I have set up a speed dial on Linksys /Cisco SPA962 phones, for those of you who are familiar with these devices, I have set the following for one of the lines:
The goal is to transfer a call to extension 610 by pressing the line button. Up to Asterisk 1.6.2.13 this worked well. After upgrading to 1.6.2.15 it still works for calls initiated from the SPA962, but not with calls answered from this phone - the incoming call is hung up. Here is the sip debugging output from when it fails (names and addresses changed to protect the innocent):
[code]<— SIP read from UDP:192.168.71.117:5060 —>
REFER sip:81@192.168.71.8 SIP/2.0
Via: SIP/2.0/UDP 192.168.71.117:5060;branch=z9hG4bK-dfb04ab7
From: sip:myext#@192.168.71.117;tag=6665db0b4bc0b74bi0
To: “Ian’s Nexus One” sip:81@my.net.com;tag=as5c9200ea
Referred-By: “Le Jardin 17” sip:myext#@my.net.com
Call-ID: 629e3fe673716bf24d4f0898482dba6a@my.net.com
CSeq: 101 REFER
Max-Forwards: 70
Contact: “Le Jardin 17” sip:myext#@192.168.71.117:5060
Refer-To: sip:610@my.net.com
User-Agent: Linksys/SPA962-6.1.5(a)
Content-Length: 0
<------------->
— (12 headers 0 lines) —
Call 629e3fe673716bf24d4f0898482dba6a@my.net.com got a SIP call transfer from callee: (REFER)!
SIP transfer to extension 610@phones by myext#@my.net.com
<— Transmitting (no NAT) to 192.168.71.117:5060 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.71.117:5060;branch=z9hG4bK-dfb04ab7;received=192.168.71.117
From: sip:myext#@192.168.71.117;tag=6665db0b4bc0b74bi0
To: “Ian’s Nexus One” sip:81@my.net.com;tag=as5c9200ea
Call-ID: 629e3fe673716bf24d4f0898482dba6a@my.net.com
CSeq: 101 REFER
Server: Asterisk PBX 1.6.2.15
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:81@192.168.71.8
Content-Length: 0
<------------>
set_destination: Parsing sip:myext#@192.168.71.117:5060 for address/port to send to
set_destination: set destination to 192.168.71.117, port 5060
Reliably Transmitting (no NAT) to 192.168.71.117:5060:
NOTIFY sip:myext#@192.168.71.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK09bae71e;rport
Max-Forwards: 70
From: “Ian’s Nexus One” sip:81@my.net.com;tag=as5c9200ea
To: sip:myext#@192.168.71.117:5060;tag=6665db0b4bc0b74bi0
Contact: sip:81@192.168.71.8
Call-ID: 629e3fe673716bf24d4f0898482dba6a@my.net.com
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 1.6.2.15
Remote-Party-ID: “Ian’s Nexus One” sip:81@my.net.com;privacy=off;screen=no
Event: refer;id=101
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 21
SIP/2.0 183 Ringing
set_destination: Parsing sip:myext#@192.168.71.117:5060 for address/port to send to
set_destination: set destination to 192.168.71.117, port 5060
Reliably Transmitting (no NAT) to 192.168.71.117:5060:
NOTIFY sip:myext#@192.168.71.117:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.71.8:5060;branch=z9hG4bK6457b520;rport
Max-Forwards: 70
From: “Ian’s Nexus One” sip:81@my.net.com;tag=as5c9200ea
To: sip:myext#@192.168.71.117:5060;tag=6665db0b4bc0b74bi0
Contact: sip:81@192.168.71.8
Call-ID: 629e3fe673716bf24d4f0898482dba6a@my.net.com
CSeq: 104 NOTIFY
User-Agent: Asterisk PBX 1.6.2.15
Remote-Party-ID: “Ian’s Nexus One” sip:81@my.net.com;privacy=off;screen=no
Event: refer;id=101
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Length: 16
SIP/2.0 200 Ok
Scheduling destruction of SIP dialog ‘629e3fe673716bf24d4f0898482dba6a@my.net.com’ in 6400 ms (Method: REFER)
== Spawn extension (phones, 610, 1) exited non-zero on ‘SIP/61blahblah1iUB-0000002d’ in macro ‘voicemail’
== Spawn extension [/code]
Any suggestions?
Ian