Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
These show that a proper bridging tech module cannot be found to run ConfBridge. The debug message showing that a capability for ulaw couldn't be found was a buggy debug message which has now been fixed (it isn't a codec capability that can't be found, but a bridge capability). You need to make sure the bridge_softmix.so is loaded. I would load the other bridge_*.so modules too just for fun. Thanks! Indeed I was missing bridge_softmix.so module. Once added to modules.conf ConfBride works as expected :) -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
Attach a debug[1] log so we can see what is happening. [1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information debug logs below: Asterisk 1.8.4: http://pastebin.com/DFnKgSse Asterisk trunk r319661: http://pastebin.com/B19tdbxJ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
Hi, I am trying to use ConfBridge application, but it throws Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw) error. Please see console output below. -- Executing [501@services:9] ConfBridge(SIP/OpenSER-0005, 1001) in new stack [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:404 join_conference_bridge: Trying to find conference bridge '1001' [May 19 13:36:05] DEBUG[7452]: bridging.c:475 ast_bridge_new: Failed to find a bridge technology to satisfy capabilities 0x4 (ulaw) [May 19 13:36:05] DEBUG[7452]: app_confbridge.c:368 destroy_conference_bridge: Destroying conference bridge '1001' [May 19 13:36:05] ERROR[7452]: app_confbridge.c:435 join_conference_bridge: Conference bridge '1001' could not be created. Could someone please let me know what is required to make it work? Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] ConfBridge - Failed to find a bridge technology to satisfy capabilities
What version of Asterisk are you using? ConfBridge was rewritten in trunk and would be good to see if you have the same issue. Hi Paul, I am using 1.8.4. Just tried with the latest trunk (SVN-trunk-r319661) and it still doesn't work, this time throwing error as below: -- Executing [501@services:3] ConfBridge(SIP/OpenSER-0001, 10001) in new stack [May 19 16:11:58] DEBUG[30778]: app_confbridge.c:775 join_conference_bridge: Trying to find conference bridge '10001' [May 19 16:11:58] DEBUG[30778]: app_confbridge.c:736 destroy_conference_bridge: Destroying conference bridge '10001' [May 19 16:11:58] ERROR[30778]: app_confbridge.c:814 join_conference_bridge: Conference bridge '10001' could not be created. Regards, Chris -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Dial with r option doesn't use 'ring' tone as defined in indications.conf
Sorry, I wasn't maybe precise in my question. What I am looking for is to use custom cadences (as defined in indications.conf) for ring tone generated by 'r' option in a Dial command. Just found this patch: https://issues.asterisk.org/view.php?id=14504 which does exactly this: Dial(SIP/1234,60,r(myring)) Thanks Chris 2009/6/13 David Backeberg dbackeb...@gmail.com: On Sat, Jun 13, 2009 at 11:27 AM, Chris Maciejewskich...@wima.co.uk wrote: Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. When I now dial with a SIP phone - 123 I can hear nice UK ring tone as per [uk] definition, however when I dial 321 ring tone is different. Is there any way to fix this? Silly question... you have reloaded the settings you changed for indications, right? I don't know the right reload command but you can certainly do it with an asterisk restart. ___ -- 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
[asterisk-users] Dial with r option doesn't use 'ring' tone as defined in indications.conf
Hi, Just noticed Asterisk is not playing 'ring' tone as defined in indications.conf when Dial command is used with 'r' option. For example: [test] exten = 123,1,PlayTones(ring) exten = 123,n,Wait(5) exten = 123,n,Playback(demo-congrats) exten = 123,n,Hangup() exten = 321,1,Dail(LOCAL/1...@test/n,60,r) When I now dial with a SIP phone - 123 I can hear nice UK ring tone as per [uk] definition, however when I dial 321 ring tone is different. Is there any way to fix this? Thanks Chris ___ -- 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
[asterisk-users] CDR after SIP blind transfer.
Hi,
I can't get Asterisk to save CDRs for calls transferred via SIP blind transfer.
