This is a heads up to everyone Apparently this is a known but in the latest version on asterisk 1.4, 1.6 and 1.8
http://www.freepbx.org/forum/freepbx/users/transfer-bug-on-asterisk-1-4-38-1-6-2-15-and-1-8-0-1 https://issues.asterisk.org/view.php?id=18185 On Thu, 2011-01-06 at 13:10 +0000, Ishfaq Malik wrote: > 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 504@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 -- 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
