip works with calls to asterisk, audio/video doesn’t matter. WARNING[977][C-00000005] chan_sip.c: Rejecting secure audio stream without encryption details: audio 35684 RTP/SAVPF 109 0 8 101
–> Asterisk sends “SIP/2.0 488 Not acceptable here”

I’ve tried both sipml5 and jssip softphones and they both work. Even video + confbridge works with some minor quirks (lost connections sometimes, I guess plain old nat issues). Just relaying audio+video with confbridge to a handful of participants seems to use quite a bit of cpu thought.

This works, but Confbridge is not very happy about a channel with video
(vp8) and not audio and is printing this 80 times a second:

WARNING[8919][C-00000000] channel.c: Unable to find a codec translation path from (vp8) to (slin)
WARNING[8919][C-00000000] chan_sip.c: Asked to transmit frame type slin, while native formats is (vp8) read/write = unknown/unknown WARNING[8919][C-00000000] channel.c: Don’t know any of (vp8) formats

How do you think about adding webrtc to a existing Asterisk/Kamailio environment? Do you use kamailio (websockets) as a front, a dedicated webrtc asterisk or something like webrtc2sip?

How do you use / plan to implement webrtc in your environment?

Any feedback is welcome. Thanks!

Possible Dahdi Compile Problem

I am having problems with system crashes in a dahdi/asterisk compile.

Asterisk To Microsoft Lync2013?

Are they any gotchas to be aware of in getting Asterisk and Lync 2013
talking to each other using SIP? Or is Lync a pretty standard implementation of SIP?

Cheers Tony

No Voice When The Calls Come From Internet


I have trouble establishing a call between between two SIP phones. One sip phone is, with asterisk server, at home behind a firewall. The second sip phone is an iPhone with 3G wireless connection.

When I call from the SIP device at home the SIP account on the Internet
(iphone + 3G) I can hear voice on both devices. But when I call from the Internet to my home SIP I get the ring but when I answer I don’t hear any thing!

I have asterisk v11 installed at home all the incoming TCP/UDP 5060 and a defined UDP range in rtp.conf forwarded to my Asterisk server.

Do you have any idea when the voice is heard only when the call is from my local network to the Internet and not in the other direction ?

Nevertheless, when both SIP devices are in the same home IP network the call is made without any problem whatever who starts the call.


PJSIP In Dialog OPTIONS Method Handling

Hi everyone, I am running asterisk with release 12.1.0.rc3 and PJSIP. I have a peer which sends OPTIONS method for session keep-alive, and the asterisk is not responding to it. That of course disconnects the call after a few minutes.

Is there a settings in the PJSIP.conf to respond to in dialog OPTIONS
method? Looking at the documentation I haven’t seen it. Does anybody know a workaround?

Thanks, Yaron.