March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 93, Issue 198
by Snabel Kabiya 31 Mar '14

31 Mar '14

Hi All, I'm trying to use freeswitch International country codes regular expression in my lua script, but i keep getting this error: unexpected symbol near '/' stack traceback: here is the regular expression I'm trying to use: "^\+(1|2[1-689]\d|2[07]|3[0-469]|3[578]\d|4[0-13-9]|42\d|5[09]\d|5[1-8]|6[0-6]|6[7-9]\d|7|8[035789]\d|8[1246]|9[0-58]|9[679]\d)(\d+)" I've tried using this also: "^\\+(1|2[1-689]\\d|2[07]|3[0-469]|3[578]\\d|4[0-13-9]|42\\d|5[09]\\d|5[1-8]|6[0-6]|6[7-9]\\d|7|8[035789]\\d|8[1246]|9[0-58]|9[679]\\d)(\\d+)") link: https://wiki.freeswitch .org/wiki/Regular_Expression#International_country_codes what should i change? Thanks, Snabel On Mon, Mar 31, 2014 at 7:38 PM, < freeswitch-users-request(a)lists.freeswitch.org> wrote: > 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: RTCP FIR on floor change/new member join (Hristo Trendev) > 2. Windows Env - Can't Build ManagedEslTest.2010 or 2012 (Sean Devoy) > > > ---------- Forwarded message ---------- > From: Hristo Trendev <dist.lists(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Mar 2014 17:52:29 +0200 > Subject: Re: [Freeswitch-users] RTCP FIR on floor change/new member join > Hi, > > One of the browsers that I used for the initial testing was Chrome on > Android 4.3. I guess this was causing some of the problems. > > I retested with the latest version from master (0876df6). When I make the > test with two desktop browsers indeed a new keyframe seems to be requested > on every floor change, as in this case the video stream switches almost > instantly. > > However, the problem is still present for anyone who has just joined a > conference. Unless the floor changes shortly after someone joins, he has to > wait for the next keyframe. This is really simple to reproduce if there is > only one speaker in the conference and all the others who are joining the > conference are muted by default - in this case there is virtually no chance > of a floor change to occur. > > Is a FIR packet also sent when someone joins a conference or is it > currently only sent on floor change events? > > Best > Hristo > > > On Sat, Mar 29, 2014 at 3:11 PM, Anthony Minessale < > anthony.minessale(a)gmail.com> wrote: > >> Fir is sent explicitly no settings req >> On Mar 28, 2014 5:15 PM, "Seven Du" <dujinfang(a)gmail.com> wrote: >> >>> AFAIK FS has that so a jira would help >>> 2014年3月29日上午4:57于 "Hristo Trendev" <dist.lists(a)gmail.com>写道： >>> >>>> Hi Brian, >>>> >>>> It's the latest v1.4.beta branch (555ef59) compiled on Debian 7.4. I >>>> can test with the master branch on Monday if needed. However, I am not >>>> really sure if the functionality to request a keyframe via RTCP FIR is >>>> even present in conference/freeswitch. In case it is indeed missing, >>>> then it will at least explain the behavior I see, otherwise I've probably >>>> missed something while testing. >>>> >>>> Best, >>>> Hristo >>>> >>>> >>>> On Fri, Mar 28, 2014 at 7:35 PM, Brian West <brian(a)freeswitch.org>wrote: >>>> >>>>> Please make sure you’re using the latest code, care to say what Rev >>>>> you’re on? >>>>> -- >>>>> Brian West >>>>> brian(a)freeswitch.org >>>>> FreeSWITCH Solutions, LLC >>>>> PO BOX 2531 >>>>> Brookfield, WI 53008-2531 >>>>> Twitter: @FreeSWITCH , @briankwest >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST >>>>> iNUM: +883 5100 1420 9001 >>>>> ISN: 410*543 >>>>> Skype:briankwest >>>>> PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED) >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> On Mar 28, 2014, at 10:13 AM, Hristo Trendev <dist.lists(a)gmail.com> >>>>> wrote: >>>>> >>>>> > Hi, >>>>> > >>>>> > I've been testing a bit with WebRTC video using Chrome on Mac and a >>>>> Freeswitch conference. The problem that