We are experiencing an issue on transfers where no progress tones are heard by the caller. If A and B are on a call and A transfers the call to C, then B doesn't hear any ringing whilst waiting for C to answer. Note that A hears ringing when calling C until the point the xfer button is pressed:
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. 1593 hears progress tones. 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:
59ba10300b9b8cb5684eba2368c90a77@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:
59ba10300b9b8cb5684eba2368c90a77@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
c90a77@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
I've tried calls in different directions in case it is to do with the particular phone firmware but the direction is irrelevant.
Note that progress tones (ringing) is heard when calling from one extension to another and when calling externally.
Any suggestions appreciated or if you require further information please ask.
