Thanks Paul and Lyle for the suggestions. I would like to keep the phones configuration to one line for now, and see if I can solve the problem rather then just work around it.
I have changed he notransfer option, will see what happens over the next few days. Thanks again for the suggestions, any further input is very much welcome. Many Thanks, Daniel -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales Sent: Tuesday, 29 January 2008 11:31 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] IAX Calls - One Way Audio Does turning off the notransfer help? I would imagine that dropping the second server out of the equation might be useful, and save some bandwidth. PaulH On Tue, 2008-01-29 at 10:38 +1100, Daniel Cole wrote: > Hello List, > > I am currently having a bit of a strange issue with a pair of asterisk > servers that we recently set up. > > For a bit of background, this particular business has two sites in two > different towns, about 10 minutes apart. They have 3 analogue PSTN > lines connected to the asterisk servers at each location, via a > Sangoma A200 (with HEC). They are trying to have just the one > receptionist for the whole organization, answering calls that come in > for both locations. > > We have a problem where some calls (seemingly randomly) appear to get > one way audio. This only happens for inbound calls off the PSTN, if > they follow this pattern (which is a fair number of calls): > > Call comes in from PSTN to site A, gets put into a queue to be > answered by receptionist as site B. Receptionist answers the call, and > then puts the call on hold to perform an attended transfer to an > extension at site A. (The call from the receptionist to the extension > is OK). When the receptionist hits the 'transfer' button to actually > transfer the call, the original caller cannot hear anything. The > internal extension can hear the caller OK. > > This problem does not occur on every call. Since the issue has risen > its head, I have enabled core, sip and iax debugging, but I am of yet > unable to get the issue to occur on its own, to have a good look at > the log files. > > FYI, I have disabled the asterisk Dial Commands in FreePBX, to solve > another issue (where call audio bounces between the servers for a call > that is transferred between sites and back again). > > Both servers are asterisk version 1.2.23, freepbx version 2.3.1.0. > > I have posted the contents of the iax.conf file below (which is > identical on both servers). If there is any further information I can > provide, please let me know and I can get this information. > > > > [general] > > disallow=all > allow=g729 > mailboxdetail=yes > > jitterbuffer=no > ;maxjitterbuffer=500 > ;jittershrinkrate=1 > bandwidth=low > tos=lowdelay > trunk=yes > notransfer=yes > > #include iax_general_custom.conf > #include iax_registrations_custom.conf #include iax_registrations.conf > #include iax_custom.conf #include iax_additional.conf > > > > > Any suggestions are very welcome. > > Regards, > > Daniel > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