My extensions.conf:
[globals]
__TRANSFER_CONTEXT = transfer
[common]
exten = 123,1,Playback(demo-congrats)
exten = 123,n,Hangup()
exten = _0X.,1,Dial(SIP/${ext...@pstn-gw,60)
exten = _0X.,n,Hangup()
exten = i,1,Hangup()
exten = h,1,Hangup()
exten = t,1,Hangup()
[transfer]
exten = 123,1,Goto(common,${EXTEN},1)
Scenario A:
SIP Phone dials 123 and hangs up after 10 seconds.
CDR is recorded just fine.
Scenario B:
SIP Phone dials 02088441234 which is routed to the external peer.
After 10 seconds call is transferred (blindly) to extension 123. After
another 10 seconds external peer hangs up.
Problem: there is only one CDR recorded for the first 10 seconds long
call. Second part of the call, after 02088441234 was transferred to
123 is NOT recorded.
Is there any way to force Asterisk to record CDR in scenario B
(without using LOCAL channel)?
Regards,
Chris
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Re: [asterisk-users] MeetMe not working with GSM codec?
Hi Martin, Yes, I do have GSM compiled for sure. $asterisk -r -x core show codecs audio Disclaimer: this command is for informational purposes only. It does not indicate anything about your configuration. INTBINARYHEX TYPE NAME DESC 1 (1 0) (0x1) audio g723 (G.723.1) 2 (1 1) (0x2) audiogsm (GSM) 4 (1 2) (0x4) audio ulaw (G.711 u-law) 8 (1 3) (0x8) audio alaw (G.711 A-law) 16 (1 4) (0x10) audio g726aal2 (G.726 AAL2) 32 (1 5) (0x20) audio adpcm (ADPCM) 64 (1 6) (0x40) audio slin (16 bit Signed Linear PCM) 128 (1 7) (0x80) audio lpc10 (LPC10) 256 (1 8)(0x100) audio g729 (G.729A) 512 (1 9)(0x200) audio speex (SpeeX) 1024 (1 10)(0x400) audio ilbc (iLBC) 2048 (1 11)(0x800) audio g726 (G.726 RFC3551) 4096 (1 12) (0x1000) audio g722 (G722) I will open a bug report. Regards, Chris 2009/5/22 Martin asteriskl...@callthem.info: it should work just fine; do you have the GSM codec compiled/loaded core show modules like codec_gsm ... ? OR that particular version has a BUG... Martin On Thu, May 21, 2009 at 3:56 AM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- 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 ___ -- 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] MeetMe not working with GSM codec?
Hi Dhaval, The reason confno '12' is not found in meetme.conf is because I am using MySQL as realtime config backend. See few lines below there is: [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12' My meetme.conf: [general] audiobuffers=32 logmembercount=yes schedule=no 2009/5/22 DHAVAL INDRODIYA dhaval.it01...@gmail.com: can you look on this from your debug app_meetme.c:3030 find_conf: The requested confno is '12'? == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf == Found [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference its on line number 318 it seems that you doesent specify valid conference number can you post meetme.conf regards Dhaval On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- 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 ___ -- 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] MeetMe not working with GSM codec?
Thanks Kinjal! Missing sound files was the problem. There were no .gsm files in my sounds directory. Despite console shows .slin, the actual files required are .gsm. Once I copied .gsm into /var/lib/asterisk/sounds everything works OK. Regards, Chris 2009/5/22 Kinjal Dixit kinjal.di...@gmail.com: On an entirely unrelated note, do you have the gsm asterisk sounds installed? Maybe that vm-*.slin files don’t exist. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Chris Maciejewski Sent: Friday, May 22, 2009 12:53 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] MeetMe not working with GSM codec? Hi Dhaval, The reason confno '12' is not found in meetme.conf is because I am using MySQL as realtime config backend. See few lines below there is: [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:1478 mysql_reconnect: MySQL RealTime: Connection okay. [May 21 09:33:23] DEBUG[6872]: res_config_mysql.c:365 realtime_mysql: MySQL RealTime: Retrieve SQL: SELECT * FROM conference WHERE confno = '12' My meetme.conf: [general] audiobuffers=32 logmembercount=yes schedule=no 2009/5/22 DHAVAL INDRODIYA dhaval.it01...@gmail.com: can you look on this from your debug app_meetme.c:3030 find_conf: The requested confno is '12'? == Parsing '/etc/asterisk/meetme.conf': [May 21 09:33:23] DEBUG[6872]: config.c:1306 config_text_file_load: Parsing /etc/asterisk/meetme.conf == Found [May 21 09:33:23] DEBUG[6872]: app_meetme.c:3082 find_conf: 12 isn't a valid conference its on line number 318 it seems that you doesent specify valid conference number can you post meetme.conf regards Dhaval On Thu, May 21, 2009 at 2:26 PM, Chris Maciejewski ch...@wima.co.uk wrote: Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- 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 ___ -- 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 ___ -- 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
[asterisk-users] Can't get G.726 to work.
