November 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Not recognizing voice in mod_pocketsphinx
by Thangappan.M 30 Jan '12

30 Jan '12

Dear all, I have installed the mod_pocketsphinx module successfully and enabled with FreeSWITCH. Copied the relevant files for pizza application and placed into appropriate directory. Made a call to "74992" and able to hear the voice. When I said "TAKEOUT' it was not recognizing my words. In the freeswitch CLI it shown the following messages. EXECUTE sofia/internal/1000(a)192.168.1.222 javascript(ps_pizza.js) 2010-09-06 13:18:48.023673 [DEBUG] sofia_glue.c:2594 AUDIO RTP [sofia/internal/1000(a)192.168.1.222] 192.168.1.222 port 32590 -> 192.168.6.60 port 8000 codec: 8 ms: 20 2010-09-06 13:18:48.024705 [DEBUG] switch_rtp.c:1182 Starting timer [soft] 160 bytes per 20ms 2010-09-06 13:18:48.025710 [DEBUG] sofia_glue.c:2774 Set 2833 dtmf send payload to 101 2010-09-06 13:18:48.026799 [DEBUG] sofia_glue.c:2779 Set 2833 dtmf receive payload to 101 2010-09-06 13:18:48.026799 [DEBUG] mod_sofia.c:636 Local SDP sofia/internal/ 1000(a)192.168.1.222: v=0 o=FreeSWITCH 1283726738 1283726739 IN IP4 192.168.1.222 s=FreeSWITCH c=IN IP4 192.168.1.222 t=0 0 m=audio 32590 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2010-09-06 13:18:48.028235 [DEBUG] switch_core_session.c:641 Send signal sofia/internal/1000(a)192.168.1.222 [BREAK] 2010-09-06 13:18:48.028235 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [completed][200] 2010-09-06 13:18:48.029256 [NOTICE] mod_spidermonkey.c:2066 Channel [sofia/internal/1000(a)192.168.1.222] has been answered 2010-09-06 13:18:48.030269 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:48.123555 [DEBUG] sofia.c:4153 Channel sofia/internal/ 1000(a)192.168.1.222 entering state [ready][200] 2010-09-06 13:18:48.171705 [DEBUG] switch_rtp.c:2066 Correct ip/port confirmed. 2010-09-06 13:18:50.212417 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(pocketsphinx pizza_order undefined) 2010-09-06 13:18:51.768030 [DEBUG] switch_core_media_bug.c:360 Attaching BUG to sofia/internal/1000(a)192.168.1.222 EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:18:51.774074 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms 2010-09-06 13:18:54.492023 [DEBUG] switch_ivr_play_say.c:1444 done playing file EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) EXECUTE sofia/internal/1000(a)192.168.1.222 detect_speech(resume) 2010-09-06 13:19:01.534474 [DEBUG] switch_ivr_play_say.c:1152 Codec Activated L16@8000hz 1 channels 20ms Please help me..... -- Regards, Thangappan.M

3 3

audo sync issues with record_session to mp3
by Tom Parrott 13 Jan '12

13 Jan '12

Hi, We are running a cluster of 3 SIP servers all running Freeswitch. Each call is recorded using the record_session application to record to mp3 files using mod_shout. Longer calls, after about 10 minutes start to introduce sync issues between the A-leg and the B-leg. We are running record_session on the A-leg, and it seems to get ahead of the B-leg. For example the caller on the A-leg will be heard to answer a question whilst the person on the B-leg is asking it. I am aware that recording to .mp3 can use additional resources, but these are quad core servers with 4GB of ram and they are only doing around 10 concurrent calls each. Additionally I have seen others experiencing these issues with Wav files too: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-July/060107.html I am using the git head version from Tue 11 Oct 2011. I have also tried turning off enable_file_write_buffering on the A-leg however this makes no difference either. Does anyone have any suggestions? Thanks

6 6

Overlap dialing problem
by Štefan Čudai 10 Jan '12

10 Jan '12

Hi, does FreeSwitch support for overlap dialing or did somebody try to use it? I am using Sangoma card A102 under Windows 7 with FreeTDM. I have configured my freetdm.conf.xml with this two lines: <param name="overlap" value="yes"/> <param name="min_digits" value="0"/> But when I'm dialing number (which has 14 digits) I hear disconnect tone after dialing first 5 digits. I think that reason is because FreeTDM module is waiting number with minimum digits equals to "0" (see my min_digits). I have also try set min_digits to 20 and then dial same number with 14 digits but result was same. So, I looked to log file and found that FreeTDM module got in thist case STATUS CONFIRM message but he was still waiting for next digits and his time expired. Can you help me please? Thanks a lot. Stefan

