Re: [asterisk-users] Asterisk and Linphone
My self-compiled Asterisk also shows that speex dependencies are not installed Speex Coder/Decoder Depends on: speex(E), speex_preprocess(E) Can use: speexdsp(E) You'll need to installed the dependencies and re-compile. Doug -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Linphone
On Friday 05 July 2019 at 16:33:56, Jerry Geis wrote: > I have no speex translation > core show translation paths speex > --- Translation paths SRC Codec "speex" sample rate 8000 --- > speex:8000 To slin:8000 : No Translation Path > Does not look good. no paths... Did something not get compiled ? I suspect you're right, but this is now beyond my expertise. I have an Asterisk 13.14.1 system here installed from Debian packages and I have the following: speex:8000 To amr:8000 : (speex@8000)->(slin@8000)->(amr@8000) speex:8000 To amrwb:16000 : (speex@8000)->(slin@8000)->(slin@16000)- >(amrwb@16000) speex:8000 To g723:8000 : No Translation Path speex:8000 To ulaw:8000 : (speex@8000)->(slin@8000)->(ulaw@8000) speex:8000 To alaw:8000 : (speex@8000)->(slin@8000)->(alaw@8000) speex:8000 To gsm:8000 : (speex@8000)->(slin@8000)->(gsm@8000) speex:8000 To g726:8000 : (speex@8000)->(slin@8000)->(g726@8000) speex:8000 To g726aal2:8000 : (speex@8000)->(slin@8000)->(g726aal2@8000) speex:8000 To adpcm:8000 : (speex@8000)->(slin@8000)->(adpcm@8000) speex:8000 To slin:8000 : (speex@8000)->(slin@8000) speex:8000 To slin:12000 : (speex@8000)->(slin@8000)->(slin@12000) speex:8000 To slin:16000 : (speex@8000)->(slin@8000)->(slin@16000) speex:8000 To slin:24000 : (speex@8000)->(slin@8000)->(slin@24000) speex:8000 To slin:32000 : (speex@8000)->(slin@8000)->(slin@32000) speex:8000 To slin:44100 : (speex@8000)->(slin@8000)->(slin@44100) speex:8000 To slin:48000 : (speex@8000)->(slin@8000)->(slin@48000) speex:8000 To slin:96000 : (speex@8000)->(slin@8000)->(slin@96000) speex:8000 To slin:192000 : (speex@8000)->(slin@8000)->(slin@192000) speex:8000 To lpc10:8000 : (speex@8000)->(slin@8000)->(lpc10@8000) speex:8000 To g729:8000 : No Translation Path speex:8000 To speex:16000 : (speex@8000)->(slin@8000)->(slin@16000)- >(speex@16000) speex:8000 To speex:32000 : (speex@8000)->(slin@8000)->(slin@32000)- >(speex@32000) speex:8000 To ilbc:8000 : No Translation Path speex:8000 To g722:16000 : (speex@8000)->(slin@8000)->(g722@16000) speex:8000 To siren7:16000 : No Translation Path speex:8000 To siren14:32000 : No Translation Path speex:8000 To testlaw:8000 : (speex@8000)->(slin@8000)->(testlaw@8000) speex:8000 To g719:48000 : No Translation Path speex:8000 To opus:48000 : No Translation Path speex:8000 To none:8000 : No Translation Path speex:8000 To silk:8000 : No Translation Path speex:8000 To silk:12000 : No Translation Path speex:8000 To silk:16000 : No Translation Path speex:8000 To silk:24000 : No Translation Path Antony. -- "There has always been an underlying argument that we should open up our source code more broadly. The fact is that we are learning from open source and we are opening our code more broadly through Shared Source. Is there value to providing source code? The answer is unequivocally yes." - Jason Matusow, head of Microsoft's Shared Source Program, in response to leaks of Windows source code on the Internet. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Linphone
I have no speex translation ulaw alaw gsm g726 g726aal2 adpcm slin8 slin12 slin16 slin24 slin32 slin44 slin48 slin96 slin192 lpc10 ilbc g722 testlaw ulaw - 9150 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 alaw 9150 - 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 gsm 15000 15000 - 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 g726 15000 15000 15000 -15000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 g726aal2 15000 15000 15000 15000- 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 adpcm 15000 15000 15000 1500015000 - 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 15000 slin8 6000 6000 6000 6000 6000 6000 - 8000 8000 8000 8000 8000 8000 80008000 6000 6000 82506000 slin12 14500 14500 14500 1450014500 14500 8500 - 8000 8000 8000 8000 8000 80008000 14500 14500 14000 14500 slin16 14500 14500 14500 1450014500 14500 8500 8500 - 8000 8000 8000 8000 80008000 14500 14500 6000 14500 slin24 14500 14500 14500 1450014500 14500 8500 8500 8500 - 8000 8000 8000 80008000 14500 14500 14500 14500 slin32 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 - 8000 8000 80008000 14500 