RE: [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU
Hi Steve I have a similiar problem with noise. Asterisk SIP to SIP calls works without problems. During outbound and inbound PSTN calls, if there is only single call, the system works perfectly as well - voice is crystal clear. However, 10 - 60 seconds after a 2nd simultaneous call in the E1 starts, the voice becomes garbled and delay starts to increase to a point where the quality is too bad for the call to continue. Any idea ? Versions : Hardware : Wildcard TE405P asterisk-1.0.7 zaptel-1.0.7 libunicall-0.0.3pre3 Best Regards, Luis M. Kibe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Friday, June 10, 2005 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU Hi, We have now solved this problem. There was a bug in selecting codecs when chan_unicall generates DTMF or supervisory tones. If anyone else is having a similar problem with high CPU usage when running chan_unicall try stable version unicall-0.0.2b, or test version unicall-0.0.3pre3. They contain the fix. Regards, Steve Andres Maduro wrote: Hi, I have recently found a bug when using Steve Underwood chan_unicall with Asterisk 1.0.x (including 1.0.8RC) When you place a call from a SIP phone with dtmfmode=rfc2833 or dtmfmode=inband through MFCR2 via chan_unicall all goes well until you press a dtmf key. When you do this, the other end hears a garbage sound (not the dtmf tone) and cpu goes to 99.9% rendering almost unusable the PBX. If there are more than 2 calls, audio start to get choppy, more calls renders unusable the pbx. If you hangup the calling extension, almost all the time it returns to normality, if there is a moderate load on the * server, the only way of shutting down * is by killing -9 it. I have been working this with Steve and have reported this finding today. If you have any suggestion in which things could be tweaked in chan_sip.c, chan_zap.c or chan_unicall.c in order to see if this bug could be solved, I will be happy to test it. Any additional info you may require please let me know. Regards. AM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU
Additional information : I use brazilian E1 variant br Luis M. Kibe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Luis M. Kibe Sent: Tuesday, June 14, 2005 6:00 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU Hi Steve I have a similiar problem with noise. Asterisk SIP to SIP calls works without problems. During outbound and inbound PSTN calls, if there is only single call, the system works perfectly as well - voice is crystal clear. However, 10 - 60 seconds after a 2nd simultaneous call in the E1 starts, the voice becomes garbled and delay starts to increase to a point where the quality is too bad for the call to continue. Any idea ? Versions : Hardware : Wildcard TE405P asterisk-1.0.7 zaptel-1.0.7 libunicall-0.0.3pre3 Best Regards, Luis M. Kibe -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Steve Underwood Sent: Friday, June 10, 2005 10:36 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU Hi, We have now solved this problem. There was a bug in selecting codecs when chan_unicall generates DTMF or supervisory tones. If anyone else is having a similar problem with high CPU usage when running chan_unicall try stable version unicall-0.0.2b, or test version unicall-0.0.3pre3. They contain the fix. Regards, Steve Andres Maduro wrote: Hi, I have recently found a bug when using Steve Underwood chan_unicall with Asterisk 1.0.x (including 1.0.8RC) When you place a call from a SIP phone with dtmfmode=rfc2833 or dtmfmode=inband through MFCR2 via chan_unicall all goes well until you press a dtmf key. When you do this, the other end hears a garbage sound (not the dtmf tone) and cpu goes to 99.9% rendering almost unusable the PBX. If there are more than 2 calls, audio start to get choppy, more calls renders unusable the pbx. If you hangup the calling extension, almost all the time it returns to normality, if there is a moderate load on the * server, the only way of shutting down * is by killing -9 it. I have been working this with Steve and have reported this finding today. If you have any suggestion in which things could be tweaked in chan_sip.c, chan_zap.c or chan_unicall.c in order to see if this bug could be solved, I will be happy to test it. Any additional info you may require please let me know. Regards. AM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU
Hi, I have recently found a bug when using Steve Underwood chan_unicall with Asterisk 1.0.x (including 1.0.8RC) When you place a call from a SIP phone with dtmfmode=rfc2833 or dtmfmode=inband through MFCR2 via chan_unicall all goes well until you press a dtmf key. When you do this, the other end hears a garbage sound (not the dtmf tone) and cpu goes to 99.9% rendering almost unusable the PBX. If there are more than 2 calls, audio start to get choppy, more calls renders unusable the pbx. If you hangup the calling extension, almost all the time it returns to normality, if there is a moderate load on the * server, the only way of shutting down * is by killing -9 it. I have been working this with Steve and have reported this finding today. If you have any suggestion in which things could be tweaked in chan_sip.c, chan_zap.c or chan_unicall.c in order to see if this bug could be solved, I will be happy to test it. Any additional info you may require please let me know. Regards. AM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com 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] chan_unicall, dtmf bug in * 1.0.x - 99.9% CPU
Hi, We have now solved this problem. There was a bug in selecting codecs when chan_unicall generates DTMF or supervisory tones. If anyone else is having a similar problem with high CPU usage when running chan_unicall try stable version unicall-0.0.2b, or test version unicall-0.0.3pre3. They contain the fix. Regards, Steve Andres Maduro wrote: Hi, I have recently found a bug when using Steve Underwood chan_unicall with Asterisk 1.0.x (including 1.0.8RC) When you place a call from a SIP phone with dtmfmode=rfc2833 or dtmfmode=inband through MFCR2 via chan_unicall all goes well until you press a dtmf key. When you do this, the other end hears a garbage sound (not the dtmf tone) and cpu goes to 99.9% rendering almost unusable the PBX. If there are more than 2 calls, audio start to get choppy, more calls renders unusable the pbx. If you hangup the calling extension, almost all the time it returns to normality, if there is a moderate load on the * server, the only way of shutting down * is by killing -9 it. I have been working this with Steve and have reported this finding today. If you have any suggestion in which things could be tweaked in chan_sip.c, chan_zap.c or chan_unicall.c in order to see if this bug could be solved, I will be happy to test it. Any additional info you may require please let me know. Regards. AM. ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