Hi, I have both codec_g726.so and format_g726.so loaded: r...@test:~# asterisk -r -x module show | grep 726 codec_g726.so ITU G.726-32kbps G726 Transcoder 0 format_g726.so Raw G.726 (16/24/32/40kbps) data 0 But when I try to dial into Asterisk with Twinkle softphone using G.726 codec: INVITE . [SIP headers omitted] v=0 o=1 1615261284 506628667 IN IP4 192.168.7.55 s=- c=IN IP4 78.105.1.131 t=0 0 m=audio 8002 RTP/AVP 102 101 a=rtpmap:102 G726-16/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 Console shows: [May 22 10:29:34] DEBUG[6071]: chan_sip.c:4222 do_setnat: Setting NAT on RTP to Off Found RTP audio format 102 Found RTP audio format 101 Peer audio RTP is at port 78.105.1.131:8002 Found unknown media description format G726-16 for ID 102 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=0x0 (nothing)/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 10:29:34] NOTICE[6071]: chan_sip.c:7495 process_sdp: No compatible codecs, not accepting this offer! And asterisk is replying with 488 Not acceptable here Any help and suggestions very much appreciated. Regards, Chris ___ -- 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.
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.
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
[asterisk-users] MeetMe not working with GSM codec?
Hi, I am not sure if I am doing something wrong, but I can't get MeetMe to work with GSM codec (Asterisk 1.6.1 SVN r190371). My config files below: sip.conf: [general] context=common canreinvite=no bindport=5060 bindaddr=78.105.1.127 disallow=all allow=alaw allow=gsm rtptimeout=600 rtpholdtimeout=3600 rtpkeepalive=30 nat=no jbenable=yes tcpenable=no realm=dev-sip.wima.co.uk [1] type=friend secret=test host=dynamic nat=yes -- - extensions.conf: - [common] exten = 501,1,MeetMe(12,MI) exten = 501,n,Hangup() exten = i,1,Hangup() exten = h,1,Hangup() exten = t,1,Hangup() Everything works OK when ALAW is used - see http://pastebin.com/f7222a6d3 but with GSM Asterisk hangs up just after starting MeetMe application - see http://pastebin.com/f78d04c95 line 327. Is there a problem with MeetMe app or I need to adjust my configuration? Regards, Chris ___ -- 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
[asterisk-users] Capture Server header in SIP reply.
Hi,
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
For example:
[default]
exten = _X.,1,Dial(SIP/u...@domain,30,M(GetOtherPartyInfo))
exten = _X.,n,Hangup()
[macro-GetOtherPartyInfo]
exten = s,1,NoOp(SIP Server: ${SIP_HEADER(Server,1)})
unfortunately the above doesn't seem to work:
-- Executing [...@macro-getotherpartyinfo:1]
NoOp(SIP/dev-sip.domain.net-08dbb610, SIP src_server: ) in new
stack
Is there any way to capture SIP headers from reply messages generated
by a called party?
Regards,
Chris
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Re: [asterisk-users] Capture Server header in SIP reply.
Hi David,
Thanks for your post.
Unfortunately SIP_HEADER(FROM) is not an option for me.
