Re: [Asterisk-Users] Codec problems (SIP)
Hi again, I've finally got Voicetronix OpenLine4 working so am a happy man ... thanks again to all those who helped! Just a few outstanding questions of curiosity : 1. I have finally got my setup to work by allowing ONLY g711alaw and nothing else. Why should enabling a few extra codecs cause problems? Surely if two phones are able to work at g711alaw, and either side had a compatibility problem with anything else (i.e. g729a at one end but not at the other) - they would automatically negotiate to use g711alaw anyway? Is the system/phones not smart enough to do this and I have to explicitly specify what everything should use? Secondly, also regarding codecs - I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path? How do you got the g729 codec? * does not include it. You must to pay for that. ... okay, fine. But where can I buy it? And is there something specific I have to buy, or does any old thing work with asterisk? Or...? Thanks again! Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec problems (SIP)
Terence Parker wrote: Hi again, I've finally got Voicetronix OpenLine4 working so am a happy man ... thanks again to all those who helped! Just a few outstanding questions of curiosity : 1. I have finally got my setup to work by allowing ONLY g711alaw and nothing else. Why should enabling a few extra codecs cause problems? Surely if two phones are able to work at g711alaw, and either side had a compatibility problem with anything else (i.e. g729a at one end but not at the other) - they would automatically negotiate to use g711alaw anyway? Is the system/phones not smart enough to do this and I have to explicitly specify what everything should use? My little experience on * tell me that some phones and/or channels do not negotiate very well the codec selection. Secondly, also regarding codecs - I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path? How do you got the g729 codec? * does not include it. You must to pay for that. ... okay, fine. But where can I buy it? And is there something specific I have to buy, or does any old thing work with asterisk? Or...? * is OS, thus I can not include any think that needs license. See Digium if you want to buy g729 codecs. Jorge Thanks again! Terence ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Codec problems (SIP)
Hi again, Thanks for your help. Unfortunately that did not seem to solve the problem. After a bit of fiddling around, this is what i've managed to achieve with my asterisk setup so far. 1. With allow=all in sip.conf, nothing seems to work - not even voicemail. The following is sample output: Executing Ringing(SIP/TerenceParker-1af0, ) in new stack -- Executing Wait(SIP/TerenceParker-1af0, 2) in new stack -- Executing VoiceMailMain(SIP/TerenceParker-1af0, ) in new stack -- Playing 'vm-login' (language 'en') WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username == Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-1af0' - Why should this happen? Surely with everything enabled, any coded should work! 2. With disallow=all ; allow=alaw ; allow=ulaw ; allow=g729 ; allow=gsm (and i've also tried without some of those and various combinations): Executing Ringing(SIP/TerenceParker-af02, ) in new stack -- Executing Wait(SIP/TerenceParker-af02, 2) in new stack -- Executing VoiceMailMain(SIP/TerenceParker-af02, ) in new stack -- Playing 'vm-login' (language 'en') NOTICE[278546]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from GSM to G729A WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 4, while native formats is 256 (read/write = 4/2) WARNING[278546]: File file.c, Line 521 (ast_readaudio_callback): Failed to write frame NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A WARNING[278546]: File file.c, Line 170 (ast_stopstream): Unable to restore format back to 4 WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username == Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-af02' - I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path? 3. With the default settings (i.e. no allow OR disallow clause) normal IP to IP calls work fine. Calls to voicemail also works fine with no problems. However, PSTN calls through my Voicetronix card or calls routed through FWD fail to work. This is what happens when I dial out with my voicetronix card: Executing Dial(SIP/TerenceParker-22f3, vpb/1-1/18501) in new stack Read_channel ## vpb/1-1: Setting record mode, bridge = 0 -- 1-1 requested, got: [vpb/1-1] -- Calling 1-1/18501 on vpb/1-1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 -- VPB Calling 1-1/18501 [t=0] on vpb/1-1 returned 0 -- Called 1-1/18501 WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame -- vpb/1-1 is ringing WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel ## vpb/1-1: Setting record mode, bridge = 0 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 -- Event [12=>[00] Loop Drop ] on vpb/1-1 -- vpb/1-1 handle_owned got event: [12=>0] -- handle_owned: putting frame: [-1=>0], bridge=(nil) WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 -- Event [102=>[00] Dial End ] on vpb/1-1 -- vpb/1-1 handle_owned got event: [102=>0] -- handle_owned: putting frame: [4=>4], bridge=(nil) -- vpb/1-1 answered SIP/TerenceParker-22f3 -- hangup on vpb (vpb/1-1) Read_channel vpb/1-1 (state=5), res=0, bridge=1 Read_channel vpb/1-1 (state=6), res=-1, bridge=1 Read_channel vpb/1-1 terminating, stopreads=1, owner=yes -- Hungup on vpb/1-1 complete == Spawn extension (sip, 918501, 1) exited non-zero on 'SIP/TerenceParker-22f3' - again, it complains about codecs. So, at the moment, I am utterly confused! Any help would be gratefully appreciated. Terence On 13 Jan 04, at 1:39 AM, Jorge Mendoza wrote: Try in sip.conf: disallow=all allow=alaw allow=ulaw allow=gsm (in that order) I never tried with FWD Jorge
