June 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Problems running SIP Internal Profile: Error Creating SIP UA for profile: internal
by Dan Carlson 17 Jun '08

17 Jun '08

Hi Freeswitch Users, I've just started with Freeswitch, just built and installed it yesterday (Redhat 5). Freeswitch seems like a really interesting platform. I'm looking forward to exploring it. It built and installed fine but unfortunately I'm having a little trouble with getting the Sofia Internal SIP Profile running. The External and Nat profiles start up fine but I get this error both at startup and when I attempt to run 'sofia profile internal start' : 2008-06-16 14:56:59 [ERR] sofia.c:532 nua() Error Creating SIP UA for profile: internal Setting sofia to a higher log level from the prompt gives me no additional information. I've not really done much configuring of internal.xml or var.xml aside from setting bind_server_ip,external_rtp_ip,external_sip_ip to the box's IP. Anyone have any suggestions? Perhaps something that needs to be configured that I've missed? I searched and read through the documentation but can't find any other mention of this issue. Thanks, Dan Carlson

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php + js
by Czaderna 17 Jun '08

17 Jun '08

Hello. How to display uuid of channel from js script in php script ? My js script looks like this: //---------------------------------------- function MakeFirstCall(caller_number) { var gateway = "sofia/gateway/myGateway/"; session_leg_a = new Session(); session_leg_a.originate(session_leg_a, gateway + caller_number); session_leg_a.execute("park()"); session_leg_a.streamFile("msg.wav"); var leg_a_uuid = session_leg_a.uuid; // How return leg_a_uuid to php script ???? } if(argv[0] != "") { MakeFirstCall(argv[0]); } exit(); //---------------------------------------- In php I have following lines: //---------------------------------------- $handle = fsockopen('127.0.0.1', '8021', $erno, $errstr, 30); if(!$handle) { die("Connect failed with erno $erno and errstr $errstr\n"); } else { $response= fread($handle, 1024); $authMessage = "auth myFSpassword\r\n\r\n"; fwrite($handle, "$authMessage"); $response = fread($handle, 1024); echo " First response $response "; $command = "api jsrun firstCall.js 0048xxxxxxxx1\r\n\r\n"; fwrite($handle, $command); $response = fread($handle, 1024); echo " Second response $response "; $response = fread($handle, 1024); echo " Third response $response "; } //---------------------------------------- After I run php script I get OK+ result but I want uuid.

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Problem with audio path
by Hafiz Al-Noor 17 Jun '08

17 Jun '08

Hi Gurus, Anyone can help me. I'm a new user and have installed FreeSwitch on a server running a private IP. I have forwarder all ports that are required from my public IP. I am not on the site where the server is installed and am trying to test SIP calls. I have configured the <X-PRE-PROCESS cmd="set" data="domain=$${local_ip_v4}"/> with my public IP address. I only have 1 public IP that my router is taking therefore having to port forward all the required ports. In the trace everything is appearing as expected now except the RTP portion. Can anyone help in giving me guidance on how to route the RTP stuff back to the public IP? Below is the debug messages 2008-06-17 09:50:28 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/1001(a)124.82.205.90 [c32557f4-3c0f-11dd-b65b-87373c122460]2008-06-17 09:50:28 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1001->5000@default2008-06-17 09:50:28 [INFO] mod_sofia.c:1029 sofia_receive_message() Asked to send early media by sofia/default/1001(a)124.82.205.902008-06-17 09:50:28 [INFO] mod_sofia.c:1070 sofia_receive_message() Ring SDP:v=0o=FreeSWITCH 1213638528 1213638529 IN IP4 192.168.1.55s=FreeSWITCHc=IN IP4 192.168.1.55t=0 0a=sendrecvm=audio 28900 RTP/AVP 0 101 13a=rtpmap:0 PCMU/8000a=rtpmap:101 telephone-event/8000a=fmtp:101 0-16a=rtpmap:13 CN/8000a=ptime:20 2008-06-17 09:50:28 [NOTICE] switch_channel.c:1380 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/default/1001(a)124.82.205.90!2008-06-17 09:50:28 [NOTICE] mod_sofia.c:1073 sofia_receive_message() Pre-Answer sofia/default/1001(a)124.82.205.90!2008-06-17 09:50:28 [NOTICE] mod_dptools.c:438 answer_function() Channel [sofia/default/1001(a)124.82.205.90] has been answered2008-06-17 09:51:59 [NOTICE] switch_core_state_machine.c:156 switch_core_standard_on_execute() Hangup sofia/default/1001(a)124.82.205.90 [CS_EXECUTE] [NORMAL_CLEARING]2008-06-17 09:51:59 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 1 (sofia/default/1001(a)124.82.205.90) Ended2008-06-17 09:51:59 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1001(a)124.82.205.90 [CS_HANGUP] _________________________________________________________________ Manage multiple email accounts with Windows Live Mail effortlessly. http://www.get.live.com/wl/all

