Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end. /Peter -----Ursprungligt meddelande----- Från: Peter Olsson Skickat: den 30 maj 2009 09:01 Till: freeswitch-users@lists.freeswitch.org Ämne: FW: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? And just to be clear, even though media flows in one direction (from FS to phone), I get no audio at all. And by the way, I mean SVN, not SNV :) Sorry for double posting... /Peter ________________________________________ Från: Peter Olsson Skickat: den 30 maj 2009 08:58 Till: freeswitch-users@lists.freeswitch.org Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I'll try to do this during this weekend. I've looked through the SNV logs, and I really can't find a good reason for this to happen. And when looking into wireshark I can see RTP audio flowing from FS to my SIP phone, but not in the other direction. So this still makes me wonder if something has happened to sofia (that sets up the media incorrectly)... And also when I hangup the call, it takes about a minute for FS to detect this, and it reports hangup reason unknown. But as I said, I'll look into this a bit deeper during this weekend, and file a jira case when I have some more information. //Peter ________________________________________ Från: freeswitch-users-bounces@lists.freeswitch.org [freeswitch-users-bounces@lists.freeswitch.org] för Michael Jerris [mike@jerris.com] Skickat: den 29 maj 2009 19:35 Till: freeswitch-users@lists.freeswitch.org Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Can you try to do a binary search and nail down the exact version that caused this issue and then file a bug on http://jira.freeswitch.org. Thanks Mike On May 29, 2009, at 9:55 AM, Peter Olsson wrote: I'm on Windows, so I have everything under my fs directory, but I deleted the complete directory and did everything from scratch... /Peter Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Anthony Minessale Skickat: den 29 maj 2009 15:46 Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? did you delete the binaries from /usr/local/freeswitch/bin , lib and mod too ? On Fri, May 29, 2009 at 8:33 AM, Peter Olsson <peter.olsson@visionutveckling.se<mailto:peter.olsson@visionutveckling.se>> wrote: Nope - it's not :) Just to make sure I even deleted the source completely, and checked everything out again. Från: freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org> [mailto:freeswitch-users-bounces@lists.freeswitch.org<mailto:freeswitch-users-bounces@lists.freeswitch.org>] För Brian West Skickat: den 29 maj 2009 15:26 Till: freeswitch-users@lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org> Ämne: Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Nope its not a sofia issue... its build skew ;) On May 29, 2009, at 8:24 AM, Peter Olsson wrote: I've looked into this a bit more now, and I think it is a sofia issue, I will look trough the changes in sofia since I had the last working configuration, and see if I find anything. /Peter !DSPAM:4a201fa932931035648682!

Thanks for the reply! Will this really help though? From what I understand of the change that breaks the compatibility, it will always send this header in the "200 OK" message (my problem is incoming calls to FS). The fix you're describing, isn't it when calling from FS to the other end? This way it works either way, it's just when the PBX gets this in the 200 OK message with this header that it stops working. /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Kristian Kielhofner Skickat: den 1 juni 2009 15:20 Till: freeswitch-users@lists.freeswitch.org Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? Peter, Ouch. Your PBX is broken. It shouldn't do that. Luckily FreeSWITCH provides a way to select RPID/PAI/none: http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation On Mon, Jun 1, 2009 at 5:57 AM, Peter Olsson <peter.olsson@visionutveckling.se> wrote:

Here is an update for this issue (SFSIP-149). I've raised a jira case for this. It was not a RTP problem, the problem is caused by the new "P-Asserted-Identity:"-header added in r13492. This causes my connected PBX not to ACK/accept the 200 OK sent from FS, and the call is never answered. FS still thinks it's answered though, and starts to send RTP data to the other end.

/Peter

-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com _______________________________________________ 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 !DSPAM:4a23d69e32932135138353!

Your PBX is broken but now I have fixed it to only do that feature with Polycom for now. /b On Jun 1, 2009, at 8:32 AM, Peter Olsson wrote:

Thanks for the reply!

Will this really help though? From what I understand of the change that breaks the compatibility, it will always send this header in the "200 OK" message (my problem is incoming calls to FS). The fix you're describing, isn't it when calling from FS to the other end? This way it works either way, it's just when the PBX gets this in the 200 OK message with this header that it stops working.

/Peter

Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com

Ahh... That's what I get for not reading the entire thread! On Mon, Jun 1, 2009 at 9:32 AM, Brian West <brian@freeswitch.org> wrote:

No he's talking about the one from SFSIP-111, Again that shouldn't matter... I'll fix that.

/b

-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

Brian - you're the man! :) /Peter -----Ursprungligt meddelande----- Från: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] För Brian West Skickat: den 1 juni 2009 16:02 Till: freeswitch-users@lists.freeswitch.org Ämne: [SPAM] - Re: [Freeswitch-users] Something broken in RTP in latest trunk (r13502)? I made the PAI in the 18X and 200 be present ONLY if you set callee_id_name. Then I fixed the update on transfer to only work on snom and polycom along with uuid_display and uuid_hold will only wend the display info for snom and polycom till I have others test and provide me the confirmation those work with other phones too. /b On Jun 1, 2009, at 8:51 AM, Kristian Kielhofner wrote:

Ahh... That's what I get for not reading the entire thread!

On Mon, Jun 1, 2009 at 9:32 AM, Brian West <brian@freeswitch.org> wrote:

...

No he's talking about the one from SFSIP-111, Again that shouldn't matter... I'll fix that.

/b

-- Kristian Kielhofner http://www.astlinux.org http://blog.krisk.org http://www.star2star.com http://www.submityoursip.com http://www.voalte.com

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Brian West brian@freeswitch.org -- Meet us at ClueCon! http://www.cluecon.com _______________________________________________ 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 !DSPAM:4a23e16232931225569523!