commit broke one of my trunks
Hi. I am having a problem where I am using an asterisk box as a gateway (asterisk 1.4) and the following commit breaks this trunk -- I get a 491 response for any revision after that commit and it works correctly before this commit. Here is what git says: commit 56d67eadf66c1b22652ff1f77002a8d024a93fca Author: Anthony Minessale <anthm@freeswitch.org> Date: Mon Aug 1 10:22:55 2011 -0500 sdp_m_per_ptime is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well So, I am not sure what is the matter, I am just using g711 ulaw, but in any case, is this a bug, or is there a way to configure for this gateway onlyto get the old behavior back? Thanks in advance for any suggestions. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
Like it says set the variable sdp_m_per_ptime=false before calling or set it it in vars.xml to make it permanent. On Aug 10, 2011 3:27 AM, <covici@ccs.covici.com> wrote:
Hi. I am having a problem where I am using an asterisk box as a gateway (asterisk 1.4) and the following commit breaks this trunk -- I get a 491 response for any revision after that commit and it works correctly before this commit. Here is what git says: commit 56d67eadf66c1b22652ff1f77002a8d024a93fca Author: Anthony Minessale <anthm@freeswitch.org> Date: Mon Aug 1 10:22:55 2011 -0500
sdp_m_per_ptime is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well
So, I am not sure what is the matter, I am just using g711 ulaw, but in any case, is this a bug, or is there a way to configure for this gateway onlyto get the old behavior back?
Thanks in advance for any suggestions.
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
Yep, but it also said the behavior is undefined, so I could not figure out why this would happen -- no strange codecs or anything that I know of -- any asterisk setting I can change instead? Anthony Minessale <anthony.minessale@gmail.com> wrote:
Like it says set the variable sdp_m_per_ptime=false before calling or set it it in vars.xml to make it permanent. On Aug 10, 2011 3:27 AM, <covici@ccs.covici.com> wrote:
Hi. I am having a problem where I am using an asterisk box as a gateway (asterisk 1.4) and the following commit breaks this trunk -- I get a 491 response for any revision after that commit and it works correctly before this commit. Here is what git says: commit 56d67eadf66c1b22652ff1f77002a8d024a93fca Author: Anthony Minessale <anthm@freeswitch.org> Date: Mon Aug 1 10:22:55 2011 -0500
sdp_m_per_ptime is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well
So, I am not sure what is the matter, I am just using g711 ulaw, but in any case, is this a bug, or is there a way to configure for this gateway onlyto get the old behavior back?
Thanks in advance for any suggestions.
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
the problem is when you mix codecs with different ptimes the only real way to do it is how we now do it with multiple m= lines per the RFC, asterisk just needs to learn how to look through multiple rtp streams or simply set the var when you know you are calling asterisk and you should be ok. On Wed, Aug 10, 2011 at 8:38 AM, <covici@ccs.covici.com> wrote:
Yep, but it also said the behavior is undefined, so I could not figure out why this would happen -- no strange codecs or anything that I know of -- any asterisk setting I can change instead?
Anthony Minessale <anthony.minessale@gmail.com> wrote:
Like it says set the variable sdp_m_per_ptime=false before calling or set it it in vars.xml to make it permanent. On Aug 10, 2011 3:27 AM, <covici@ccs.covici.com> wrote:
Hi. I am having a problem where I am using an asterisk box as a gateway (asterisk 1.4) and the following commit breaks this trunk -- I get a 491 response for any revision after that commit and it works correctly before this commit. Here is what git says: commit 56d67eadf66c1b22652ff1f77002a8d024a93fca Author: Anthony Minessale <anthm@freeswitch.org> Date: Mon Aug 1 10:22:55 2011 -0500
sdp_m_per_ptime is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well
So, I am not sure what is the matter, I am just using g711 ulaw, but in any case, is this a bug, or is there a way to configure for this gateway onlyto get the old behavior back?
Thanks in advance for any suggestions.
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
OK, I will try and see what happens. Anthony Minessale <anthony.minessale@gmail.com> wrote:
the problem is when you mix codecs with different ptimes the only real way to do it is how we now do it with multiple m= lines per the RFC, asterisk just needs to learn how to look through multiple rtp streams or simply set the var when you know you are calling asterisk and you should be ok.
On Wed, Aug 10, 2011 at 8:38 AM, <covici@ccs.covici.com> wrote:
Yep, but it also said the behavior is undefined, so I could not figure out why this would happen -- no strange codecs or anything that I know of -- any asterisk setting I can change instead?
Anthony Minessale <anthony.minessale@gmail.com> wrote:
Like it says set the variable sdp_m_per_ptime=false before calling or set it it in vars.xml to make it permanent. On Aug 10, 2011 3:27 AM, <covici@ccs.covici.com> wrote:
Hi. I am having a problem where I am using an asterisk box as a gateway (asterisk 1.4) and the following commit breaks this trunk -- I get a 491 response for any revision after that commit and it works correctly before this commit. Here is what git says: commit 56d67eadf66c1b22652ff1f77002a8d024a93fca Author: Anthony Minessale <anthm@freeswitch.org> Date: Mon Aug 1 10:22:55 2011 -0500
sdp_m_per_ptime is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well
So, I am not sure what is the matter, I am just using g711 ulaw, but in any case, is this a bug, or is there a way to configure for this gateway onlyto get the old behavior back?
Thanks in advance for any suggestions.
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
---------------------------------------------------- Alternatives:
---------------------------------------------------- _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire
AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch
FreeSWITCH Developer Conference sip:888@conference.freeswitch.org googletalk:conf+888@conference.freeswitch.org pstn:+19193869900
_______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
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Anthony Minessale
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