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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080129/289bf8e6/attachment.htm
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
Why not give the receptionist a two line phone? Register one line on server 1 and the other on server 2. Then the bounce back and forth goes away saving bandwidth. Lyle 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-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080128/b0ced81b/attachment.htm