14500 14500 14500 slin44 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 - 8000 80008000 14500 14500 14500 14500 slin48 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 8500 - 80008000 14500 14500 14500 14500 slin96 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 8500 8500 -8000 14500 14500 14500 14500 slin192 14500 14500 14500 1450014500 14500 8500 8500 8500 8500 8500 8500 8500 8500 - 14500 14500 14500 14500 lpc10 15000 15000 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 - 15000 17250 15000 ilbc 15000 15000 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 - 17250 15000 g722 15600 15600 15600 1560015600 15600 9600 17500 9000 17000 17000 17000 17000 17000 17000 15600 15600 - 15600 testlaw 15000 15000 15000 1500015000 15000 9000 17000 17000 17000 17000 17000 17000 17000 17000 15000 15000 17250 - core show translation paths speex --- Translation paths SRC Codec "speex" sample rate 8000 --- speex:8000 To g723:8000 : No Translation Path speex:8000 To ulaw:8000 : No Translation Path speex:8000 To alaw:8000 : No Translation Path speex:8000 To gsm:8000: No Translation Path speex:8000 To g726:8000 : No Translation Path speex:8000 To g726aal2:8000 : No Translation Path speex:8000 To adpcm:8000 : No Translation Path speex:8000 To slin:8000 : No Translation Path speex:8000 To slin:12000 : No Translation Path speex:8000 To slin:16000 : No Translation Path speex:8000 To slin:24000 : No Translation Path speex:8000 To slin:32000 : No Translation Path speex:8000 To slin:44100 : No Translation Path speex:8000 To slin:48000 : No Translation Path speex:8000 To slin:96000 : No Translation Path speex:8000 To slin:192000 : No Translation Path speex:8000 To lpc10:8000 : No Translation Path speex:8000 To g729:8000 : No Translation Path speex:8000 To speex:16000 : No Translation Path speex:8000 To speex:32000 : No Translation Path speex:8000 To ilbc:8000 : No Translation Path speex:8000 To g722:16000 : No Translation Path speex:8000 To siren7:16000: No Translation Path speex:8000 To siren14:32000 : No Translation Path speex:8000 To testlaw:8000: No Translation Path speex:8000 To g719:48000 : No Translation Path speex:8000 To opus:48000 : No Translation Path speex:8000 To none:8000 : No Translation Path speex:8000 To silk:8000 : No Translation Path speex:8000 To silk:12000 : No Translation Path speex:8000 To silk:16000 : No Translation Path speex:8000 To silk:24000 : No Translation Path Does not look good. no paths... Did something not get compiled ? Jerry > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check
Re: [asterisk-users] Asterisk and Linphone
On Friday 05 July 2019 at 16:03:42, Jerry Geis wrote: > I think this is what your looking for: > [Jul 5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a > codec translation path: (speex) -> (speex32) Indeed, it was. > My linphone side only has speex@32K enabled. > > My extension definition has: > disallow=all > allow=speex > allow=speex16 > allow=speex32 > allow=g722 > allow=ulaw > allow=alaw > allow=gsm > > It looks like its the codec translation ? So then I enabled speex and > speex32 on Linphone Got past that - I presume it will use speex32 for > audio... You can always see which codec is in use by doing a SIPpacket capture and looking at the above negotiation exchange to see what got agreed on. > But then I am trying to place that call in a conference (confbridge) and I > get this error: > Unable to find a codec translation path: (slin) -> (speex) > so I think then it hangs up. Try "core show translation" on your Asterisk command line and check that the table has an entries in both directions for speex (left) to slin (top) and slin (left) to speex (top). The numbers tell you how many microseconds *your* server takes to transcode 1 second of audio between the two codecs. You can also try "core show translation paths speex" to get a list of the codecs which can and cannot be converted to, with a guide to the method used for trancoding that combination where possible. Antony. -- All matter in the Universe can be placed into one of two categories: 1. Things which need to be fixed. 2. Things which need to be fixed once you've had a few minutes to play with them. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Linphone
