February 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Global Variables forgotten through the public context?
by UV 09 Feb '09

09 Feb '09

Another question: When I try routing calls through the public context to the default context, global variables (set in vars.xml) seem to be “forgotten” and appear blank. I’m trying a very simple scenario of an incoming call on the public context routed to a phone number which is rightfully captured at “local.example.com” but all three variables ${outbound_caller_id_number}, ${outbound_caller_id_name} and ${default_gateway} appear to be blanked out (according to the console debug output). Again, any idea what am I doing wrong?

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Openzap r654 doesn't compile. Error with ozmod_libpri
by Helmut Kuper 09 Feb '09

09 Feb '09

Hello, during some imporvements on q931toPcap as well as debugging my TDM PRI problem with loosing state sync after some time I updated the code to latest trunk (r654). After doing a bootstrap in FS trunk directory I tried to compile openzap in libs/openzap. I got this error: /bin/sh ./libtool --tag=CC --mode=link gcc -g -O2 -o ozmod_libpri.la -rpath /usr/local/openzap/mod -lm -lpthread -ldl libtool: link: libtool library `ozmod_libpri.la' must begin with `lib' Try `libtool --help --mode=link' for more information. make: *** [ozmod_libpri.la] Error 1 Additionally I wonder what ozmod_libpri is. Do I have consider it for Q931 capturing? regards Helmut

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FreeSwitch setup as a "Dumb" SBC
by Adam Long 09 Feb '09

09 Feb '09

Hi Guys, I've been working at setting up a couple of FreeSwitch nodes as a topology hiding SBCs that handles both ingress traffic from my providers/peers and pass traffic up to an openser router that then routes call across the cluster of SBCs through which they reach the destination. I have OpenSIPS/SER setup doing DB route lookups and ENUM with LCR/Serial forking etc. My question is what would be the best way to send a call out to a destination choosen by the OpenSER router? For example: SIP Provider -- > SBC --- > OpenSER ---- ( route lookup returns 123.123.123.4 as dest ) -- > SBC --- > 123.123.123.4 I was thinking something along the lines of adding a "X-Route-To: +1NXXNXXXXXX(a)123.123.123.4" with openser and then something like this in the SBC. <context name="from-sipcore"> <extension name="outboundroute"> <action application="bridge" data="sofia/external/${sip_h_X-Route-To}" /> </extension> </context> Is this a wise approach, is there anything I could do to do this better? I'd like to keep the logic in the SBCs as simple as possible. I am pretty familiar with SIP but my knowledge fades when it gets into the nitty gritty of routing. ie the Contact: and Via: headers and all that good stuff. I should also state I have two profiles defined one for the internal/private "core" network and one for the outside "external" network. Any thoughts on this at all would be greatly appreciated. Am I missing something in the SIP spec that would allow for this is a standardized way? Regards, -Adam

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Newbie - point me in the right direction
by John O'Brien 09 Feb '09

09 Feb '09

Hi, I am a real newbie. I have been building Asterisk based applications for a couple of years now. I am looking at migrating these apps to FreeSwitch - eventually. I want to do this gradually - I need to keep things running in the meantime. I have two Asterisk boxes, A1 & A2, each running a separate telephony app. We have an external SIP service with DID's NNNNN200 -> NNNNN299. We want to direct the incoming SIP calls so that the DID's NNNNN200 -> NNNNN219 go to Asterisk server A1 and NNNNN220 -> NNNNN299 to Asterisk server A2. Yes we really just want the calls switched on the DID. I'm struggling to know where to start - can someone point me in the right direction? Regards, John

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Making a system call with LUA
by Nik Middleton 09 Feb '09

09 Feb '09

In the absence of any directives on lua/mysql, is it possible to launch a PHP script from lua? All I need to do is pass some data to update a db record. I don't need to have any links to the call as I intend to call is in the hang-up callback Regards,

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Re: [Freeswitch-users] SIP Authentication
by Ali Al-Rubaie 09 Feb '09

