August 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

using sipp to test call transfers
by Raimund Sacherer 25 Aug '09

25 Aug '09

Hi List, i have some scripts to test our lab: I have scripts to create sipp instances which act as individual agents, which log on, take a random amount of calls, log off, wait a bit, log on again, etc. I have scripts to generate call traffic for our queues to saturate them But what I have difficulties is how to create a good scenario xml file for call transfers? Has anybody here a sample for me to sink my teeth in? thanks and best regards -- Raimund Sacherer - RunSolutions Open Source It Consulting - Parc Bit - Centro Empresarial Son Espanyol Edificio Estel - Local 3D 07121 - Palma de Mallorca Baleares

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MFC-R2 support for FreeSWITCH
by Moises Silva 25 Aug '09

25 Aug '09

So, I finally took some days to put up OpenR2 working with OpenZAP, which means FreeSWITCH now supports MFC-R2 for all variants that OpenR2 has support for. Including Mexico, Brazil, Argentina and others. The stack has been used by Asterisk starting with Asterisk 1.6.2 so I feel it covers most countries that users may be interested in, support for new variants will be added on-demand only (in any case users can always tweak the advanced configuration file to create their own variants as a last resort). I created a wiki page to illustrate the basic setup: http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2 <http://wiki.freeswitch.org/wiki/OpenZAP_OpenR2>Now is time for testing. I just did minimal testing on my development environment, no serious testing, and I know that some stuff is not working at this point (I had some issues with variable length DNIS and ANI) which should be fixed soon. If anyone around happens to have an R2 link and wants to test R2 support in OpenZAP, I can give them a hand with the configuration and any issues you may find. You can find me on IRC at #freeswitch, #freeswitch-dev and #openzap as moy. -- Moises Silva Software Developer Sangoma Technologies Inc. | 50 McIntosh Drive, Suite 120, Markham ON L3R 9T3 Canada t. 1 905 474 1990 x 128 | e. moy(a)sangoma.com

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Yet another question about A500 + FS
by Vassil Panayotov 25 Aug '09

25 Aug '09

Hi, I managed to get our A500 running with FreeSWITCH 1.0.4 stable using wanpipe 3.4.4 drivers. But now I have another problem... I want to originate calls through event socket, and I only want to receive ANSWERED(+OK) reply when the user actually answers. Now the situation is: ==================================== originate openzap/1/a/123456 023 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT: CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] Ci=[0000000000] 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082] 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer OpenZAP/1:1/123456! API CALL [originate(openzap/1/a/123456 023)] output: +OK f8fca2be-8fa7-11de-9076-511e29dfc082 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer OpenZAP/1:1/123456 to XML[023@default] freeswitch@emo-voip> 2009-08-23 08:44:06.743475 [INFO] mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel [OpenZAP/1:1/123456] has been answered 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING] 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N): CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2 (OpenZAP/1:1/123456) Ended 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close Channel OpenZAP/1:1/123456 [CS_DESTROY] ==================================== Extension 023 is an IVR. As you can see FreeSWITCH answers the call (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel [OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5). So Sangoma drivers/daemons report the events correctly. How can I set FreeSWITCH to answer after receiving RX EVENT (N): CALL_ANSWERED from the driver? Thank you, V. Panayotov

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compile freeswitch, lua, luasql for 64-bit arch
by Tapan Parikh 25 Aug '09

25 Aug '09

Hi - Does anyone know how to compile freeswitch for a 64-bit architecture on mac os x? MySQL on MAC OS X only comes pre-compiled for 64-bit. I compiled luasql against this libmysqlclient.dylib. Now when I call lua from freeswitch I get: 2009-08-24 21:06:44.370439 [ERR] mod_lua.cpp:182 error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': dlopen(/usr/local/lib/lua/5.1/luasql/mysql.so, 2): no suitable image found. Did find: /usr/local/lib/lua/5.1/luasql/mysql.so: mach-o, but wrong architecture stack traceback: [C]: ? [C]: in function 'require' /usr/local/freeswitch/scripts/otalo.lua:3: in main chun Any pointers?

