On Wed, 2011-01-05 at 15:47 +0000, Ishfaq Malik wrote:
> 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?
> 
> 

I spotted something new here, when I try to do the blind transfer I get
the following output on the console

== Spawn extension (pack-local, 504, 0) exited non-zero on

So why would it be looking at priority 0 rather than priority 1?

-- 
Ishfaq Malik
Software Developer
PackNet Ltd

Office:   0161 660 3062


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