09 Feb '09

Thanks so much Anthony but I have one more question: I was checking the source file sofia_reg.c and it seems that the code had been written iin such a way that FreeSWITCH can authenticate SIP agents based on RFC2069 and RFC2617. Is that conclusion correct? Thanks in advance, Message: 2 Date: Thu, 5 Feb 2009 10:46:54 -0600 From: Anthony Minessale <anthony.minessale(a)gmail.com> Subject: Re: [Freeswitch-users] SIP Authentication To: freeswitch-users(a)lists.freeswitch.org Message-ID: <191c3a030902050846o60047c30pa2890707eae386d6(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" It's optional for us but it's mandatory for the client if we exercise the option which we have opted to always do =D There is no way in the code to disable sending it because we prefer the more secure version of SIP auth. So again it's a bug in the client for not following the protocol. It would be considered a feature in FreeSWITCH to support limping for the sake of this broken client and we currently do not have any plans for implementing this feature. On Thu, Feb 5, 2009 at 10:34 AM, Ali Al-Rubaie <kerrada2003(a)yahoo.com>wrote: > > We're using HelpCaster softphone. > > The issue here is that in Digest Authentication, if the server sends the > parameter "qop" in the challenge then the client should respond with the > "cnonce" parameter. The parameter "qop" is optional in Digest Auth. So the > question here is that, can we configure FreeSWITCH so that it will not send > "qop" in the challenge? > > Thanks! > > --- On *Wed, 2/4/09, freeswitch-users-request(a)lists.freeswitch.org < > freeswitch-users-request(a)lists.freeswitch.org>* wrote: > > From: freeswitch-users-request(a)lists.freeswitch.org < > freeswitch-users-request(a)lists.freeswitch.org> > Subject: Freeswitch-users Digest, Vol 32, Issue 39 > To: freeswitch-users(a)lists.freeswitch.org > Date: Wednesday, February 4, 2009, 2:05 PM > > Send Freeswitch-users mailing list submissions to > freeswitch-users(a)lists.freeswitch.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > or, via email, send a message with subject or body 'help' to > freeswitch-users-request(a)lists.freeswitch.org > > You can reach the person managing the list at > freeswitch-users-owner(a)lists.freeswitch.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Freeswitch-users digest..." > > > Today's Topics: > > 1. Re: SIP Authentication (Brian West) > 2. Re: origainate through sofia gateway (Michael Collins) > 3. Recording background music and voice is out of sync (Daniel Liang) > 4. Re: Q931 decoding Update (Gopalakrishnan A.N) > 5. mod_limit (Chav Paskov) > 6. Re: mod_limit (Michael Collins) > 7. Re: mod_limit (Chav > Paskov) > 8. Re: mod_limit (Michael Collins) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Wed, 4 Feb 2009 10:52:45 -0600 > From: Brian West <brian(a)freeswitch.org> > Subject: Re: [Freeswitch-users] SIP Authentication > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: <7DAC91F6-2FD3-464D-AA81-321EBCADC8C0(a)freeswitch.org> > Content-Type: text/plain; charset=US-ASCII; format=flowed; delsp=yes > > What client is this? I also notice we receive port 3458 and reply to > port 1059... > > /b > > On Feb 4, 2009, at 10:17 AM, Ali Al-Rubaie wrote: > > > What I have noted is that the client does not send the values for > > "cnonce" and "nc" in the response. I'm not sure if > this is the > > reason, however how this problem can be solved? > > > > Thanks, > > > > Ali > > > > > ------------------------------ > > Message: > 2 > Date: Wed, 4 Feb 2009 09:41:07 -0800 > From: Michael Collins <msc(a)freeswitch.org> > Subject: Re: [Freeswitch-users] origainate through sofia gateway > To: freeswitch-users(a)lists.freeswitch.org > Message-ID: > <87f2f3b90902040941r61d669aaie949aa7cc8578a9a(a)mail.gmail.com> > Content-Type: text/plain; charset=ISO-8859-1 > > I'll make sure the substance of this is in the wiki and I'll look for > references to the deprecated way and remove those. > -MC > > On Wed, Feb 4, 2009 at 6:09 AM, Anthony Minessale > <anthony.minessale(a)gmail.com> wrote: > > Where did you learn how to use js this way? > > session.originate is being misused here and is depricated and may be > > removed. > > > > the first arg to session.originate is either undefined or a *different* > > session (the a leg) > > > > session1 = new Session(); > > session1.originate(undefined, > > > "{ignore_early_media=true}user/1008(a)192.168.1.122"); > > > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > > > //once you have session1 when you originate session2 you pass session1 as > > the arg > > // the effective_caller_id is taken from session1 > > > > session2 = new Session(); > > session2.originate(session1, > "sofia/gateway/halonet/0225490317"); > > > > Anyway this whole code is depricated in favor of this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008(a)192.168.1.122"); > > if (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session2 = new Session("sofia/gateway/halonet/0225490317", > session1); > > } > > > > and could be further refactored down to this: > > > > session1 = new > Session("{ignore_early_media=true}user/1008(a)192.168.1.122"); > > if > (session1.ready()) { > > > session1.setVariable("effective_caller_id_number","fixed0248b"); > > session1.execute("bridge", > "sofia/gateway/halonet/0225490317"); > > } > > > > or down to this one line of code that will setup the call detached from > the > > script and exit. > > > > var result = apiExecute("originate", > > > "{effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008(a)192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > if you dont care about the result and want to exit even before the call is > > completed. > > > > var result = apiExecute("bgapi", "originate > > > {effective_caller_id_number=fixed0248b,origination_caller_id_number=1000,ignore_early_media=true}user/1008(a)192.168.1.122 > > bridge:sofia/gateway/halonet/0225490317 inline"); > > > > > > > > On Wed, Feb 4, 2009 at > 2:51 AM, Jacek Sokulski > <jsokulski(a)dotsystems.pl> > > wrote: > >> > >> We have tried setting both effective_caller_id_number and > >> origination_caller_id_number: > >> > >> > >> > session1.originate(session1,"{origination_caller_id_number=fixed0248b}sofia/gateway/halonet/0225490317",15); > >> but the problem still exists. The solution we have found for the case > >> when we originate two calls, local and external, is as follow: > >> > >> session1 = new Session(); > >> > session1.originate(session1,"user/1003(a)192.168.1.122",15);//local > >> if(session1.ready()) { > >> session1.execute("execute_extension","00930691688627 > XML > >> default");//external > >> } > >> > >> so the external call goes through the dialplan. > >> It does not work if both calls are external. One possible solution > could > >> > be > >> to pass the originating call through dialplan (loopback?) but we have > not > >> managed > >> to figure out how to do it. > >> > >> Thanks > >> Jacek > >> > >> Dnia 03-02-2009, wto o godzinie 14:31 -0300, Nicolas Brenner pisze: > >> > Oops! Well, fortunately I don't use that voip provider > anymore (nor the > >> > script). > >> > > >> > Thanks Brian. > >> > > >> > Nicolas > >> > > >> > On Tue, Feb 3, 2009 at 2:25 PM, Brian West > <brian(a)freeswitch.org> wrote: > >> > > YOU should NEVER use this method or call setCallerData at > all you > >> > > should use the correct methods to override the callerid. > >> > > > >> > > If its a B-Leg born from an A-Leg you use these on the