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Freeswitch vs. Asterisk speed on ARM
by Rogelio Perez 24 Aug '09

24 Aug '09

Hi Everyone, I'm working on a PBX project for the Sheevaplug ARM based computer, with the following specs: CPU 1.2 GHz, 512MB DDR2, no FPU. So far I've found a big difference between Freeswitch and Asterisk performance times. This is a comparison of the time it takes them to perform different actions: startup Freeswitch: 3 min. startup Asterisk: 2 sec. call extension Freeswitch: 6 sec. call extension Asterisk: 0 sec. shutdown Freeswitch: 6.5 sec shutdown Asterisk: 0 sec. reload config Freeswitch: 1 sec. reload config Asterisk: 1 sec. Both were built from sources natively (no cross-compiling), and they use the default startup configurations. I have managed to lower the Freeswitch times by disabling most of the modules and recompiling, but it is still far away from Asterisk (i.e. FS startup time 2.5 min). 1. Is there any way to further improve Freeswitch performance for the ARM architecture? 2. Can this be related to the lack of a FPU (the Sheevalug emulates the floating point operations). 3. On the startup I see this error repeated many times: [ERR] switch_core_sqldb.c:95 SQL ERR [database is locked]. Can this be related? Thanks, Rogelio Perez

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Couple of questions
by Matt Riddell 24 Aug '09

24 Aug '09

Hi, I don't see how I can read some responses to command using esl. I.E. esl_send_recv(&handle, "api show calls count\n\n"); and printf("Header Test %s\n", esl_event_get_header(event, "API-Command")); printf("Body Test %s\n", esl_event_get_body(event)); the header details are returned. The body is null. Also, I can originate a call and set the account code for it, but how do I get a list of calls with their account codes? Do I get a list of calls then go through them one by one and get the variables for those calls by uuid? Does anyone have any documentation for the esl api? Even if I could read some comments from a usage of it would be useful. -- Cheers, Matt Riddell Director _______________________________________________ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems)

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pass-thru issue (G729)
by Javier Aristizábal 24 Aug '09

24 Aug '09

Hello All, I'm having an issue with g.729 pass-thru. On an older 1.0.3 install it's all good, but on a newer machine, more cores, etc. the quality of voice on g.729 is very poor. Anyone here dealt with something like this? With all other codecs are perfect, even in pass-thru... Thanks in ddvance Javier

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Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand ACK message
by Tihomir Culjaga 24 Aug '09

