Re: [asterisk-users] How can I get SIP/SDP values to a variable?
Thanks for the quick reply, I was afraid of that. Oh well. :) Rennes Neps Elion Ettevõtted AS rennes.n...@elion.ee -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kevin P. Fleming Sent: Tuesday, March 13, 2012 1:31 PM To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] How can I get SIP/SDP values to a variable? On 03/13/2012 05:58 AM, Rennes Neps wrote: > I wonder if there's a way to read SDP values into a variable in Asterisk? I > have successfully used SIP_HEADER function to get all I want out of SIP part > of the message, no problem. But I would like to be able to read SDP part as > well, anyone? > Reason I want to to this is: testing the SIP_ALG condition in incoming invite > message. Majority of cases can be detected just by comparing src ip address > and "via", but some devices only rewrite SDP "c" value for example and so on > ... > I haven't found any information anywhere how to achieve reading SDP with > asterisk. I can use 1.8 and 10 versions. There is no mechanism in Asterisk to do this. The most practical approach would probably be to put a stateless SIP proxy in front of Asterisk and write whatever logic you like there, causing it to add one or more headers to the SIP messages that can be accessed from the dialplan in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- 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
Re: [asterisk-users] How can I get SIP/SDP values to a variable?
On 03/13/2012 05:58 AM, Rennes Neps wrote: I wonder if there's a way to read SDP values into a variable in Asterisk? I have successfully used SIP_HEADER function to get all I want out of SIP part of the message, no problem. But I would like to be able to read SDP part as well, anyone? Reason I want to to this is: testing the SIP_ALG condition in incoming invite message. Majority of cases can be detected just by comparing src ip address and "via", but some devices only rewrite SDP "c" value for example and so on ... I haven't found any information anywhere how to achieve reading SDP with asterisk. I can use 1.8 and 10 versions. There is no mechanism in Asterisk to do this. The most practical approach would probably be to put a stateless SIP proxy in front of Asterisk and write whatever logic you like there, causing it to add one or more headers to the SIP messages that can be accessed from the dialplan in Asterisk. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com & www.asterisk.org -- _ -- 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] How can I get SIP/SDP values to a variable?
Hello all! I wonder if there's a way to read SDP values into a variable in Asterisk? I have successfully used SIP_HEADER function to get all I want out of SIP part of the message, no problem. But I would like to be able to read SDP part as well, anyone? Reason I want to to this is: testing the SIP_ALG condition in incoming invite message. Majority of cases can be detected just by comparing src ip address and "via", but some devices only rewrite SDP "c" value for example and so on ... I haven't found any information anywhere how to achieve reading SDP with asterisk. I can use 1.8 and 10 versions. Thanks in advance Rennes Neps Elion Ettevõtted AS rennes.n...@elion.ee -- _ -- 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