I have is that the WebRTC client >>>>> (Chrome) sends video keyframes every 3000 frames, which means that when a >>>>> new participant joins the conference it can be some time before he/she gets >>>>> a keyframe. The same problem occurs when the floor changes and the video >>>>> stream source of the conference is switched to another participant, whose >>>>> WebRTC client won't necessarily send a keyframe on floor change. >>>>> > >>>>> > As far as I understood after reading through a couple of Chrome >>>>> tickets, a keyframe can be requested by the receiving party - in this case >>>>> a Freeswitch conference, via RTCP FIR (Full Intra Request) or PIL (Picture >>>>> Loss Indication) packet. >>>>> > >>>>> > Ticket FS-5596 mentions FIR in its description, so it seems as if FS >>>>> already has support for FIR (or maybe I got it all wrong and the case >>>>> described in this ticket refers to RTCP packets, which are simply being >>>>> proxied by FS). Additionally FS indicates in the SDP that it supports FIR. >>>>> > >>>>> > BTW, I am not even sure if I have configured the RTCP in Freeswitch >>>>> correctly. What I did was to uncommented "rtcp-video-interval-msec" and >>>>> "rtcp-audio-interval-msec" in the sofia SIP profile. However, I still see >>>>> in Chrome's log: >>>>> > webrtc: (rtp_rtcp_impl.cc:225): Process: Timeout: No RTCP RR >>>>> received. >>>>> > webrtc: (rtp_rtcp_impl.cc:227): Process: Timeout: No increase in >>>>> RTCP RR extended highest sequence number. >>>>> > >>>>> > The real question is if the conference app can be configured/patched >>>>> to send an RTCP FIR packet to the member who has the floor whenever a new >>>>> participant joins the conference or in case the floor itself just moved to >>>>> someone else. >>>>> > >>>>> > Has anyone tested something similar? >>>>> > >>>>> > BR >>>>> > Hristo >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting(a)freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>>> http://www.cudatel.com >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> 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 >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting(a)freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>>> http://www.cudatel.com >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> 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 >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting(a)freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >>> http://www.cudatel.com >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> 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 >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting(a)freeswitch.org >> http://www.freeswitchsolutions.com >> >> FreeSWITCH-powered IP PBX: The CudaTel Communication Server >> http://www.cudatel.com >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> 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 >> >> > > > ---------- Forwarded message ---------- > From: Sean Devoy <sdevoy(a)bizfocused.com> > To: "FreeSWITCH-users(a)lists.freeswitch.org" < > FreeSWITCH-users(a)lists.freeswitch.org> > Cc: > Date: Mon, 31 Mar 2014 16:37:50 +0000 > Subject: [Freeswitch-users] Windows Env - Can't Build ManagedEslTest.2010 > or 2012 > > Hi, > > > > Can someone look at this pastebin of build errors on VS2012 and tell me > what I am missing? > > http://pastebin.freeswitch.org/22267 > > > > > > I did a new GIT and built the entire project on Windows with now errors. > I really just want to build code like Program.cs in this sample to talk to > ESL on my server. > > > > The problem appears to be the external cdecl bs CSharp_Entern flags on > ESL.LIB. But I don’t know how to fix it. > > > > Thanks in advance. > > > > Sean Devoy > > _______________________________________________ > 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 > >

