I have an Asterisk server on our LAN that serves our office VOIP phones with a SIP trunk to voipfone (UK ITSP). All LAN calls are ulaw/alaw
We use attended transfer extensively. If our trunk is ulaw/alaw they work fine. If the trunk is ilbc we have problems 1- incoming PSTN call routed via voipfone SIP down the trunk to our server 2- our phones ring ok, caller can be answered (e.g. by A) 3- A requests attended transfer to another phone (B) on the LAN- incoming caller put on hold, A can talk to B, B can talk to A 4- A hangs up, B is connected to caller. B can hear caller, but caller cannot hear B. Console output: Asked to transmit frame type 64, while native formats is 0x400 (ilbc)(1024) read/write = 0x40 (slin)(64)/0x400 (ilbc)(1024) Running Asterisk 1.6.2.9 on Ubuntu Karmic- self compiled (do not seem to be able to compile deb source package with ilbc, and deb package does not have ilbc) Any idea what may be happening? John -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