on > the A-Leg: > >> > > > >> > > > >> > > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#effective_caller_id_number > >> > > > >> > > If you're originating you use this: > >> > > > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_name > >> > > > >> > > > http://wiki.freeswitch.org/wiki/Channel_Variables#origination_caller_id_num… > >> > > > >> > > /b > >> > > >> > _______________________________________________ > >> > Freeswitch-users mailing list > >> > Freeswitch-users(a)lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> _______________________________________________ > >> Freeswitch-users mailing list > >> Freeswitch-users(a)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/ > > > > AIM: anthm > > MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale(a)hotmail.com> > > GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <PAYPAL%3Aanthony.minessale(a)gmail.com> > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888@conference.freeswitch.org <sip%3A888(a)conference.freeswitch.org> > > > iax:guest@conference.freeswitch.org/888 > > googletalk:conf+888@conference.freeswitch.org <googletalk%3Aconf%2B888(a)conference.freeswitch.org> > > pstn:213-799-1400 > > > > _______________________________________________ > > Freeswitch-users mailing list > > Freeswitch-users(a)lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > ------------------------------ > > Message: 3 > Date: Wed, 4 Feb 2009 09:43:10 -0800 > From: "Daniel Liang" <Daniell(a)airg.com> > Subject: [Freeswitch-users] Recording background music and voice is > out of sync > To: <freeswitch-users(a)lists.freeswitch.org> > Message-ID: > <0B02E756F603CC409EB553879B090CC80A23EBB5(a)HPEXCHVS01.exchange.airg> > Content-Type: text/plain; charset="us-ascii" > > What I did was the following: > > First, I sent the > playback command: > > call-command: execute > execute-app-name: playback > execute-app-arg: <filename> > > Then I send uuid_record (Sorry, it was not Record command): > > api uuid_record <uuid> start <filename> 120 > > I also tried replacing the playback command with: > api uuid_displace <uuid> start <filename> 0 mux > > But the end results are the same. The recorded user's voice is about 0.5 > second behind the expected result. > > Thanks, > Daniel > > > -----Original Message----- > From: freeswitch-users-bounces(a)lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of > Brian West > Sent: February 3, 2009 6:36 PM > To: freeswitch-users(a)lists.freeswitch.org > Subject: Re: [Freeswitch-users] Recording background music and voice is > outof sync > > Can you show us an example of how you're doing this? Playback and > Record aren't async so > you'll need to show us how you're doing > this. > > Also don't hijack threads you hit replay on the one "Re: [Freeswitch- > users] FreeSwitch setup as a "Dumb" SBC" as the subject, deleted > the > subject and started a new body. That hijacks the thread and that can > cause your problem to go ignored in some cases if people aren't > interested in the thread topic depending on how their reader threads the > emails. > > Please click new message and type freeswitch- users(a)lists.freeswitch.org > in and then input your subject and body to start a new thread. > > Thanks, > Brian West > FreeSWITCH.org > > > On Feb 3, 2009, at 8:21 PM, Daniel Liang wrote: > > > Hi, > > > > I was trying to record a background music with a user's voice at the > > same time. I did a playback and started recording. But the recorded > > user's voice and the background music is about 0.5 second out of sync. > > > I also tried > to use uuid_displace instead of playback, but I got the > > same result. > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org <http://www.freeswitch.org/> > >

