We had very similar problems to this which drove us insane for ages. Basically we use VoIP trunks (SIP) for all our inbound + outbound calls. Call quality was good however we would get random problems where people could not hear us or us hear them for about 5-10 seconds at a time.
After weeks of trying to get to the bottom of the problem it appeared our VoIP trunk provider (sentiro/sip2go) had ropey connectivity at best. We have since changed provider and now experience no call problems whatsoever (after running extensive tests to the sip host such as mtr etc.) Jamie > -----Original Message----- > From: [EMAIL PROTECTED] > [mailto:[EMAIL PROTECTED] On Behalf Of Matt > Sent: 05 November 2006 13:30 > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Audio goes one way during the > call for a fewseconds. Is it RTP, NAT, dyndns, or what it is? > > Sounds like a bad Internet connection messing with the IAX > jitterbuffer. Try running ping plotter from your location to > your host, and see if it goes 'red'/down. > > On 11/3/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote: > > Hi everybody, > > > > I finally want to get rid of 1-way audio problem. Please > help me here. > > > > I have 3 scenarios. > > > > 1. Audio is always one way. Caller who dialed can't listen > the called > > party but called party can listen him. In this scenatio > Asterisk is on > > dynamic IP with dyndns FQDN. sip.conf has externip = abc.dyndns.org > > and localnet = xxx.xxx.xxx.xxx entry. Trunk and extensions are SIP. > > Where is the voice getting lost from the called party? NAT > is there but Asterisk is in DMZ. > > > > 2. Conversation is going fine when all of a sudden you realize that > > other parth has started saying 'hello, hello' because they > can't hear > > you. But you are hearing them loud and clear. Now you are > on static IP with dyndns FQDN. > > externip and localnet settings in sip.conf (do we need them > for static IP?). > > After about 15-20 seconds, again 2-way converstaion is > established again. > > IAX trunk, SIP extension, no NAT. > > > > 3. Conversation goes one way for 15-20 sec during the most > important > > part of the conversation (Murphy's Law). You are on a > static IP with > > no dyndns enrty. Trunk is ZAP on PRI, extensions SIP. NAT > present but > > router properly configures for port forwarding. externip > and localnet > > settings present in sip.conf > > > > Is think may be due to some reason RTP stream gets lost, > routed to wrong IP. > > But why would this happen during a call and how to stop it > from happening. > > Or is there some other reason behind this? Does dyndns > setting have to > > do anything with this problem? How can I overcome this problem once > > and forever. > > > > -- > > Zeeshan A Zakaria > > _______________________________________________ > > --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