24 Aug '09

Hello Brian, it doesn't work .. tried this today as well: freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 20:28:09.367300: ------------------------------------------------------------------------ INVITE sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403(a)10.4.4.252> > To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 Call-ID: 1-7019(a)10.4.4.252 CSeq: 1 INVITE Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 131 v=0 o=user1 53655765 2353687637 IN IP4 10.4.4.252 s=- c=IN IP4 10.4.4.252 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ send 328 bytes to udp/[10.4.4.252]:5060 at 20:28:09.367634: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > Call-ID: 1-7019(a)10.4.4.252 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.371759: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >;tag=ygQBtp6QpKtcD Call-ID: 1-7019(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported 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 ------------------------------------------------------------------------ recv 401 bytes from udp/[10.4.4.252]:5060 at 20:28:09.371989: ------------------------------------------------------------------------ ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-7019-1-3;rport To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >;tag=ygQBtp6QpKtcD From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 Call-ID: 1-7019(a)10.4.4.252 CSeq: 1 ACK Contact: sip:sipp@10.4.4.252:5060 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[10.4.4.252]:5060 at 20:28:09.873045: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-7019-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >;tag=ygQBtp6QpKtcD Call-ID: 1-7019(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported 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 This thing is driving me crazy, pls help. T. > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 24 Aug 2009 14:15:40 -0500 > Subject: Re: [Freeswitch-users] SIPp issues - seems FS doesn't understand > ACK message > In your scenario you need to add [peer_tag_param] at the end of the to on > the Ack. > > /b > > On Aug 24, 2009, at 2:03 PM, Tihomir Culjaga wrote: > > >> ------------------------------------------------------------------------ >> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809: >> ------------------------------------------------------------------------ >> ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 >> Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport >> To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >> > >> From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >> >;tag=1 >> Call-ID: 1-6962(a)10.4.4.252 >> CSeq: 1 ACK >> Contact: sip:sipp@10.4.4.252:5060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Length: 0 >> > > > > > > ---------- Forwarded message ---------- > From: "Jerry Richards" <jerry.richards(a)teotech.com> > To: <freeswitch-users(a)lists.freeswitch.org> > Date: Mon, 24 Aug 2009 12:24:42 -0700 > Subject: [Freeswitch-users] Cannot create outgoing channel type [error] > cause: [FACILITY_NOT_SUBSCRIBED] > Hello All, > > I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP > machine > for the first time using the Getting Started Guide. I can register three > lines (1000, 1001, and 1002), but when I attempt to call one phone to the > other I hear the operator say: > > "The person at extension 1000 is not available..." > > Also, the Freeswitch log shows: > > Cannot create outgoing channel type [error] cause: > [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause: > [FACILITY_NOT_SUBSCRIBED] > > Does anyone know why I get this error? > > Best Regards, > Jerry > > > > > > ---------- Forwarded message ---------- > From: Brian West <brian(a)freeswitch.org> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 24 Aug 2009 14:33:22 -0500 > Subject: Re: [Freeswitch-users] Cannot create outgoing channel type [error] > cause: [FACILITY_NOT_SUBSCRIBED] > Are you trying to test everything on the same machine? > > /b > > On Aug 24, 2009, at 2:24 PM, Jerry Richards wrote: > > Hello All, >> >> I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP >> machine >> for the first time using the Getting Started Guide. I can register three >> lines (1000, 1001, and 1002), but when I attempt to call one phone to the >> other I hear the operator say: >> >> "The person at extension 1000 is not available..." >> >> Also, the Freeswitch log shows: >> >> Cannot create outgoing channel type [error] cause: >> [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause: >> [FACILITY_NOT_SUBSCRIBED] >> >> Does anyone know why I get this error? >> >> Best Regards, >> Jerry >> > > > > > > ---------- Forwarded message ---------- > From: Michael Jerris <mike(a)jerris.