What I want to do is record in CDRs User-Agent header of calling
party (this can be easily done with ${CHANNEL(useragent)}), and SIP
Server header of called party (from 200 OK response to INVITE
generated by Asterisk).
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 6:43 AM, Chris Maciejewski ch...@wima.co.uk wrote:
I am trying to capture Server header in a 200 OK reply message.
My idea was to use Dail(SIP/u...@domain,30,M(GetOtherPartyInfo)),
and inside of GetOtherPartyInfo macro use SIP_HEADER function.
unfortunately the above doesn't seem to work:
Is there any way to capture SIP headers from reply messages generated
by a called party?
http://www.voip-info.org/wiki/view/Asterisk+func+sip_header
You might prefer the SIP_HEADER(FROM) field.
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Re: [asterisk-users] Capture Server header in SIP reply.
User-Agent header is present in SIP *request* i.e. INVITE received
by Asterisk from UAC.
RFC 3261 - 20.41 User-Agent
The User-Agent header field contains information about the UAC
originating the request. The semantics of this header field are
defined in [H14.43].
Revealing the specific software version of the user agent might allow
the user agent to become more vulnerable to attacks against software
that is known to contain security holes. Implementers SHOULD make
the User-Agent header field a configurable option.
Example:
User-Agent: Softphone Beta1.5
Server header is present in SIP *response* i.e. 200 OK generated
by UAS to INVITE generated by Asterisk.
RFC 3261 - 20.35 Server
The Server header field contains information about the software used
by the UAS to handle the request.
Revealing the specific software version of the server might allow the
server to become more vulnerable to attacks against software that is
known to contain security holes. Implementers SHOULD make the Server
header field a configurable option.
Example:
Server: HomeServer v2
My scenario:
Phone 1 - INVITE [1] - Asterisk -- INVITE [2] -- Phone 2
--- 200
OK [3] ---
What I want to do is capture Server header in 200 OK reply
generated by Phone 2.
2009/5/17 David Backeberg dbackeb...@gmail.com:
On Sun, May 17, 2009 at 9:04 AM, Chris Maciejewski ch...@wima.co.uk wrote:
Maybe you need a better name for it than server. To me server means
the hostname / address of the other side of the SIP conversation, aka:
FROM.
You can use SipAddHeader to make your own X-blah tags for your
packets, and then pick them off on the other side. I don't seem to
understand what you mean by 'server', despite my command of the
english language. Perhaps you want
${SIPUSERAGENT}
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[asterisk-users] SHARED() variables and ZOMBIE channel
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like:
[default]
exten = _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL})
exten = _X.,n,Dial(SIP/u...@domain.net,30,M(getCalledInfo))
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CHANNEL})})
[macro-getCalledInfo]
exten =
s,1,Set(SHARED(X-DST-CODEC,${X-SRC_CHANNEL})=${CHANNEL(audionativeformat)})
The above works great, however there is a problem when call is
transferred via SIP attended transfer and channel is renamed to
ChannelZOMBIE.
-- Executing [...@default:1] Set(SIP/somechannelZOMBIE,
CDR(DST_CODEC)=) in new stack
Is there any workaround for the above issue?
Regards,
Chris
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Re: [asterisk-users] SHARED() variables and ZOMBIE channel
2009/5/17 Tilghman Lesher tilgh...@mail.jeffandtilghman.com:
On Sunday 17 May 2009 09:10:47 Chris Maciejewski wrote:
Hi,
I am using SHARED() function to push destination channel info (i.e.
audio codec) into source channel, in order to record into a customer
CDR field.
My dialplan looks like:
[default]
exten = _X.,1,Set(_X-SRC_CHANNEL=${CHANNEL})
exten = _X.,n,Dial(SIP/u...@domain.net,30,M(getCalledInfo))
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CHANNEL})})
[macro-getCalledInfo]
exten =
s,1,Set(SHARED(X-DST-CODEC,${X-SRC_CHANNEL})=${CHANNEL(audionativeformat)})
The above works great, however there is a problem when call is
transferred via SIP attended transfer and channel is renamed to
ChannelZOMBIE.