3 2

How to troubleshoot "503 Maximum Calls In Progress"?
by Chris Fowler 05 Jan '12

05 Jan '12

Hi, On our Production FreeSWITCH box: FreeSWITCH Version 1.0.head (git-4c435ec 2011-03-14 11-54-08 -0500) UP 0 years, 14 days, 6 hours, 28 minutes, 44 seconds, 474 milliseconds, 751 microseconds 12519 session(s) since startup 2 session(s) 0/30 1000 session(s) max min idle cpu 0.00/100.00 At 14:30:37 a reloadxml command was issued. At 14:30:43 the box started rejecting all calls with 503 error. At 14:38 it started working again. Looking at the code I see three conditions can trigger this behavior: if (sess_count >= sess_max || !sofia_test_pflag(profile, PFLAG_RUNNING) || !switch_core_ready()) { nua_respond(nh, 503, "Maximum Calls In Progress", SIPTAG_RETRY_AFTER_STR("300"), TAG_END()); switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_CRIT, "No more sessions allowed at this time.\n"); goto done; } Logs show it wasn't lack of available sessions. How can sofia_test_pflag(profile, PFLAG_RUNNING) or switch_core_ready() fail and busy the system? Thoughts? Thx, Chris.

3 2

LUA transfer result:0
by Anestis Mavro 24 Dec '11

24 Dec '11

Hi, I have tried two different ways of transfer in LUA: session:transfer(number,"XML","default") And Session:execute("transfer","12345 XML default") And I can't make it working. I get "switch_cpp.cpp:797 transfer result: 0" The same transfer from within the dialplan works fine The call is not answered before the transfer. Does anybody have an idea? Thanks Anestis

4 4

Feedback
by Chad Vogel 23 Dec '11

23 Dec '11

Hello, I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users. Here are some of the changes we have made: * We converted FS core to C++ and having it compiling to .NET 4.0 CLI * FS now runs inside its own namespace * We replace FS_CLI with a PowerShell shell app * Supports SQL Server support via the Native SQL Server Client API * Support for windows clustering (up to 32 node active/active cluster) * Ties more directly into the Win32 API and has less reliance on open source Libs * Replaces OpenSSL sockets with Windows encrypted sockets * Added windows performance monitoring * No longer need to use mod_managed for managed modules * Memory management relies on .Net garbage collection * Added support for Microsoft Speech * VoiceXML 3.0 and SCXML support * No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages) * Fixed RTP clock timing issues in virtualized environments * New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

18 27

FreeSWITCH cluster with IM/Presence
by Even André Fiskvik 22 Dec '11

22 Dec '11

Hi. I currently have a project where I'm researching how to establish a clustered and scalable platform with: - registration - presence/blf between eyebeam/bria and Linksys SPA962 clients - IM between eyebeam/bria clients. - calling between the clients The first scenario I want to test is with two FreeSWITCH servers and shared ODBC db running in a "cluster". How the registrations and calls etc. are balanced between them is not decided yet, but I'm thinking that a UltraMonkey setup or DNS SRV records could to the trick here? The main challenge in this setup is sharing presence, and doing call-setup and IM between two FreeSWITCH servers (which can be scaled by adding n-nodes). I'm interested in user's that can confirm wether this solution is possible or not with FreeSWITCH and it's current state, and what it would take to make something like this work. If there's any part of FS that is lacking some implementation to make such a scenario work, we are interested in providing the funding to support those features. I've seen that the core db has added support for ODBC, but I'm not sure wether this is something that would be helpful in this scenario or not. This is only conceptual right now, so I'm interested in paying someone that could help me configure the two FS servers to support this scenario on an hourly basis to get a quick proof-of-concept up and running. The proof-of-concept has to be completed by March 19th, after which we will evaluate what technologies and setup we want to commit to. Some other variations I'm concidering is Opensips/Kamailio -> Asterisk, Opensips/Kamailio -> FreeSWITCH, Opensips/Kamailio -> Asterisk and Openfire + Asterisk-im plugin for IM/presence. I'm also open to other suggestions, but I'd like this topic to stay on how FreeSWITCH could facilitate such a platform. Best regards, Even André Fiskvik CTO - Oyatel AS

2 1

Sofia late-negotiation on re-INVITE (codec-modification)
by Visciano Giovanni 21 Dec '11

21 Dec '11

Basic call SIP vs SIP with: <param name="inbound-late-negotiation" value="true"/> <action application="set" data="inherit_codec=true"/> <param name="renegotiate-codec-on-reinvite" value="true"/> This help to avoid FS transcoding, the SIP endpoints negotiate codec end-to-end. FS still in the media path but no transcoding applied. INVITE [pcma,g729] ---> FS ---> INVITE [pcma,g729] 200 OK [pcma] <--- FS ---> 200 OK [pcma] If a SIP ep send a re-INVITE for codec modification, sofia negotiate the new codec on that leg without "asking" the remote party. This result in transcoding and since I have mod_g729 only in passthrough mode the call is released by the core. re-INVITE [g729] ---> FS 200 OK [g729] <--- FS ************* It seems that "late_negotiation + inherit_codec" doesn't apply to a codec modification. Do I miss some configuration options in the SIP profile? (Note: I cannot use proxy-media/no-media mode in my configuration) ************* Regards Giovanni Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. -----------------------------------------------------------------------------------------------------