1 0

Audio(RTP) Stops after first message played
by Gopalakrishnan N 31 Mar '14

31 Mar '14

Hi, I have a setup as per the following, Server A - FreeSWITCH (Location A) Server B - Asterisk (Location A) Server C - Asterisk (Location B) Two Asterisk servers are trunked with FreeSWITCH. In FreeSWITCH am establishing Conference via a Javascript. >From Server B (Asterisk) if I initiate the call, it works absolutely fine by entering into the conference room. >From Server C (Asterisk) if I initiate the call, am able to hear the first word (Please) from the message "Please enter your conference number" and then its blank. The network connection between Location A and Location B is MPLS. My dialplan is pasted here http://pastebin.freeswitch.org/22228 Comments would be much appreciated. Thanks.

2 5

freeswitch-meta-all broken?
by Stanislav Sinyagin 31 Mar '14

31 Mar '14

I'm installing FreeSWITCH from debian packages on a fresh machine, i686 wheezy. apt-get install -y freeswitch-meta-all has resulted in just 3 packages installed: # aptitude search freeswitch | egrep '^i' i A freeswitch - Cross-Platform Scalable Multi-Protocol Sof i freeswitch-meta-all - Cross-Platform Scalable Multi-Protocol Sof i A libfreeswitch1 - Cross-Platform Scalable Multi-Protocol Sof Previously freeswitch-meta-all was installing the whole lot.

1 0

set the Mark bit in the live audio RTP stream
by Moishe Grunstein 31 Mar '14

31 Mar '14

Hi All, How can tell Freeswitch to set the mark bit when going from early media to media? I am having a one way audio issue for the 1st 20 seconds with calls originating from a Cyberdata door box. FS Version 1.2.21 Below is my response from Cyberdata Support. Hello Moishe, I really appreciate your time working with me today attempting to get the intercom registered! I know I said I would just touch bases with you tomorrow, but I think I have resolved the issue on my end after talking to our engineers and also figured out what is happening with the intercoms experiencing delayed audio at the intercom's speaker during calls. Recently, we resolved a similar issue with a FreeSWITCH provider who reported the same symptoms with their intercoms. Their FreeSWITCH server also transitions from early media (where the ringback tone plays to the calling device) to live audio just like your FreeSWITCH implementation. We learned that our device wasn't properly identifying the Mark bit in the RTP stream and synchronizing the audio stream. A fix was implemented and the audio delay was resolved with v9.1.1b01. In your FreeSWITCH implementation, we are seeing the same symptoms but in your case FreeSWITCH does not set the Mark bit in the live audio RTP stream. It does not change the SSRC either. Instead, the timestamp jumps which causes the intercom to start recording missed timestamps until it hits a threshold after about 20 seconds, and after this point it resynchronizes the audio stream and the intercom starts playing it out of the speaker. In the attached trace you provided to us, frame 198 shows the transition from early media to live audio. The marker bit is not set. I also included a .jpg highlighting frame 184 and 198 for your reference. You will see the timestamp jumps but the SSRC/sequence numbers don't change and the mark bit is not set to indicate a new stream. Based on this information, we are recommending you consult with your FreeSWITCH engineers and advise setting the Mark bit in the RTP stream when transitioning from early media to live audio since the SSRC and sequence numbers of the audio stream do not change. Any calls with the Mark bit set in the new audio stream when switching from early media to live audio should play immediately without delay. However, for calls with new audio streams without a Mark bit set, audio will be delayed until this is resolved in FreeSWITCH. I hope this information is helpful, Moishe! Could you please keep us posted? We would be happy to get on a conference call with you if it will help. We know of another FreeSWITCH reseller in your area who had a similar issue so you are not alone. :) Many thanks, Christina Thanks, Moishe Grunstein Tornado Computer Systems, Inc. 212.400.7650 888.IPPBX.US Service Request Email: support(a)nysolutions.com<mailto:support@nysolutions.com> Polycom Certified VAR Microsoft Small Business Specialist, Cisco SMB Select Certified [cid:image001.jpg@01C72F94.9EE45D60]<http://www.nysolutions.com/> Computer Networking * Managed Services * IP Video Surveillance * Network Assessments * Web Solutions * Voice over IP * Disaster Recovery * Network Security * Site Surveys * CMS

3 5

FreeSWITCH changing max_sessions automatically
by Muhammad Naseer Bhatti 31 Mar '14