-- Executing [...@default:1] Set(SIP/somechannelZOMBIE,
CDR(DST_CODEC)=) in new stack
Is there any workaround for the above issue?
I suppose you could use CUT to guarantee that the ZOMBIE portion won't show
up in the channel name, i.e.
exten = h,1,Set(CDR(DST_CODEC)=${SHARED(X-DST-CODEC,${CUT(CHANNEL,,1)})})
I tried that already, but Asterisk throws the following error:
-- Executing [...@default:1] Set(SIP/OpenSER-0831a618ZOMBIE,
X-CHAN-NAME=SIP/OpenSER-0831a618) in new stack
[May 17 18:24:32] ERROR[6101]: func_global.c:106 shared_read: Channel
'SIP/OpenSER-0831a618' not found! Variable 'X-DST-CODEC' will be
blank.
as OpenSER-0831a618 doesn't exist any more. Looks like maybe SHARED()
variables are not inherited by ZOMBIE channel?
--
Tilghman
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Re: [asterisk-users] enum agi interesting problem
Maybe it is something to do with AGI - Dial command.
IFAIK you can't control Dial via AGI script.
From http://www.voip-info.org/wiki/view/Asterisk+AGI :
Dialing out
If the AGI application dials outward by executing Dial, control over
the call returns to the dialplan and the script loses contact with the
Asterisk server. The script continues to run in the background by
itself and is free to clean up and do post-dial processing.
If you want your application to initiate a call out without being
started through the dialplan:
* Asterisk auto-dial out Move (not copy) a file into an Asterisk
spool directory and a call will be placed
* Asterisk Manager API Use the Originate command
Regards,
Chris
2009/5/13 Dan Caescu dcae...@eqnet.us:
Forget the typo (s/ANSWERED/ANSWER/g)
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Dan Caescu
Sent: Tuesday, May 12, 2009 7:07 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] enum agi interesting problem
Hi,
I am having a strange problem with enum and AGI.
Here is what happens:
I have in my agi something like that:
foreach my $resolver (e164.arpa, e164.info, e164.org) {
my @enums = get_enums($phone, $resolver);
foreach my $enum (@enums) {
$dialstring = $enum . |90|HL( .
($maxtime * 60 * 1000) . :6:3);
$res = $AGI-exec(DIAL $dialstring);
$answeredtime =
$AGI-get_variable(ANSWEREDTIME);
$dialstatus =
$AGI-get_variable(DIALSTATUS);
print LOGFILE Dialstring: $dialstr
DIALSTATUS: $dialstatus\n;
$callstart = time();
if ($dialstatus eq ANSWERED) { last; }
}
}
}
Here’s the output from my logfile:
Call 1:
Dialstring:
sip/16416418003569...@tollfree.sip-happens.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418003569...@sip.tollfreegateway.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring: sip/18003569...@tf.voipmich.com|90|HL(576:6:3)
DIALSTATUS: ANSWER
Call 2:
Dialstring: sip/18002662...@tf.voipmich.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418002662...@sip.tollfreegateway.com|90|HL(576:6:3)
DIALSTATUS:
Dialstring:
sip/16416418002662...@tollfree.sip-happens.com|90|HL(576:6:3)
DIALSTATUS: ANSWER
And so on.
The call gets answered the first time (call 1 – through sip-happens, call 2,
through voipmich).
Problem is that after I hang up , it doesn’t return a status, so it cycles
through the loop and dials the rest of the entries. The last one gets
dialstatus.
I believe it’s a stupid mistake but I cannot think of anything right now.
Any ideas?
Thanks,
Dan
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[asterisk-users] How to send 404 Not found SIP reply?
Hi, I am trying to send 404 Not found reply, without any luck with the following: exten = 555,1,Playback(you-dialed-wrong-number,noanswer) exten = 555,n,Playback(check-number-dial-again,noanswer) exten = 555,n,Congestion() However the above results in 500 Service Unavailable being send out. What would be the correct application/function to generate 404 Not found? Thanks for help, Chris ___ -- 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] How to send 404 Not found SIP reply?