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Compiling FreeSWITCH for AstLinux
by Kristian Kielhofner 09 Feb '09

09 Feb '09

Hey guys, I've finally gotten around to trying to compile FreeSWITCH for AstLinux. Here is the branch I've created for it: http://astlinux.svn.sourceforge.net/viewvc/astlinux/branches/astlinux-frees… Right now it is bombing trying to run configure for libs/sqlite, as shown in the build log here: http://astbuild.star2star.com/astlinux-freeswitch-build.log Here is the config.log for sqlite: http://astbuild.star2star.com/sqlite-config.log I'll continue to dig into this but in the meantime I thought I'd get some extra eyeballs on it... Thanks! P.S. - Yes, yes I know "AstLinux" isn't the best name for a distro with FreeSWITCH. Depending on my success here I have some other ideas... -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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connecting to mysql using lua
by Nik Middleton 09 Feb '09

09 Feb '09

Hi Guys I want to access Mysql 5 from lua. The wiki is not too clear on this. Do I need to install lua and lua mysql? Regards

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Questions about speex
by Tamas 09 Feb '09

09 Feb '09

Hello, I'm looking for the best codec/scenario for the last-mile and checked speex codec capabilities (http://www.speex.org/comparison/nb_codecs.png) This will be FS-FS interconnect (where one side uses portaudio, aka a simple softphone). As I see, it would be worth to use Speex in VBR mode and also to turn on DTX. Would it be possible with current mod_speex? As I saw, currently vad and vbr options are set to 0. Would setting these hardwired options to 1 make the trick? I was thinking about G.726-32 as it was suggested last week on irc, but it has still too high bandwidth requirements and Speex seems to have better MOS values over 10kbps (or 6kbps with VBR). Thanks in advance, Tamas

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Freeswitch not processing calls
by Ankit Gandhi 09 Feb '09

09 Feb '09

Recently I noticed that fs (1.0.3 RC1) is not processing the calls. Previously about 50-60 calls were remaining active at any time. But then suddenly, only 5-6 active calls were there. So I checked sip INVITE's in ngrep, and I noticed that originator were sending CANCEL after INVITE. So I contacted them and they were telling that while sending calls, your gateways time outs after a call is sent, and so they send CANCEL after INVITE. My architecture is as follows : originator -> switch (our fs) -> terminator. CPU : Intel(R) Xeon(R) CPU X3220 @ 2.40GHz OS : Ubuntu 6.06.1 LTS 32-bit RAM : 4 GB We had also set ulimits on command prompt every time before starting fs with nc mode as follows : ulimit -c unlimited; ulimit -n 999999; ulimit -s 244; /usr/local/freeswitch/bin/freeswitch -nc In the ngrep trace, I noticed that fs was not sending INVITE to terminator in between the INVITE and CANCEL from originator. So what could be the reason for INVITE not being processed by fs and getting timeout from originator. Here is the sip trace for a sample call. Check the duration between INVITE and CANCEL from originator. -------------------------------------------------------------------- U 2009/02/07 13:16:44.354443 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 INVITE sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>. Content-Type: application/sdp. CSeq: 1 INVITE. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Contact: sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone. Allow: INVITE,CANCEL,BYE,ACK,REFER,UPDATE,INFO. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 317. . v=0. o=MG4000|2.0 492612 492612 IN IP4 66.151.208.138. s=-. c=IN IP4 66.151.208.138. t=0 0. m=audio 38832 RTP/AVP 97 18 98 96 101 13. a=rtpmap:97 G.729b/8000. a=rtpmap:98 G.729a/8000. a=rtpmap:96 G.729ab/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=fmtp:18 annexb=yes. a=ptime:20. a=rtpmap:13 CN/8000. # U 2009/02/07 13:16:44.354633 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 100 Trying. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Content-Length: 0. U 2009/02/07 13:16:59.208008 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 CANCEL sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>. CSeq: 1 CANCEL. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Supported: timer,100rel. Max-Forwards: 70. Content-Length: 0. . # U 2009/02/07 13:16:59.208095 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 200 OK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. CSeq: 1 CANCEL. Content-Length: 0. . # U 2009/02/07 13:16:59.208134 fs.fs.fs.fs:5060 -> ori.ori.ori.ori:2000 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>;tag=7B6p78S5SDF0j. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-hacked. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH. Supported: timer, precondition, path, replaces. Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Length: 0. U 2009/02/07 13:16:59.253482 ori.ori.ori.ori:2000 -> fs.fs.fs.fs:5060 ACK sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone SIP/2.0. Call-ID: 7008117790783130205-1234012604(a)ori.ori.ori.ori. From: <sip:xxxxxxxxxxx@ori.ori.ori.ori:2000;user=phone>;tag=13475. To: <sip:yyyyyyyyyyyy@fs.fs.fs.fs:5060;user=phone>;tag=7B6p78S5SDF0j. CSeq: 1 ACK. Via: SIP/2.0/UDP ori.ori.ori.ori:2000;branch=z9hG4bK-6141d60007b94a5d-4297d09d-1. Max-Forwards: 70. Content-Length: 0. ------------------------------------------------------- -- View this message in context: http://www.nabble.com/Freeswitch-not-processing-calls-tp21909687p21909687.h… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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