Hi,
We are using asterisknow rpms, asterisk 1.8.15 on centos 5.8
For the last 3 versions we have been unable to perform handset-initiated transfers on outbound calls
We can also replicate this issue on other asterisk 1.8 systems running similar release versions.
1) The extension will make an outgoing call & establish communications
2) The extension then presses the transfer key and it normally sets up an attended transfer to another extension
3) After transferring the call, the external callee gets about 1 second of audio with the extension that they were being transferred to before the call drops.
We have tried this with various softphones, and desk phones such as snom & polycom. All seem to be using the REFER method to transfer.
When the asterisk system receives inbound calls, they always transfer correctly without any problems at all. The issue only occurs when an extension initiates a call then tries to transfer to another extensions.
Here's an example SIP trace, using a softphone on this occasion:
2012-06-29 16:20:22: 0000:000:
m_req->m_async...
<<<
0000:000: ACK sip:201@10.10.10.100:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.41:5060;branch=z9hG4bK-0AAEAE814AE13D6C2CD672A0AB1A350
From: <sip:214@10.10.10.100>;tag=aaeac5f
To: <sip:201@10.10.10.100>;tag=as154c4e2d
Call-ID: 024AB3E0-0AB1A34E18BE67844AE13D6C@10.10.10.41
CSeq: 2 ACK
Contact: "214" <sip:214@10.10.10.41:5060>
Max-Forwards: 70
User-Agent: xtelsio CTI Client Pro 2.2.074
Content-Length: 0
0000:000: RTPChannel::ConnectToRemoteParty(8220,10.10.10.100)
0000:000: call->m_rtpChannel.StartRecvThread();
0000:000: m_sipAudioCodecIdx=8
2012-06-29 16:20:26: 0004:648:
m_req->m_async...
<<<
0000:000: REFER sip:201@10.10.10.100:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.41:5060;branch=z9hG4bK-0AAEAE814AE13D6C2CD672A0AB1C1E8
From: <sip:214@10.10.10.100>;tag=aaeac5f
To: <sip:201@10.10.10.100>;tag=as154c4e2d
Call-ID: 024AB3E0-0AB1A34E18BE67844AE13D6C@10.10.10.41
CSeq: 3 REFER
Contact: "214" <sip:214@10.10.10.41:5060>
Max-Forwards: 70
User-Agent: xtelsio CTI Client Pro 2.2.074
Refer-To: sip:07720705116@10.10.10.100?Replaces=57ba7af21f0d56e00c38b1987bba4ebc%4010.10.10.100:5060%3Bto-tag%3Das729b8c73%3Bfrom-tag%3Daaeac5e
Referred-By: "214" <sip:214@10.10.10.100>
Content-Length: 0
2012-06-29 16:20:26: 0000:000:
>>>
0000:000: SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.10.10.41:5060;branch=z9hG4bK-0AAEAE814AE13D6C2CD672A0AB1C1E8;received=10.10.10.41;rport=5060
From: <sip:214@10.10.10.100>;tag=aaeac5f
To: <sip:201@10.10.10.100>;tag=as154c4e2d
Call-ID: 024AB3E0-0AB1A34E18BE67844AE13D6C@10.10.10.41
CSeq: 3 REFER
Server: Onecomms_PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:201@10.10.10.100:5060>
Content-Length: 0
0000:000:
2012-06-29 16:20:26: 0000:000:
>>>
0000:000: NOTIFY sip:214@10.10.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK64777b52;rport
Max-Forwards: 70
From: <sip:201@10.10.10.100>;tag=as154c4e2d
To: <sip:214@10.10.10.100>;tag=aaeac5f
Contact: <sip:201@10.10.10.100:5060>
Call-ID: 024AB3E0-0AB1A34E18BE67844AE13D6C@10.10.10.41
CSeq: 102 NOTIFY
User-Agent: Onecomms_PBX
Event: refer;id=3
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 16
SIP/2.0 200 OK
0000:000:
2012-06-29 16:20:26: 0000:000:
sync...
<<<
0000:000: SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK64777b52;rport
From: <sip:201@10.10.10.100>;tag=as154c4e2d
To: <sip:214@10.10.10.100>;tag=aaeac5f
Contact: <sip:201@10.10.10.100:5060>
Call-ID: 024AB3E0-0AB1A34E18BE67844AE13D6C@10.10.10.41
CSeq: 102 NOTIFY
Allow: REGISTER, SUBSCRIBE, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY
Content-Length: 0
2012-06-29 16:20:26: 0000:000:
>>>
0000:000: BYE sip:214@10.10.10.41:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK30b42efe;rport
Max-Forwards: 70
From: <sip:07720705116@10.10.10.100>;tag=as729b8c73
To: <sip:214@10.10.10.100>;tag=aaeac5e
Call-ID: 57ba7af21f0d56e00c38b1987bba4ebc@10.10.10.100:5060
CSeq: 103 BYE
User-Agent: Onecomms_PBX
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
0000:000:
2012-06-29 16:20:26: 0000:000:
sync...
<<<
0000:000: SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.10.10.100:5060;branch=z9hG4bK30b42efe;rport
From: <sip:07720705116@10.10.10.100>;tag=as729b8c73
To: <sip:214@10.10.10.100>;tag=aaeac5e
Contact: "214" <sip:214@10.10.10.41:5060>
Call-ID: 57ba7af21f0d56e00c38b1987bba4ebc@10.10.10.100:5060
CSeq: 103 BYE
Allow: REGISTER, SUBSCRIBE, INVITE, ACK, CANCEL, BYE, REFER, NOTIFY
Content-Length: 0
2012-06-29 16:20:26: 0000:000:
m_req->m_async...
<<<
0000:000: BYE sip:201@10.10.10.100:5060 SIP/2.0
Via: SIP/2.0/UDP 10.10.10.41:5060;branch=z9hG4bK-0AAEAE814AE13D6C2CD672A0AB1C1E9
From: <sip:214@10.10.10.100>;tag=aaeac5f
To: <sip:201@10.10.10.100>;tag=as154c4e2d
Call-ID: 024AB3E0-0AB1A34E18BE67844AE13D6C@10.10.10.41
CSeq: 4 BYE
Contact: "214" <sip:214@10.10.10.41:5060>
Max-Forwards: 70
User-Agent: xtelsio CTI Client Pro 2.2.074
Content-Length: 0
