Re: [asterisk-users] ${HASH(SIP_CAUSE, ...)} and peer name
On Tue, Jul 12, 2011 at 00:22, Philippe Sultan philippe.sul...@gmail.comwrote: The destination channel dies right after your Dial statement exits, but you can retrieve the info in the channel that's still alive : exten = _XXX,n,Dial(SIP/${EXTEN}) exten = _XXX,n,NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})}) Works fine on the Asterisk server I'm running (1.8.3.3). Thanks, that works for me as well. Philippe Ido On Mon, Jul 11, 2011 at 11:01 PM, ik ido...@gmail.com wrote: Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error message. So how can I get SIP_CAUSE, or how can I get the peer name ? Thanks, Ido -- _ -- 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 -- Philippe Sultan -- _ -- 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 -- _ -- 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
[asterisk-users] ${HASH(SIP_CAUSE, ...)} and peer name
Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error message. So how can I get SIP_CAUSE, or how can I get the peer name ? Thanks, Ido -- _ -- 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] ${HASH(SIP_CAUSE, ...)} and peer name
The destination channel dies right after your Dial statement exits, but you can retrieve the info in the channel that's still alive : exten = _XXX,n,Dial(SIP/${EXTEN}) exten = _XXX,n,NoOp(SIP return code : ${HASH(SIP_CAUSE,${CDR(dstchannel)})}) Works fine on the Asterisk server I'm running (1.8.3.3). Philippe On Mon, Jul 11, 2011 at 11:01 PM, ik ido...@gmail.com wrote: Hello, I'm trying to figure out what was the return code of SIP for a call. The problem is that HASH(SIP_CAUSE) require a peer name, but when I try to retrieve the peer name using ${CHANNEL(peername)}, I have an error message that CHANNEL does not have peername or it is not available to be used. I tried to print it with NOOP on a live channel, and also after hangup, both with the same error message. So how can I get SIP_CAUSE, or how can I get the peer name ? Thanks, Ido -- _ -- 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 -- Philippe Sultan -- _ -- 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