Re: [asterisk-users] Blind Transfer not working - 1.4.38
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 +, Ishfaq Malik wrote: On Wed, 2011-01-05 at 15:47 +, 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 7c5d5a603b2803fd7e451de826e4@x.x.x.x got a SIP call transfer from caller: (REFER)! SIP transfer to extension 504@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:PACK501@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: 7c5d5a603b2803fd7e451de826e4@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:PACK501@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:PACK501@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:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp Contact: sip:incoming mobile number@x.x.x.x Call-ID: 7c5d5a603b2803fd7e451de826e4@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 -- _ -- 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
Re: [asterisk-users] Blind Transfer not working - 1.4.38
Hi, 1.6.2.16rc1 does not have this problem (that`s why I am running a release candidate right now). Can`t say about 1.4 versions, but it`s safe to say whatever they fixed will be out in the next version. Mike -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ishfaq Malik Sent: Friday, January 14, 2011 9:55 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Blind Transfer not working - 1.4.38 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 +, Ishfaq Malik wrote: On Wed, 2011-01-05 at 15:47 +, 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 7c5d5a603b2803fd7e451de826e4@x.x.x.x got a SIP call transfer from caller: (REFER)! SIP transfer to extension 504@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;rpor t=52753 From: sip:PACK501@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: 7c5d5a603b2803fd7e451de826e4@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:PACK501@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:PACK501@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:PACK501@192.168.1.105:3072;line=guuuyf05;tag=xck40ix9vp Contact: sip:incoming mobile number@x.x.x.x Call-ID: 7c5d5a603b2803fd7e451de826e4@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 -- _ -- 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
Re: [asterisk-users] Blind Transfer not working - 1.4.38
On Wed, 2011-01-05 at 15:47 +, 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 7c5d5a603b2803fd7e451de82...@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: 7c5d5a603b2803fd7e451de82...@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: 7c5d5a603b2803fd7e451de82...@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 -- _ -- 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