Thank you all for help! What I was trying to achieve was: UA Asterisk - INVITE - --- 100 Trying -- 183 Sess. Prog (sdp) - [ here we play You dialled wrong... ] -- 404 Not found - And all is needed to do this, is to use correct 'causecode' as Hangup parameter :-) exten = i,1,Playback(you-dialed-wrong-number,noanswer) exten = i,n,Playback(check-number-dial-again,noanswer) exten = i,n,Hangup(1) ; - NOTE: causecode 1 for 404 Not found Best regards, Chris ___ -- 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] How to send 404 Not found SIP reply?
Yes, 'causecode' parameter of Hangup application was missing at: http://www.voip-info.org/wiki/view/Asterisk+cmd+Hangup I have added 'causecode' to the above wiki page now. Thanks for your help, Chris 2009/4/16 Tilghman Lesher tilgh...@mail.jeffandtilghman.com: On Thursday 16 April 2009 10:28:38 ContactTel Business wrote: Exten = _X.,1,Busy() or playback not found.. just a catch all... or modify source to add another kind of dialplan entry etc.. Actually, that should send a 486 Busy here. Close, though. The OP could instead do a Hangup(1), Hangup(2), Hangup(3), or Hangup(26). All of these cause codes map to a SIP status 404. See RFC 3398 for the complete mapping of cause codes to SIP status codes. -- Tilghman ___ -- 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
[asterisk-users] X-Asterisk-HangupCause - how to disable this?
Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( Thanks for help. Chris ___ -- 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] X-Asterisk-HangupCause - how to disable this?
For curiosity's sake, what are the troubling consequences of having those headers included ? My PSTN termination provider sometimes replies with SIP/2.0 513 Message too big to my BYEs with additional headers included. Just wanted to check if this is the reason, or maybe it is related to something else. 2009/3/15 Olivier oza-4...@myamail.com: 2009/3/15 Chris Maciejewski ch...@wima.co.uk Hi, Is there any way to tell Asterisk not to generate additional headers like: X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 I can't find any relevant option in sip.conf file :-( For curiosity's sake, what are the troubling consequences of having those headers included ? Thanks for help. Chris ___ -- 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 ___ -- 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
[asterisk-users] Info about dstchannel
Hi,
Is it possible to get information about SIP destination channel (created
after Dial command) somehow?
For example I would like to know what codec was used. I can do this for
originating channel with:
${CHANNEL(audionativeformat)}
but not sure how to do the same for destination channel?
Any suggestions?
Chris
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Re: [asterisk-users] how to detect pickup...
Hi, One of the solutions would be to overwrite standard *8 behaviour with your custom macro that will 1) pickup a call as usual b) send notification via AMI or whatever else you want. This can be done with [applicationmap] in features.conf - see http://www.voip-info.org/wiki-Asterisk+config+features.conf Regards, Chris 2008/9/18 Gergo Csibra [EMAIL PROTECTED]: Hello asterisk-users, My SIP phones are in pickupgroup, and if some of them ringing from other phone can pick up with *8 as usual. But I want to know if this happen. I've tried the a extension, but seems not working. Any other idea? -- Best regards, Gergo mailto:[EMAIL PROTECTED] ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 - 1.6
Hi, You can find some info about differences between 1.4 and 1.6 here: http://svn.digium.com/view/asterisk/branches/1.6.0/UPGRADE.txt?view=markup Kind regards, Chris 2008/8/28 --[ UxBoD ]-- [EMAIL PROTECTED]: Hi, I would like to give 1.6 a try and was wondering about the configuration files. Can I just copy them across to a new install or are they completely different ? Is there a document which shows what I would need to change ? Best Regards, -- --[ UxBoD ]-- // PGP Key: curl -s http://www.splatnix.net/uxbod.asc | gpg --import // Fingerprint: F57A 0CBD DD19 79E9 1FCC A612 CB36 D89D 2C5A 3A84 // Keyserver: www.keyserver.net Key-ID: 0x2C5A3A84 // Phone: +44 845 869 2749 SIP Phone: [EMAIL PROTECTED] -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net 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 -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
