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 [email protected] got a SIP call transfer from > caller: (REFER)! > SIP transfer to extension 5...@pack-local by [email protected] > > <--- 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:[email protected]:3072;line=guuuyf05>;tag=xck40ix9vp > To: "<incoming mobile number>" <sip:<incoming mobile > number>@x.x.x.x>;tag=as4d0dbc04 > Call-ID: [email protected] > 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:[email protected]: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:[email protected]: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:[email protected]:3072;line=guuuyf05>;tag=xck40ix9vp > Contact: <sip:<incoming mobile number>@x.x.x.x> > Call-ID: [email protected] > 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 -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
