Re: [asterisk-users] Can't get G.726 to work.
Chris Maciejewski wrote: Found unknown media description format G726-16 for ID 102 It's right there. And asterisk is replying with 488 Not acceptable here Asterisk does not support G726-16, it only supports G726-32. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
Hi Kevin, Thanks for your reply. I switched to G726 32Kbps but still no luck: INVITE [SIP headers omitted] v=0 o=1 1291673978 653998617 IN IP4 192.168.7.55 s=- c=IN IP4 78.105.1.131 t=0 0 m=audio 8002 RTP/AVP 104 101 a=rtpmap:104 G726-32/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Console SIP debug output: [May 22 16:48:20] DEBUG[6071]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to Off Found RTP audio format 104 Found RTP audio format 101 Peer audio RTP is at port 78.105.1.131:8002 Found audio description format G726-32 for ID 104 Found audio description format telephone-event for ID 101 Got unsupported a:fmtp in SDP offer Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) [May 22 16:48:20] NOTICE[6071]: chan_sip.c:7495 process_sdp: No compatible codecs, not accepting this offer! I note Got unsupported a:fmtp in SDP offer from RFC 2327: a=fmtp:format format specific parameters This attribute allows parameters that are specific to a particular format to be conveyed in a way that SDP doesn't have to understand them. The format must be one of the formats specified for the media. Format-specific parameters may be any set of parameters required to be conveyed by SDP and given unchanged to the media tool that will use this format. It is a media attribute, and is not dependent on charset. Is Twinkle sending this SDP incorrectly? Or some other issue? Thanks Chris 2009/5/22 Kevin P. Fleming kpflem...@digium.com: Chris Maciejewski wrote: Found unknown media description format G726-16 for ID 102 It's right there. And asterisk is replying with 488 Not acceptable here Asterisk does not support G726-16, it only supports G726-32. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
On 22 May 2009, at 16:55, Chris Maciejewski wrote: Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Codec not enabled on that peer? S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
Chris Maciejewski wrote: Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) 'us' does not include g726, so you have not configured your SIP user/peer to support G.726. I note Got unsupported a:fmtp in SDP offer No, that is not relevant. Asterisk's SDP parser does not pay much attention to a:fmtp entries at this time. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
Yes, I was missing allow=g726 for this peer :-( Playback(/var/lib/asterisk/moh/fpm-sunshine) works OK now, however I still can't get MeetMe to work. Before I had similar problem, when MeetMe wasn't working with GSM codec because I was missing .gsm audio files. I suspect now it is the same problem, as I don't have audio files for G726? Will try converting .pcm to .g726 and see if that will fix MeetMe issue. Regards, Chris 2009/5/22 Steve Howes st...@geekinter.net: On 22 May 2009, at 16:55, Chris Maciejewski wrote: Capabilities: us - 0x100f (g723|gsm|ulaw|alaw|g722), peer - audio=0x800 (g726)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x0 (nothing) Codec not enabled on that peer? S ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
Chris Maciejewski wrote: Yes, I was missing allow=g726 for this peer :-( Playback(/var/lib/asterisk/moh/fpm-sunshine) works OK now, however I still can't get MeetMe to work. Before I had similar problem, when MeetMe wasn't working with GSM codec because I was missing .gsm audio files. I suspect now it is the same problem, as I don't have audio files for G726? Will try converting .pcm to .g726 and see if that will fix MeetMe issue. If you have codec_g726 loaded, you should be able to use prompt files in any format that Asterisk can transcode from/to. 'core show translations' should show you what formats Asterisk can convert to and from G.726. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
I do have codec_g726 loaded. As I mentioned before Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite there is only fpm-sunshine.wav file. It is only MeetMe which is not working: -- SIP/OpenSER-08208098 Playing 'entering-conf-number.slin' (language 'en') [May 22 18:07:04] WARNING[16881]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/OpenSER-08208098 for entering-conf-number -- Executing [...@services:7] SayNumber(SIP/OpenSER-08208098, 1) in new stack -- SIP/OpenSER-08208098 Playing 'digits/1.slin' (language 'en') -- Executing [...@services:8] Wait(SIP/OpenSER-08208098, 1) in new stack -- Executing [...@services:9] MeetMe(SIP/OpenSER-08208098, 11,MI) in new stack == Parsing '/etc/asterisk/meetme.conf': == Found -- Created MeetMe conference 1023 for conference '11' -- SIP/OpenSER-08208098 Playing 'vm-rec-name.slin' (language 'en') -- Hungup 'DAHDI/pseudo-1131226973' 2009/5/22 Kevin P. Fleming kpflem...@digium.com: Chris Maciejewski wrote: Yes, I was missing allow=g726 for this peer :-( Playback(/var/lib/asterisk/moh/fpm-sunshine) works OK now, however I still can't get MeetMe to work. Before I had similar problem, when MeetMe wasn't working with GSM codec because I was missing .gsm audio files. I suspect now it is the same problem, as I don't have audio files for G726? Will try converting .pcm to .g726 and see if that will fix MeetMe issue. If you have codec_g726 loaded, you should be able to use prompt files in any format that Asterisk can transcode from/to. 'core show translations' should show you what formats Asterisk can convert to and from G.726. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
Chris Maciejewski wrote: I do have codec_g726 loaded. As I mentioned before Playback(/var/lib/asterisk/moh/fpm-sunshine) works just fine - despite there is only fpm-sunshine.wav file. It is only MeetMe which is not working: -- SIP/OpenSER-08208098 Playing 'entering-conf-number.slin' (language 'en') [May 22 18:07:04] WARNING[16881]: app_playback.c:447 playback_exec: ast_streamfile failed on SIP/OpenSER-08208098 for entering-conf-number This is not MeetMe, it's Playback. You specified a filename with '.slin' in it to Playback, so then Asterisk attempts to find a filename called 'entering-conf-number.slin.foo' where foo is the possible formats that Asterisk could transcode from. Filenames specified to Playback should not include the format extension. -- Executing [...@services:7] SayNumber(SIP/OpenSER-08208098, 1) in new stack -- SIP/OpenSER-08208098 Playing 'digits/1.slin' (language 'en') This did not fail. The .slin extension was added by ast_streamfile after it found the correct format to play for this channel. -- Executing [...@services:8] Wait(SIP/OpenSER-08208098, 1) in new stack -- Executing [...@services:9] MeetMe(SIP/OpenSER-08208098, 11,MI) in new stack == Parsing '/etc/asterisk/meetme.conf': == Found -- Created MeetMe conference 1023 for conference '11' -- SIP/OpenSER-08208098 Playing 'vm-rec-name.slin' (language 'en') Again, this did not fail. -- Hungup 'DAHDI/pseudo-1131226973' The only failure of any kind that I see in this log is the call to Playback. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com www.asterisk.org ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Can't get G.726 to work.
On Fri, 22 May 2009, Kevin P. Fleming wrote: This is not MeetMe, it's Playback. You specified a filename with '.slin' in it to Playback, so then Asterisk attempts to find a filename called 'entering-conf-number.slin.foo' where foo is the possible formats that Asterisk could transcode from. Filenames specified to Playback should not include the format extension. The number of times I and others have forgotten that Asterisk chooses the file based on codec and file type availability automagically makes me wonder if it makes sense to change this so that if the file type (from the list of known file types based on the formats loaded) is specified, use it. Otherwise, proceed with the current code. Thanks in advance, Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
