Re: [Asterisk-Users] Audio Drops out at Random - one way
Craig Waddington wrote: I found this: http://lists.digium.com/pipermail/asterisk-dev/2003-May/000764.html But it is old, and I am sure lots of changes have been made to the source, since then. Where and how do you set absolutetimeout=0, would this help? A test I want to perform is, we make a call, and say nothing for 20 seconds, and see if that's why the audio stream is being dropped. ??? What I am doing currently is running debug IAX2 when users make a call, to try pinpoint the issue, but I don't know what I am looking for in the output. Are you using Cisco Phones? If so, what firmware, that is the only common thing at my end. This install worked fine for months, the audio issue has just started occurring. The quality is perfect, except this loss of Audio for a few seconds. Is your problem purely outgoing? If you haven't set absolutetimeout in your dialplan then there will be no reason to have to reset it back to 0.. I tried the debug but didn't see much useful in there.. probably because I don't know what its telling me.. I am using Snom phones but I am almost sure its not the phone to Asterisk leg because I have an analog phone connected directly to the Asterisk box and it also drops calls.. Its almost certainly the IAX leg.. My problem appears to be purely on outgoing but we don't get a huge amount of incoming calls over the IAX and as far as I know we have never dropped an inbound call, but because of the small inbound call volume I wouldn't take it as fact.. Later.. ___ 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] Audio Drops out at Random - one way
I found this: http://lists.digium.com/pipermail/asterisk-dev/2003-May/000764.html But it is old, and I am sure lots of changes have been made to the source, since then. Where and how do you set absolutetimeout=0, would this help? A test I want to perform is, we make a call, and say nothing for 20 seconds, and see if that's why the audio stream is being dropped. ??? What I am doing currently is running debug IAX2 when users make a call, to try pinpoint the issue, but I don't know what I am looking for in the output. Are you using Cisco Phones? If so, what firmware, that is the only common thing at my end. This install worked fine for months, the audio issue has just started occurring. The quality is perfect, except this loss of Audio for a few seconds. Is your problem purely outgoing? -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut Sent: 29 November 2004 13:01 To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Audio Drops out at Random - one way Craig Waddington wrote: > Have a strange problem. > > 2 different asterisk servers, running different CVS. > > One behind Firewall, one not. > > Cisco 7940 phones. > > Over the past two weeks, users have had a problem with one way audio, > after about 2 minutes into a call, they can hear the other person, but > the other person cannot hear them, this happens for about 3-5 seconds, > then all is fine again. > > It doesn't happen on every call, about one in 5. > > Hardware is good, 2mb Connection, QOS enabled. > > If it was only one Asterisk server I would be ok, but it happens on > two completely different places. > > I cannot work out what is causing it, can anyone offer anyone offer a > solution or a method to track this down. > > Thanks. > >--- - > > > I have a similar problem with my IAX connection to my termination provider.. No one seems to be able to help and I have replaced or reinstalled just about every component in the chain except the internet itself and the termination provider.. Have updated Asterisk to 1.0.2, have added a switch to my network (was using a hub), have changes to a different firewall, have setup port mapping through the NAT, have tried different DSL routers and put in a high quality microfilter.. So the only things I think it can be are a) my termination provider (but they service many people and I am sure others would have brought it up if it was a problem), b) Asterisk itself or c) my DSL line or ISP.. Unfortunately these are all hard to check and the debug logging on Asterisk didn't help much when I tried looking at it.. I know this doesn't help much but if you come up with anything please let me know.. Its driving us crazy having calls drop on us especially when talking to customers.. Later.. ___ 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 ___ 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] Audio Drops out at Random - one way
Craig Waddington wrote: Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesn’t happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. I have a similar problem with my IAX connection to my termination provider.. No one seems to be able to help and I have replaced or reinstalled just about every component in the chain except the internet itself and the termination provider.. Have updated Asterisk to 1.0.2, have added a switch to my network (was using a hub), have changes to a different firewall, have setup port mapping through the NAT, have tried different DSL routers and put in a high quality microfilter.. So the only things I think it can be are a) my termination provider (but they service many people and I am sure others would have brought it up if it was a problem), b) Asterisk itself or c) my DSL line or ISP.. Unfortunately these are all hard to check and the debug logging on Asterisk didn't help much when I tried looking at it.. I know this doesn't help much but if you come up with anything please let me know.. Its driving us crazy having calls drop on us especially when talking to customers.. Later.. ___ 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] Audio Drops out at Random - one way
Sorry forgot to mention this is with IAX2 only, SIP works fine. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Craig Waddington Sent: 29 November 2004 10:46 To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Audio Drops out at Random - one way Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesn’t happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. ___ 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
[Asterisk-Users] Audio Drops out at Random - one way
Have a strange problem. 2 different asterisk servers, running different CVS. One behind Firewall, one not. Cisco 7940 phones. Over the past two weeks, users have had a problem with one way audio, after about 2 minutes into a call, they can hear the other person, but the other person cannot hear them, this happens for about 3-5 seconds, then all is fine again. It doesn’t happen on every call, about one in 5. Hardware is good, 2mb Connection, QOS enabled. If it was only one Asterisk server I would be ok, but it happens on two completely different places. I cannot work out what is causing it, can anyone offer anyone offer a solution or a method to track this down. Thanks. ___ 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
