Asterisk 1.4 We are experiencing an issue on transfers where no progress tones are heard by the caller: 1. Call from 1593 (SPA525G 0026998D2FFFF) to 1595 (SPA922 000B820AFFFFF). 1595 answers 2. From 1593 initiate transfer to 1597 (SPA508G 1CDF0F4AFFFF). 1595 hears MoH. 3. 1597 starts ringing and 1593 presses transfer again. MoH stops but 1595 hears no ringing When xfer is pressed and the extension is dialled: U 203.89.001.001:5060 -> 121.98.001.001:1034 INVITE sip:1CDF0F4AFFFF@192.168.1.72:5060 SIP/2.0..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e;rport..From: "C Allerid" <sip:1593@203.89.001.001>;tag=as72616c50..To: <sip:1CDF0F4AFFFF@192.168.1.72:5060>..Contact: <sip:1593@203.89.001.001>..Call-ID: 59ba10300b9b8cb5684eba2368c90...@203.89.001.001..cseq: 102 INVITE..User-Agent: Asterisk PBX..Max-Forwards: 70..Date: Tue, 05 Jun 2012 08:05:02 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO..Supported: replaces..Content-Type: application/sdp..Content-Length: 262....v=0..o=root 3031 3031 IN IP4 203.89.001.001..s=session..c=IN IP4 203.89.001.001..t=0 0..m=audio 13728 RTP/AVP 0 3 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:3 GSM/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1010-16..a=ptime:20..a=sendrecv.. U 121.98.001.001:1034 -> 203.89.001.001:5060 SIP/2.0 100 Trying..To: <sip:1CDF0F4AFFFF@192.168.1.72:5060>..From: "C Allerid" <sip:1593@203.89.001.001>;tag=as72616c50..Call-ID: 59ba10300b9b8cb5684eba2368c90...@203.89.001.001..cseq: 102 INVITE..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Server: Cisco/SPA508G-7.4.9c..Content-Length: 0.... U 121.98.001.001:1034 -> 203.89.001.001:5060 SIP/2.0 180 Ringing..To: <sip:1CDF0F4AFFFF@192.168.1.72:5060>;tag=53e23c5265d60f06i0..From: "C Allerid" <sip:1593@203.89.001.001>;tag=as72616c50..Call-ID:59ba10300b9b8cb5684eba2368 c90...@203.89.001.001..cseq: 102 INVITE..Via: SIP/2.0/UDP 203.89.001.001:5060;branch=z9hG4bK5286810e..Contact: "$USER" <sip:1CDF0F4AFFFF@192.168.1.72:5060>..Server: Cisco/SPA508G-7.4.9c..Content-Length: 0.... After transfer is pressed the second time there is no further SIP messages with
Asterisk CLI -- Executing [s@macro-dial:12] Dial("SIP/000B820AFFFFF-00002d0a", "SIP/000E08D6FFFF&SIP/1CDF0F4AFFFF&SIP/000E08D6FFFF1|20|tTwWr") in new stack -- Called 1CDF0F4AFFFF -- SIP/1CDF0F4AFFFF-00002d0b is ringing -- Stopped music on hold on SIP/0026998D2FFFF-00002d08 Updated sip.conf progressinband=yes This didn't make any difference I've tried calls in different directions in case it is to do with the particular phone firmware but the direction is irrelevant. Any suggestions appreciated or if you require further information please ask. -- _____________________________________________________________________ -- 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