I think this is what your looking for: Found RTP audio format 119 Found audio description format speex for ID 119 Capabilities: us - (speex|speex16|speex32|g722|ulaw|alaw|gsm), peer - audio=(speex32)/video=(nothing)/text=(nothing), combined - (speex32) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing) Peer audio RTP is at port 192.168.1.176:7078 [Jul 5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a codec translation path: (speex32) -> (speex) ^M^[[Kdevgeis*CLI> ^M^[[0K[Jul 5 09:55:34] WARNING[19832]: channel.c:5751 set_format: Unable to find a codec translation path: (speex) -> (speex32) My linphone side only has speex@32K enabled. My extension definition has: disallow=all allow=speex allow=speex16 allow=speex32 allow=g722 allow=ulaw allow=alaw allow=gsm It looks like its the codec translation ? So then I enabled speex and speex32 on Linphone Got past that - I presume it will use speex32 for audio... But then I am trying to place that call in a conference (confbridge) and I get this error: Unable to find a codec translation path: (slin) -> (speex) so I think then it hangs up. What do I do about that ? - thanks Jerry On Fri, Jul 5, 2019 at 8:22 AM Jerry Geis wrote: > Hi all - I am using asterisk 13.27.0 with Linphone. > I turned off all codes on linphone except the one I want to try. For > example: > opus and speex (so only one enabled at a time). > Then did this same on asterisk for the linphone extension. > disallow=all > allow=speex > > (for example). > > Then I place my call and the call fails. if I enable something like gsm, > ulaw, alaw the call works fine. Why does the call fail with opus and speex ? > Thanks, > > Jerry > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk and Linphone
On Friday 05 July 2019 at 14:22:22, Jerry Geis wrote: > Hi all - I am using asterisk 13.27.0 with Linphone. > I turned off all codes on linphone except the one I want to try. For > example: > opus and speex (so only one enabled at a time). > Then did this same on asterisk for the linphone extension. > disallow=all > allow=speex > > (for example). > > Then I place my call and the call fails. if I enable something like gsm, > ulaw, alaw the call works fine. Why does the call fail with opus and speex? Show us the SIP INVITE from Linphone and the response from Asterisk where they negotiate codecs - that should tell us why they disagree. Antony. -- The lottery is a tax for people who can't do maths. Please reply to the list; please *don't* CC me. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Asterisk and Linphone
Hi all - I am using asterisk 13.27.0 with Linphone. I turned off all codes on linphone except the one I want to try. For example: opus and speex (so only one enabled at a time). Then did this same on asterisk for the linphone extension. disallow=all allow=speex (for example). Then I place my call and the call fails. if I enable something like gsm, ulaw, alaw the call works fine. Why does the call fail with opus and speex ? Thanks, Jerry -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Asterisk to Linphone sound playback delay, and then choppy
Hi, I've got this PXA270 board set-up with Linphone 1.2.0 and am trying to get linphonec to work with Asterisk. I have the echo test working, but when I dial in to this, to voicemail or anything else using Playback() to play a sample, I hear nothing for ages (10-15 secs) and then little sections. With the echo test, I get the tail of the message (...pressing the pound keypressing the pound key...) echoed rather well, but sound quality is a little poor. I have Sipomatic tested over the same cross-over network connection... perfect. I have made calls to my mobile via SipGate... takes a while to start, but then perfect. I thought it might be because I was using OSS emulation, but I recompiled to use ALSA pure (without 'Jack', that had issues) and got the exact same results. Does anyone have a working config they could post, or any idea what may be the issue? I am running Asterisk on a 1600Mhz laptop, so I doubt it's short of cycles - and I don't think I did anything special as regards config. I used this as a template for the linphone config http://www.voip-info.org/wiki/index.php?page=Asterisk+phone+linphone I've tried the apt-get asterisk (1.2 I think) on Debian, and also this PXE version (booting my laptop from the board) http://www.automated.it/asterisk/pxeindex.html And this as a starting point for my Asterisk config files. http://www.automated.it/guidetoasterisk.htm#_Toc49248767 Regards, Adam ___ --Bandwidth and Colocation provided by Easynews.com -- Asterisk-Users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
