[Asterisk-Users] One Way Audio....in the middle of a call
We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --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
Re: [Asterisk-Users] One Way Audio....in the middle of a call
Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; youwill see if they change port or something like that after a while. Cheers, Frederic - Original Message - From: Geoff Manning To: Asterisk Users Mailing List - Non-Commercial Discussion Sent: Tuesday, April 25, 2006 17:37 Subject: [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ___ --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
Re: [Asterisk-Users] One Way Audio....in the middle of a call
I experienced this today. Doing a 'show channels' in Asterisk showed a Zap line perpetually ringing the sip phone even though the sip phone was reset a few times. Doing a 'soft hangup' on the stuck Zap and the Sip allowed 2-way audio to resume. Phil Frederic Jean wrote: Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; you will see if they change port or something like that after a while. Cheers, Frederic - Original Message - *From:* Geoff Manning mailto:[EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Tuesday, April 25, 2006 17:37 *Subject:* [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --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 ___ --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 ___ --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
Re: [Asterisk-Users] One Way Audio....in the middle of a call
Is there an easy fix?- Dan L.On 4/25/06, Philip Edelbrock [EMAIL PROTECTED] wrote: I experienced this today.Doing a 'show channels' in Asterisk showed aZap line perpetually ringing the sip phone even though the sip phone wasreset a few times.Doing a 'soft hangup' on the stuck Zap and the Sip allowed 2-way audio to resume.PhilFrederic Jean wrote: Hi Geoff, You might want to try tcdump, specifying the source and destination IP (to minimize the info) and see where are the RTP packets going ; you will see if they change port or something like that after a while. Cheers, Frederic - Original Message - *From:* Geoff Manning mailto: [EMAIL PROTECTED] *To:* Asterisk Users Mailing List - Non-Commercial Discussion mailto:asterisk-users@lists.digium.com *Sent:* Tuesday, April 25, 2006 17:37 *Subject:* [Asterisk-Users] One Way Audioin the middle of a call We had a user report that they were on a SIP --- PSTN call for about 4.5 minutes before the call went to on-way audio. The user called the person back and they reported being able to hear my user, but my user couldn't hear them. The audio condition persisted for about 15 seconds before the user hung up. Where do I start to troubleshoot one way audio that occurs during a call? ___ --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 ___ --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___--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 ___ --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