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 24 Aug 2009 15:44:18 -0400 > Subject: Re: [Freeswitch-users] Problem with cnam.js? > Every page on the wiki should be editable. If you don't already have an > account, go to: > http://wiki.freeswitch.org/index.php?title=Special:UserLogin&type=signup > > Mike > > On Aug 22, 2009, at 12:42 PM, Lars Zeb wrote: > > I think there’s something wrong with the script at > http://wiki.freeswitch.org/wiki/Examples_cnam.js. > > If you use it as is, it displays “Content-type: text/html” for the > effective_caller_id_name. In cnam.pl, the first two output lines are > generated by: > > if (!$debug) {print "Content-type: text/html\n\n";} > > with the actual name in the third line. > > So I changed: > > fd.open("read"); > buff = fd.readln(); > > if(buff) { > logger(buff, "info"); > session.setVariable("effective_caller_id_name", buff); > } > > To: > > fd.open("read"); > buff = fd.readAll(); > > if(buff[2]) { > logger(buff, "info"); > session.setVariable("effective_caller_id_name", buff[2]); > } > > Or remove the print statement from cnam.pl. > > Sorry for the code, but the page was not editable. > > Lars > > _______________________________________________ > 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: Michael Jerris <mike(a)jerris.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 24 Aug 2009 15:46:58 -0400 > Subject: Re: [Freeswitch-users] Yet another question about A500 + FS > Do you have an answer in the dialplan for that extension? Also, check out > the ignore_early_media variable. > > Mike > > On Aug 23, 2009, at 2:21 AM, Vassil Panayotov wrote: > > Hi, >> >> I managed to get our A500 running with FreeSWITCH 1.0.4 stable using >> wanpipe 3.4.4 drivers. But now I have another problem... >> I want to originate calls through event socket, and I only want to receive >> ANSWERED(+OK) reply when the user actually answers. >> >> Now the situation is: >> >> ==================================== >> originate openzap/1/a/123456 023 >> 2009-08-23 08:44:06.458166 [WARNING] ozmod_ss7_boost.c:319 TX EVENT: >> CALL_START:(80) [w1g1] CSid=2 Seq=2 Cn=[FreeSWITCH] Cd=[123456] >> Ci=[0000000000] >> 2009-08-23 08:44:06.729889 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): >> CALL_START_ACK:(81) [w1g1] Rc=0 CSid=2 Seq=4 >> 2009-08-23 08:44:06.731279 [NOTICE] switch_channel.c:602 New Channel >> OpenZAP/1:1/123456 [f8fca2be-8fa7-11de-9076-511e29dfc082] >> 2009-08-23 08:44:06.740256 [NOTICE] mod_openzap.c:1522 Pre-Answer >> OpenZAP/1:1/123456! >> API CALL [originate(openzap/1/a/123456 023)] output: >> +OK f8fca2be-8fa7-11de-9076-511e29dfc082 >> >> 2009-08-23 08:44:06.741332 [NOTICE] switch_ivr.c:1349 Transfer >> OpenZAP/1:1/123456 to XML[023@default] >> freeswitch@emo-voip> 2009-08-23 08:44:06.743475 [INFO] >> mod_dialplan_xml.c:315 Processing FreeSWITCH->023 in context default >> 2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel >> [OpenZAP/1:1/123456] has been answered >> 2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): >> CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5 >> 2009-08-23 08:44:28.903602 [WARNING] ozmod_ss7_boost.c:1141 RX EVENT (N): >> CALL_STOPPED:(85) [w1g1] Rc=16 CSid=2 Seq=6 >> 2009-08-23 08:44:28.903602 [NOTICE] mod_openzap.c:1500 Hangup >> OpenZAP/1:1/123456 [CS_EXECUTE] [NORMAL_CLEARING] >> 2009-08-23 08:44:28.903602 [WARNING] ss7_boost_client.c:218 TX EVENT (N): >> CALL_STOPPED_ACK:(86) [w1g1] Rc=0 CSid=0 Seq=3 >> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1086 Session 2 >> (OpenZAP/1:1/123456) Ended >> 2009-08-23 08:44:30.24814 [NOTICE] switch_core_session.c:1088 Close >> Channel OpenZAP/1:1/123456 [CS_DESTROY] >> ==================================== >> >> Extension 023 is an IVR. As you can see FreeSWITCH answers the call >> (2009-08-23 08:44:06.748816 [NOTICE] mod_dptools.c:649 Channel >> [OpenZAP/1:1/123456] has been answered) 20 seconds before user actually pick >> up the phone (2009-08-23 08:44:20.206010 [WARNING] ozmod_ss7_boost.c:1141 RX >> EVENT (N): CALL_ANSWERED:(84) [w1g1] Rc=0 CSid=2 Seq=5). >> >> So Sangoma drivers/daemons report the events correctly. >> How can I set FreeSWITCH to answer after receiving RX EVENT (N): >> CALL_ANSWERED from the driver? >> >> Thank you, >> V. Panayotov >> _______________________________________________ >> 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 > >