31 Mar '14

Just have noticed something new today. Testing a local instance on CentOS 6.0 with FreeSWITCH Version 1.2.23~64bit. FreeSWITCH seems to be changing the max_sessions param automatically. This happens when typically I bridge around 1000 sessions and whatever value I set to sps and max_sessions, it comes down. UP 0 years, 0 days, 0 hours, 37 minutes, 54 seconds, 362 milliseconds, 294 microseconds FreeSWITCH (Version 1.2.23 64bit) is ready 74716 session(s) since startup 976 session(s) - peak 1022, last 5min 985 112 session(s) per Sec out of max 400, peak 9053, last 5min 317 <- BTW This peak was never tried, not sure why it says 9053 5000 session(s) max min idle cpu 0.00/55.00 Current Stack Size/Max 240K/8192K ======= followed by ======= switch_core_session.c:2288 Over Session Limit! 1013 mod_sofia.c:4884 Error Creating Session ======= And then ======= freeswitch@internal> status UP 0 years, 0 days, 0 hours, 38 minutes, 35 seconds, 372 milliseconds, 831 microseconds FreeSWITCH (Version 1.2.23 64bit) is ready 76231 session(s) since startup 560 session(s) - peak 1023, last 5min 1023 0 session(s) per Sec out of max 400, peak 9053, last 5min 317 1013 session(s) max min idle cpu 0.00/99.00 Current Stack Size/Max 240K/8192K Is this something changed recently? Or some sort of server meltdown protection mechanism being forked :) -- Thanks, Muhammad Naseer Bhatti

5 8

freeswitch error message
by 曾雅醇 31 Mar '14

31 Mar '14

Hello, I am new to freeswitch near a user, I installed a sangoma a108 card, can successfully inbound, outbound, but an error message will be 404,503, and ask the question, what is the solution, thank you !!

2 1

How to get SIP response string (not header)
by Adam Raszynski 31 Mar '14

31 Mar '14

Hi All, I'm looking for the possibility to get SIP response string from B leg, do some transformations and pass it back to the A leg. Example response from B leg: SIP/2.0 500 Trunk Prefix Mismatch When FS receives that response from B, it ignores text message "Trunk Prefix Mismatch" and send to the A leg reply with it's own message SIP/2.0 500 Internal Server Error How to get that response or pass it unchanged to the A leg? Best Regards

1 0

Re: [Freeswitch-users] Bypass_media
by Andre Demattia 30 Mar '14

30 Mar '14

I'm using the last version that just came out. 1.2.23 I'll get the log. Thanks Andre -----Original Message----- From: "Steven Ayre" <steveayre(a)gmail.com> Sent: ‎3/‎29/‎2014 10:55 PM To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Bypass_media Can you confirm the version you're running and supply a debug-level log of the call failing? On 29 March 2014 15:28, Andre <andretodd(a)verizon.net> wrote: > The code was just a sample code but > > The way I know it didn't work is I call a command on answer > > a.ExecuteString(String.Format("uuid_media {0} ", s.uuid)); > > > > This will put the call in full media. Since by default its already full > media and the code on the bridge says bypass media (in my real application) > it should return a OK, but it returns an error that it's already full media. > > > > Andre > > > > > > *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: > freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, March 29, 2014 3:21 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Bypass_media > > > > How do you know it didn't work? > > It looks like you're bridging to the same gateway each time... > > > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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 > >

1 0

Bypass_media
by Andre 30 Mar '14

30 Mar '14

I need a suggestion on how to bypass_media based on the gateway I'm using. Say we have 3 bridge's. One has to be full media but the other 2 can be bypass. If 1 fails, 2 will be the failover but since you have to set the media on leg a how do I change it to full media? Hangup after bridge Bridge 1 bypass_media = true Bridge 2 bypass_media = false Bridge 3 bypass_media = true

