Hi
We've been running asterisk 1.4.17 (deb package) in a production
environment for some while now and are finally taken the plunge to
update to 1.4.38 (Ubuntu servers). All of this is using the RealTime
Architecture
I have upgraded the asterisk version in one of our test environments and
blind transferring seems to have suddenly stopped working. It was
working fine under 1.4.17
So, call comes in to extension 501 who does a blind transfer to
extension 504 at which point the call gets completely cut off.
I ran a SIP trace of this happening and it appears to be attempting to
do the transfer:
<------------->
--- (12 headers 0 lines) ---
Call 7c5d5a603b2aaaa803fd7e451de82...@x.x.x.x got a SIP call transfer from
caller: (REFER)!
SIP transfer to extension 5...@pack-local by pack...@domain.co.uk
<--- Transmitting (NAT) to x.x.x.x:52753 --->
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP
192.168.1.105:3072;branch=z9hG4bK-sgoqylu125ma;received=x.x.x.x;rport=52753
From: <sip:pack...@192.168.1.105:3072;line=guuuyf05>;tag=xck40ix9vp
To: "<incoming mobile number>" <sip:<incoming mobile
number>@x.x.x.x>;tag=as4d0dbc04
Call-ID: 7c5d5a603b2aaaa803fd7e451de82...@x.x.x.x
CSeq: 2 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:<incoming mobile number>@x.x.x.x>
Content-Length: 0
<------------>
set_destination: Parsing <sip:pack...@192.168.1.105:3072;line=guuuyf05> for
address/port to send to
set_destination: set destination to 192.168.1.105, port 3072
Reliably Transmitting (NAT) to x.x.x.x:52753:
NOTIFY sip:pack...@192.168.1.105:3072;line=guuuyf05 SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK121bb8ff;rport
From: "<incoming mobile number>" <sip:<incoming mobile
number>@x.x.x.x>;tag=as4d0dbc04
To: <sip:pack...@192.168.1.105:3072;line=guuuyf05>;tag=xck40ix9vp
Contact: <sip:<incoming mobile number>@x.x.x.x>
Call-ID: 7c5d5a603b2aaaa803fd7e451de82...@87.237.58.231
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Remote-Party-ID: "<incoming mobile number>" <sip:<incoming mobile
number>@x.x.x.x>;privacy=off;screen=no
Event: refer;id=2
Subscription-state: active
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 21
SIP/2.0 183 Ringing
_______________________________________________________________________________________________________________
But as stated above, extension 504 doesn't ring and the call dies.
Now 504 is a valid extensions in the context pack-local
select * from extensions where exten='_5XX';
+-------+------------+-------+----------+-------+-----------------------------------+
| id | context | exten | priority | app | appdata
|
+-------+------------+-------+----------+-------+-----------------------------------+
| 65127 | pack-local | _5XX | 1 | Macro |
stdexten|${EXTEN}|pack-local|PACK |
+-------+------------+-------+----------+-------+-----------------------------------+
Also, attended transfers work without a problem.
Both SIP phones used were Snom phones.
Has anyone encountered an issue like this before?
--
Ishfaq Malik
Software Developer
PackNet Ltd
Office: 0161 660 3062
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