Hullo :)
I'm using Debian's Asterisk 1.0.7 bristuffed (though I'm only using
CAPI for
ISDN, and not HFC-S cards) and trying to transfer an incoming SIP call from
sipgate.co.uk to any other extension.
My phones are AT-320s (PA168S 1.43 firmware) whose documentation says to blind
transfer, simply dial the number you want to transfer to, and press
'FWD'...
This is what happens when I start the sip debug after the initial call
setup...
01618313800 is the callerID of the person making the call, 1301 is the
internal SIP extension logged in as Agent 1600 at 10.0.0.82.
10.0.0.242 and 194.24.251.3 are the same machine, just two IPs on the same
eth0.
All I'm doing is answering the SIP phone, tapping 500 and pressing FWD to
transfer the incoming caller to the screaming monkeys gsm. If I dial 500 from
the phone directly, I immediately hear the monkeys, so assumed that a
transfer should be possible. e.g. in [from-ip] I have:
exten => 500,1,Playback(tt-monkeys)
and the sip.conf section is...
[1301]
type=friend
username=1301
secret=1301
host=dynamic
context=from-ip
nat=no
canreinvite=no
In extensions.conf's [internal] context (used by AgentCallbackLogin) I have
exten => _13XX,1,Dial(SIP/${EXTEN},20,t)
so that the agent has the ability to transfer calls (I also tried 'Tt'
for
completeness)
-- Executing SetCIDName("SIP/217.10.79.218-40b8edd8",
"CCUK") in new stack
-- Executing Queue("SIP/217.10.79.218-40b8edd8",
"ccuk|r") in new stack
-- outgoing agentcall, to agent '1600', on
'Local/1301@internal-17aa,1'
-- Called Agent/1600
-- Executing Dial("Local/1301@internal-17aa,2",
"SIP/1301|20|t") in new
stack
-- Called 1301
-- SIP/1301-9ebb is ringing
-- Agent/1600 is ringing
-- SIP/1301-9ebb answered Local/1301@internal-17aa,2
-- Agent/1600 answered SIP/217.10.79.218-40b8edd8
qax*CLI> sip debug
SIP Debugging Enabled
qax*CLI>
Sip read:
REFER sip:01618313800@194.24.251.3 SIP/2.0
Via: SIP/2.0/UDP 10.0.0.82:5060;branch=z9hG4bK95BVFY7yDu90YX0A
Max-Forwards: 7
User-Agent: PA168S
From: <sip:1301@10.0.0.82:5060>;tag=g5VVthPSslPbjLib
To: "CCUK" <sip:01618313800@194.24.251.3>;tag=as0eb5392e
Call-ID: 27584e3a339e535209ea89102043184e@194.24.251.3
Contact: <sip:1301@10.0.0.82:5060>
CSeq: 1 REFER
Refer-To: "500" <sip:500@10.0.0.242>
Referred-By: <sip:1301@10.0.0.242>
Content-Length: 0
12 headers, 0 lines
Looking for 500 in from-ip
Looking for 1301 in from-ip
Transmitting (no NAT):
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 10.0.0.82:5060;branch=z9hG4bK95BVFY7yDu90YX0A
From: <sip:1301@10.0.0.82:5060>;tag=g5VVthPSslPbjLib
To: "CCUK" <sip:01618313800@194.24.251.3>;tag=as0eb5392e
Call-ID: 27584e3a339e535209ea89102043184e@194.24.251.3
CSeq: 1 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:01618313800@194.24.251.3>
Content-Length: 0
to 10.0.0.82:5060
set_destination: Parsing <sip:1301@10.0.0.82:5060> for address/port to
send to
set_destination: set destination to 10.0.0.82, port 5060
Reliably Transmitting:
NOTIFY sip:1301@10.0.0.82:5060 SIP/2.0
Via: SIP/2.0/UDP 194.24.251.3:5060;branch=z9hG4bK08ef2ac1
From: "CCUK" <sip:01618313800@194.24.251.3>;tag=as0eb5392e
To: <sip:1301@10.0.0.82:5060>;tag=g5VVthPSslPbjLib
Contact: <sip:01618313800@194.24.251.3>
Call-ID: 27584e3a339e535209ea89102043184e@194.24.251.3
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Event: refer;id=1
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 14
SIP/2.0 200 OK (no NAT) to 10.0.0.82:5060
set_destination: Parsing <sip:1301@10.0.0.82:5060> for address/port to
send to
set_destination: set destination to 10.0.0.82, port 5060
Reliably Transmitting:
BYE sip:1301@10.0.0.82:5060 SIP/2.0
Via: SIP/2.0/UDP 194.24.251.3:5060;branch=z9hG4bK56da28c8
From: "CCUK" <sip:01618313800@194.24.251.3>;tag=as0eb5392e
To: <sip:1301@10.0.0.82:5060>;tag=g5VVthPSslPbjLib
Contact: <sip:01618313800@194.24.251.3>
Call-ID: 27584e3a339e535209ea89102043184e@194.24.251.3
CSeq: 104 BYE
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 10.0.0.82:5060
monitor executing ( nice -n 19 soxmix
"/var/spool/asterisk/monitor/agent-1600-asterisk-3407-1115714965-75-in.wav"
"/var/spool/asterisk/monitor/agent-1600-asterisk-3407-1115714965-75-out.wav"
"/var/spool/asterisk/monitor/agent-1600-asterisk-3407-1115714965-75.wav"
&&
rm -f
"/var/spool/asterisk/monitor"/agent-1600-asterisk-3407-1115714965-75-*
) &
after that it's just a series of shutdown SIP messages...
I can't understand that "Subscription-state:
terminated;reason=noresource"
message from * to the phone - any ideas would be warmly welcomed!
Oh, I've tried all combinations of nat=yes/no and canreinvite=yes/no :/
Cheers,
Gavin.