3 6

Using proxy with SIP trunk
by Jose Miguel Sucasas Mejuto 29 Mar '14

29 Mar '14

Hi all, I’m trying to set my SIP provider on my fresh installation of FreeSWITCH. My SIP provider needs to use a proxy and authenticated username, something like this: USERID: +3467******* DOMAIN: ims.vodafone.es AUTH. NAME: 214**************(a)ims.mnc001.mcc214.3gppnetwork.org PROXY: 62.87.93.90:5090 I have used this configuration in directory/default/example.com.xml: <include> <user id="$${default_provider}"> <gateways> <gateway name="$${default_provider}">  <param name="username" value="214**************(a)ims.mnc001.mcc214.3gppnetwork.org"/> <param name="password" value="******"/> <param name="from-user" value="+34672099215"/> <param name="from-domain" value="ims.vodafone.es"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> <param name="extension" value="1001"/> <param name="contact-params" value="domain_name=$${domain}"/> <param name="context" value="public"/> <!—Added by me --> <param name="realm" value="ims.vodafone.es"/> <param name="proxy" value="62.87.93.90:5090"/>  </gateway> </gateways> <params> <param name="password" value="******"/> </params> </user> </include> but this configuration doesn’t work: The registration is OK: freeswitch@debrix> sofia status Name Type Data State ================================================================================================= 192.168.1.20 alias internal ALIASED internal profile sip:mod_sofia@192.168.1.20:5060 RUNNING (0) external profile sip:mod_sofia@192.168.1.20:5080 RUNNING (0) external::ims.vodafone.es gateway sip: 214**************@@ims.mnc001.mcc214.3gppnetwork.org@62.87.93.90:5090 REGED ================================================================================================= but when I try to make a call, I see the following: freeswitch@debrix> originate sofia/gateway/ims.vodafone.es/+34677****** &playback("hello world") 2014-03-29 20:06:40.634625 [NOTICE] switch_channel.c:1050 New Channel sofia/external/+34677****** [8b2f3c51-c95a-4b12-a82a-bcbb11f8626a] send 1046 bytes to udp/[62.87.93.90]:5090 at 19:06:40.981203: ------------------------------------------------------------------------ INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090> Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE Contact: <sip:gw+ims.vodafone.es@47.62.8.53:5080;domain_name=192.168.1.20;transport=udp;gw=ims.vodafone.es> User-Agent: FreeSWITCH-mod_sofia/1.2.23-1~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 197 X-FS-Support: update_display,send_info Remote-Party-ID: <sip:0000000000@ims.vodafone.es>;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1396103996 1396103997 IN IP4 47.62.8.53 s=FreeSWITCH c=IN IP4 47.62.8.53 t=0 0 m=audio 16004 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ recv 289 bytes from udp/[62.87.93.90]:5090 at 19:06:40.985906: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080 From: "" <sip:+34672099215@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090> Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE ------------------------------------------------------------------------ recv 508 bytes from udp/[62.87.93.90]:5090 at 19:07:13.017222: ------------------------------------------------------------------------ SIP/2.0 487 Request Terminated Via: SIP/2.0/UDP 47.62.8.53:5080;received=47.62.8.53;branch=z9hG4bKQvryjNK4mDNeH;rport=5080 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090>;tag=9888 Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 INVITE Content-Length: 0 Contact: <sip:172.21.32.22:5070> P-Charging-Vector: icid-value=eb15701b0bdd73150b21ba59b77106f P-Charging-Function-Addresses: ccf="aaa://mm.ims.vodafone.es:3868;transport=tcp" ------------------------------------------------------------------------ send 334 bytes to udp/[62.87.93.90]:5090 at 19:07:13.017683: ------------------------------------------------------------------------ ACK sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090>;tag=9888 Call-ID: 1a188752-3218-1232-e79c-94de8085f14a CSeq: 57723936 ACK Content-Length: 0 ------------------------------------------------------------------------ 2014-03-29 20:07:13.014627 [NOTICE] sofia.c:6659 Hangup sofia/external/+34677****** [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] -ERR ORIGINATOR_CANCEL 2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1577 Session 5 (sofia/external/+34677******) Ended 2014-03-29 20:07:13.014627 [NOTICE] switch_core_session.c:1581 Close Channel sofia/external/+34677****** [CS_DESTROY] I think that the error ( SIP/2.0 487 Request Terminated) is in the to-field of the header: INVITE sip:+34677******@62.87.93.90:5090 SIP/2.0 Via: SIP/2.0/UDP 47.62.8.53:5080;rport;branch=z9hG4bKQvryjNK4mDNeH Max-Forwards: 70 From: "" <sip:+34672******@ims.vodafone.es>;tag=pDBN9taHSHX3p To: <sip:+34677******@62.87.93.90:5090> FreeSWITCH writes the proxy instead of the realm domain (ims.vodafone.es). Do you know how can I solve this problem? Thanks in advance, Regards

1 0