4 8

Linksys PAP2-T and echo
by Boris Kovalenko 21 Dec '11

21 Dec '11

Hello! I'm wondering if it is possible to solve echo with Linksys PAP2-T devices. Does anybody here uses it? I read many articles and still can't understand how to eleminate echo :( I read http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found that Asterisk has its own EC. Does Freeswitch has it too? I found this article http://docs.freeswitch.org/echo_can_page.html but can't understand - has Freeswitch EC or not? If yes - how to turn it on and off? -- Regards, Boris

5 13

Call/Transaction Does Not Exist
by Joe Flemmings 21 Dec '11

21 Dec '11

When i call from one extension and hangup, the other call does not hangup and continues ringing. Please see log bellow. Any ideas? U 10.10.78.24:55479 -> 10.10.78.36:5060 INVITE sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@10.10.78.36;user=phone>..CSeq: 1 INVITE..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MES SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000.. # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: < sip:213@10.10.78.36;user=phone> ..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 1 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Content-Length: 0.... # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@6 6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 1 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Accept: appl ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..Allow-Events: talk, hold, prese nce, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Proxy-Authenticate: Digest realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac ", algorithm=MD5, qop="auth"..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 ACK sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <si p:213@10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 ACK..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL, OPTION S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Max-Forwards: 70..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 INVITE sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@10.10.78.36;user=phone>..CSeq: 2 INVITE..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24:52732>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MES SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: en..Supported: 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz ation: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000001, uri="sip:213@10.10.78.36:5060 ;user=phone", response="82a75edb7ba97bfe7dea585684701d0c", algorithm=MD5..Max-Forwards: 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000.. # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: < sip:213@10.10.78.36;user=phone> ..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 -0500..Content-Length: 0.... # U 10.10.78.36:5060 -> 10.10.78.24:52783 INVITE sip:pbxjker_213@10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: 69..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81 rv8Fp..To: <sip:pbxjker_213@10.10.78.24:52783>..Call-ID: 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact: < sip:mod_sofia@10.10.78.36:5060>..User-Agent: FreeSWITCH-mod_sofia/1.0. head-git-1086cba 2011-05-23 22-51-43 -0500..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces.. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Content-Type: application/sdp..Content-Disposition: session.. Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH 1318333156 1318333157 IN IP4 10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio 29124 RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98 bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320 00..a=fmtp:18 annexb=no..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-16..a=ptime:20.. # U 10.10.78.24:55516 -> 10.10.78.36:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two" <sip:pbxjker_213@99.65.2 17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID: 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: < sip:pbxjker_213@10.10.78.24:52783>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/3.2 .2.0477..Accept-Language: en..Content-Length: 0.... # U 10.10.78.24:55516 -> 10.10.78.36:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=9KgeNX81rv8Fp..To: "Polycom Two" <sip:pbxjker_213@99.65. 217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID: 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: < sip:pbxjker_213@10.10.78.24:52783>..User-Agent: PolycomSoundPointIP-SPIP_320-UA/3. 2.2.0477..Allow-Events: talk,hold,conference..Accept-Language: en..Content-Length: 0.... # U 10.10.78.36:5060 -> 10.10.78.24:52732 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" <sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: < sip:213@10.10.78.36;user=phone >;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 INVITE..Contact: <sip:213@10.10.78.36:5060;transport=udp>..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2 3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, precondition, path, replaces..A llow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" <pbxjker _213>;party=calling;privacy=off;screen=no.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCR IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-462 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, uri="sip:213@10.10.78.36:5060;user=phone", response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards: 70..Content-Length: 0.. .. # U 10.10.78.36:5060 -> 10.10.78.24:5060 SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@66.11 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 CANCEL..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCR IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-462 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, uri="sip:213@10.10.78.36:5060;user=phone", response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards: 70..Content-Length: 0.. .. # U 10.10.78.36:5060 -> 10.10.78.24:5060 SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@66.11 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 CANCEL..Content-Length: 0.... # U 10.10.78.24:55479 -> 10.10.78.36:5060 CANCEL sip:213@10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:2 13(a)10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..Contact: < sip:pbxjker_2564@10.10.78.24>..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCR IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: Digest username="pbxjker_2564", realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-462 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, uri="sip:213@10.10.78.36:5060;user=phone", response="d47848babd8c396d8da15145503875d0", algorithm=MD5..Max-Forwards: 70..Content-Length: 0.. .. # U 10.10.78.36:5060 -> 10.10.78.24:5060 SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" < sip:pbxjker_2564@10.10.78.36>;tag=F5263502-30CB1633..To: <sip:213@66.11 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: 424791be-fbadec9f-83bb2654@192.168.1.72..CSeq: 2 CANCEL..Content-Length: 0.... #

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