Re: [Asterisk-Users] Codec problems (SIP)
Terence Parker wrote: Hi again, Thanks for your help. Unfortunately that did not seem to solve the problem. After a bit of fiddling around, this is what i've managed to achieve with my asterisk setup so far. 1. With allow=all in sip.conf, nothing seems to work - not even voicemail. The following is sample output: Executing Ringing(SIP/TerenceParker-1af0, ) in new stack -- Executing Wait(SIP/TerenceParker-1af0, 2) in new stack -- Executing VoiceMailMain(SIP/TerenceParker-1af0, ) in new stack -- Playing 'vm-login' (language 'en') WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username == Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-1af0' - Why should this happen? Surely with everything enabled, any coded should work! This log is not relate to codec problem. 2. With disallow=all ; allow=alaw ; allow=ulaw ; allow=g729 ; allow=gsm (and i've also tried without some of those and various combinations): Executing Ringing(SIP/TerenceParker-af02, ) in new stack -- Executing Wait(SIP/TerenceParker-af02, 2) in new stack -- Executing VoiceMailMain(SIP/TerenceParker-af02, ) in new stack -- Playing 'vm-login' (language 'en') NOTICE[278546]: File channel.c, Line 1478 (ast_set_read_format): Unable to find a path from G729A to ULAW NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from GSM to G729A WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 4, while native formats is 256 (read/write = 4/2) WARNING[278546]: File file.c, Line 521 (ast_readaudio_callback): Failed to write frame NOTICE[278546]: File channel.c, Line 1448 (ast_set_write_format): Unable to find a path from ULAW to G729A WARNING[278546]: File file.c, Line 170 (ast_stopstream): Unable to restore format back to 4 WARNING[278546]: File app_voicemail.c, Line 2707 (vm_execmain): Couldn't read username == Spawn extension (sip, 86, 3) exited non-zero on 'SIP/TerenceParker-af02' - I don't understand this as, surely, I have already enabled g729a and ulaw ... how can it complain that it can't transmit in that format, or that it can't find a path? How do you got the g729 codec? * does not include it. You must to pay for that. 3. With the default settings (i.e. no allow OR disallow clause) normal IP to IP calls work fine. Calls to voicemail also works fine with no problems. However, PSTN calls through my Voicetronix card or calls routed through FWD fail to work. This is what happens when I dial out with my voicetronix card: Executing Dial(SIP/TerenceParker-22f3, vpb/1-1/18501) in new stack Read_channel ## vpb/1-1: Setting record mode, bridge = 0 -- 1-1 requested, got: [vpb/1-1] -- Calling 1-1/18501 on vpb/1-1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel vpb/1-1 (state=0), res=0, bridge=1 -- VPB Calling 1-1/18501 [t=0] on vpb/1-1 returned 0 -- Called 1-1/18501 WARNING[278546]: File chan_sip.c, Line 1182 (sip_write): Asked to transmit frame type 8, while native formats is 4 (read/write = 4/4) WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame -- vpb/1-1 is ringing WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=0), res=0, bridge=1 Read_channel ## vpb/1-1: Setting record mode, bridge = 0 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 -- Event [12=[00] Loop Drop ] on vpb/1-1 -- vpb/1-1 handle_owned got event: [12=0] -- handle_owned: putting frame: [-1=0], bridge=(nil) WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 WARNING[278546]: File app_dial.c, Line 279 (wait_for_answer): Unable to forward frame Read_channel vpb/1-1 (state=5), res=0, bridge=1 -- Event [102=[00] Dial End ] on vpb/1-1 -- vpb/1-1 handle_owned got event: [102=0] -- handle_owned: putting frame: [4=4], bridge=(nil) -- vpb/1-1 answered SIP/TerenceParker-22f3 -- hangup on vpb (vpb/1-1) Read_channel vpb/1-1 (state=5), res=0, bridge=1 Read_channel vpb/1-1 (state=6), res=-1, bridge=1 Read_channel vpb/1-1 terminating, stopreads=1, owner=yes -- Hungup on vpb/1-1 complete == Spawn extension (sip, 918501, 1) exited non-zero on 'SIP/TerenceParker-22f3' - again, it complains about codecs. So, at the moment, I am utterly confused! Any help would be gratefully appreciated. Verify if you sip phone has codec alaw as preferred
Re: [Asterisk-Users] Codec problems (SIP)
Terence Parker wrote: I am trying to get my Voicetronix OpenLine4 card working in FXO mode in a PBX setup - so far I can only get it working as an IVR. I have managed to get my card to at least not crash now, and Asterisk does recognise it's existence... but I seem to be having codec problems. The same problems exist when testing bridging of calls through FWD too. I tried setting 'allow=all' in sip.conf - this eliminated the above problems, but resulted in another problem whereby any call between any phone would simply not work - 'dead air'. I get similar problems with FWD. Try in sip.conf: disallow=all allow=alaw allow=ulaw allow=gsm (in that order) I never tried with FWD Jorge ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