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SIPp issues - seems FS doesn't understand ACK message
by Tihomir Culjaga 24 Aug '09

24 Aug '09

Hi Anthony, I'm aware it is generating 30 retries per a call and this is killing me ... I lost my entire working day to figure out what is missing in the damn ACK message SIPp is sending back... ACK looks quite ok to me. pls can you help ? freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at 16:44:26.527236: ------------------------------------------------------------------------ INVITE sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251>SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport Max-Forwards: 70 Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403(a)10.4.4.252> > To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE Max-Forwards: 70 Subject: Performance Test Content-Type: application/sdp Content-Length: 131 v=0 o=user1 53655765 2353687637 IN IP4 10.4.4.252 s=- c=IN IP4 10.4.4.252 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 ------------------------------------------------------------------------ send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported Content-Length: 0 ------------------------------------------------------------------------ send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582: ------------------------------------------------------------------------ SIP/2.0 302 Moved Temporarily Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >;tag=Hr4mHDUeBSNyH Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 INVITE Contact: <sip:12345616094191500@pgw01.ot.hr:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported 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 ------------------------------------------------------------------------ recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809: ------------------------------------------------------------------------ ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> > From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >;tag=1 Call-ID: 1-6962(a)10.4.4.252 CSeq: 1 ACK Contact: sip:sipp@10.4.4.252:5060 Max-Forwards: 70 Subject: Performance Test Content-Length: 0 What m'i missing ? > Your ACK message must not be valid (dialog matching or something else) > so every 1 call will generate 30 retries that are queued up in the sip > stack. > > at 100cps you will be generating this problem 100 times per second and > queue up countless unfinished dialogs thus > eating up the cpu. > > > > > > > On Mon, Aug 24, 2009 at 12:19 PM, Tihomir Culjaga <tculjaga(a)gmail.com>wrote: > >> Hello, >> >> I've been with freeswittch for a while now.. and i can say it is worth >> developing it. >> >> anyhow i got into a strange issue... I'm tryng to see what load FS on my >> server can take. The Call flow is like this: >> >> SIPp FS >> >> INVITE --------> >> <------- 100 Trying >> <------- 302 Moved Temporary >> ACK ---------> >> >> >> >> I use a dummy dialplan for that. All custom functions i've build are >> disabled and i'm not using it here. Also custom modules are not loaded as >> well. >> >> >> <extension name="ServiceLookup"> >> <condition field="destination_number" expression="(^300030)(.*)"> >>  >> <action application="log" data="INFO ######################## >> ServiceLookup ########################\n"/> >> <action application="log" data="INFO ######################## >> contact = '${red_contact}' ##############\n"/> >> <action application="log" data="INFO ######################## >> CallerNum = '${caller_id_number:6:16}' ##########\n"/> >> <action application="log" data="INFO ######################## >> RADIUS auth = '${authResult}' ##########\n"/> >> <action application="execute_extension" data="doRedirect XML >> public"/> >> </condition> >> </extension> >> >> >> <extension name="doRedirect"> >> <condition field="destination_number" expression="^doRedirect$"/> >> <condition field="${authResult}" expression="^0$|^60$"> >> <action application="log" data="INFO ######################## >> RADIUS auth OK!!!' ##########\n"/> >>  >>  >> <action application="redirect" data=" >> sip:12345616094191500@pgw01.ot.hr:5060"/> >>  >>  >> <action application="hangup" data="USER_BUSY"/> >> <anti-action application="redirect" data=" >> sip:12345616094191500@pgw01.ot.hr:5060"/> >> <anti-action application="log" data="INFO >> ######################## RADIUS auth NOK!! ##########\n"/> >>  >> <anti-action application="hangup" data="USER_BUSY"/> >> </condition> >> </extension> >> >> >> When i place a call from x-lite everything works fine ... x-lite sends an >> invite, gets SIP 302 and ACKs it correctly... FS is happy. >> >> When i place a call from SIPp i have the same scenario except FS seems not >> understand ACK message from SIPp and re-sends SIP 302 multiple times untill >> it gives up. >> >> >> I beleive this is due to 302 resend issue but; when i load FS with 100 >> CPS, i can see high CPU usage (just one thread taking most load... the rest >> does almost nothing) on FS. Also, starting from 40 CPS there is a big delay >> in receiving SIP 302 messages meaning i've sent 6000 calls and so far only >> for half of them got 302 response. >> >> >> Does anybody have a clue ? >> >> >> >> >> >> Here is a trace taken on FS for calls originated from SIPp (sipp -sn uac >> 10.4.4.251 -sf uac_redirect.xml -s 30003016094191500 -trace_err -r 1 -rp 100 >> -trace_msg -inf test.txt -m 1 -l 4000): >> >> freeswitch@l01sipindir1> recv 573 bytes from udp/[10.4.4.252]:5060 at >> 16:44:26.527236: >> >> ------------------------------------------------------------------------ >> INVITE sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251>SIP/2.0 >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport >> Max-Forwards: 70 >> Contact: <sip:22222238515000403@10.4.4.252<sip%3A22222238515000403(a)10.4.4.252> >> > >> To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >> > >> From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >> >;tag=1 >> Call-ID: 1-6962(a)10.4.4.252 >> CSeq: 1 INVITE >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Type: application/sdp >> Content-Length: 131 >> >> v=0 >> o=user1 53655765 2353687637 IN IP4 10.4.4.252 >> s=- >> c=IN IP4 10.4.4.252 >> t=0 0 >> m=audio 6000 RTP/AVP 0 >> a=rtpmap:0 PCMU/8000 >> >> ------------------------------------------------------------------------ >> send 328 bytes to udp/[10.4.4.252]:5060 at 16:44:26.527566: >> >> ------------------------------------------------------------------------ >> SIP/2.0 100 Trying >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >> >;tag=1 >> To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >> > >> Call-ID: 1-6962(a)10.4.4.252 >> CSeq: 1 INVITE >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:26.535582: >> >> ------------------------------------------------------------------------ >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >> >;tag=1 >> To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >> >;tag=Hr4mHDUeBSNyH >> Call-ID: 1-6962(a)10.4.4.252 >> CSeq: 1 INVITE >> Contact: <sip:12345616094191500@pgw01.ot.hr:5060> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> 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 >> >> >> ------------------------------------------------------------------------ >> recv 383 bytes from udp/[10.4.4.252]:5060 at 16:44:26.535809: >> >> ------------------------------------------------------------------------ >> ACK sip:30003016094191500@10.4.4.251:5060 SIP/2.0 >> Via: SIP/2.0/UDP 10.4.4.252:5060;branch=z9hG4bK-6962-1-3;rport >> To: "30003016094191500"<sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >> > >> From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >> >;tag=1 >> Call-ID: 1-6962(a)10.4.4.252 >> CSeq: 1 ACK >> Contact: sip:sipp@10.4.4.252:5060 >> Max-Forwards: 70 >> Subject: Performance Test >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------ >> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:27.037070: >> >> ------------------------------------------------------------------------ >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >> >;tag=1 >> To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >> >;tag=Hr4mHDUeBSNyH >> Call-ID: 1-6962(a)10.4.4.252 >> CSeq: 1 INVITE >> Contact: <sip:12345616094191500@pgw01.ot.hr:5060> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> 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 >> >> >> ------------------------------------------------------------------------ >> send 722 bytes to udp/[10.4.4.252]:5060 at 16:44:28.037063: >> >> ------------------------------------------------------------------------ >> SIP/2.0 302 Moved Temporarily >> Via: SIP/2.0/UDP 10.4.4.252;branch=z9hG4bK-6962-1-0;rport=5060 >> From: "22222238515000403"<sip:22222238515000403@10.4.4.251<sip%3A22222238515000403(a)10.4.4.251> >> >;tag=1 >> To: "30003016094191500" <sip:30003016094191500@10.4.4.251<sip%3A30003016094191500(a)10.4.4.251> >> >;tag=Hr4mHDUeBSNyH >> Call-ID: 1-6962(a)10.4.4.252 >> CSeq: 1 INVITE >> Contact: <sip:12345616094191500@pgw01.ot.hr:5060> >> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-exported >> 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 >> >> >> Tihomir. >> >> >> >> >> >> _______________________________________________ >> 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/ > Twitter: http://twitter.com/FreeSWITCH_wire > > 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 > > > ---------- Forwarded message ---------- > From: "Raffaele P. Guidi" <raffaele.p.guidi(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Mon, 24 Aug 2009 20:24:28 +0200 > Subject: Re: [Freeswitch-users] FreeSWITCH 1.0.4 windows installer - great > but I have a little problem > Actually I did that and it worked fine. I had the problem the SECOND time I > run FS and freepbx. And (@Brian) mod_sofia was loaded but sip_profiles were > not > > On Sun, Aug 16, 2009 at 16:04, Carlos Talbot <carlos.talbot(a)gmail.com>wrote: > >> When you configure FreePBX for the first time it wipes out the >> sip_profiles directory. If you follow the FreePBX shortcut on your desktop >> it'll mention this on the last screen of the configuration. You might see >> something such as the following below. If you plan to use FreePBX you'll >> need to define trunk groups, trunks, etc in order to have the sip_profiles >> directory populated. >> regards, >> >> Carlos >> >> >> Incompatible Configuration WARNING: THE FOLLOWING FILES WILL BE DELETED! >> >> - D:/FreeSWITCH/conf/sip_profiles/external.xml >> - D:/FreeSWITCH/conf/sip_profiles/internal-ipv6.xml >> - D:/FreeSWITCH/conf/sip_profiles/internal.xml >> >> >> On Sun, Aug 16, 2009 at 4:43 AM, Raffaele P. Guidi < >> raffaele.p.guidi(a)gmail.com> wrote: >> >>> I had the sweet surprise to find the installer packaged with FreePBX... >>> really great! Why it has not been advertised as it deserves? It worked like >>> a breeze once launched, with the automatic configuration and all of that., >>> Only thing: once stopped I cannot get it to load sofia profiles anymore - >>> issueing sofia status doesn't show anything. I had to copy internal.xml and >>> default.xml from a previous installation and everything got to work again - >>> but no FreePBX anymore :( I'm sure I'm missing something important. >>> Can you give me a hint? Should sofia profiles be served by curl or >>> something? >>> >>> Thanks, >>> Raffaele >>> >>> _______________________________________________ >>> 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 >> >> > > _______________________________________________ > 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 > >

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Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]
by Jerry Richards 24 Aug '09

24 Aug '09

Hello All, I am a Freeswitch Newbie and bringing up Freeswitch on my Windows XP machine for the first time using the Getting Started Guide. I can register three lines (1000, 1001, and 1002), but when I attempt to call one phone to the other I hear the operator say: "The person at extension 1000 is not available..." Also, the Freeswitch log shows: Cannot create outgoing channel type [error] cause: [FACILITY_NOT_SUBSCRIBED]Cannot create outgoing channel type [user] cause: [FACILITY_NOT_SUBSCRIBED] Does anyone know why I get this error? Best Regards, Jerry

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