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unable to test SIP call
by Hafiz Al-Noor 17 Jun '08

17 Jun '08

Hi FreeSwitch users, I am trying to test SIP calls by calling extension 1000 from extension 1001 using xlite. Can anyone help show me whats incorrectly configured? Cheers, Hafiz The debug message shows the following: - 2008-06-17 10:44:59 [INFO] mod_dialplan_xml.c:223 dialplan_hunt() Processing 1001->1000@default2008-06-17 10:44:59 [INFO] switch_ivr_async.c:1357 switch_ivr_bind_dtmf_meta_session() Bound: 1 execute_extension::dx XML features2008-06-17 10:44:59 [INFO] switch_ivr_async.c:1357 switch_ivr_bind_dtmf_meta_session() Bound: 2 record_session::/usr/local/freeswitch/recordings/1001.2008-06-17-10-44-59.wav2008-06-17 10:44:59 [INFO] switch_ivr_async.c:1357 switch_ivr_bind_dtmf_meta_session() Bound: 3 execute_extension::cf XML features2008-06-17 10:44:59 [WARNING] switch_ivr_originate.c:537 switch_ivr_originate() No origination URL specified!2008-06-17 10:44:59 [ERR] switch_ivr_originate.c:813 switch_ivr_originate() Cannot create outgoing channel of type [user] cause: [DESTINATION_OUT_OF_ORDER]2008-06-17 10:44:59 [INFO] mod_dptools.c:1551 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER2008-06-17 10:44:59 [NOTICE] mod_dptools.c:438 answer_function() Channel [sofia/default/1001(a)124.82.205.90] has been answered2008-06-17 10:45:35 [NOTICE] switch_core_state_machine.c:156 switch_core_standard_on_execute() Hangup sofia/default/1001(a)124.82.205.90 [CS_EXECUTE] [NORMAL_CLEARING]2008-06-17 10:45:35 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 1 (sofia/default/1001(a)124.82.205.90) Ended2008-06-17 10:45:35 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/1001(a)124.82.205.90 [CS_HANGUP] _________________________________________________________________ NEW! Get Windows Live FREE. http://www.get.live.com/wl/all

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custom ringback behaviour
by Cavalera Claudio Luigi 16 Jun '08

16 Jun '08

Hello guys, I'm keeping on playing with freeswitch (now with 1.0) on my spare work time with two final goals: 1)create a nice demo portfolio to show funtional potential of fs. 2)show fs performance under heavy load compared to commercial carrier level softswitches (millions of CDRs per day). First, I have to enrich my demo portfolio which is still quite poor. Things should become easier now, as I've succeed in routing call from my fs box to my softswitch connected to public landline. This means for example that I can call from a softphone registered to fs any telephone number with a rough dialplan like this:  <include> <context name="default"> <extension name="claudio_external"> <condition field="destination_number" expression="^(\d+)$"> <action application="set" data="effective_caller_id_name=John Freeswitch"/> <action application="set" data="effective_caller_id_number=31337"/> <action application="bridge" data="sofia/default/$1@MySswIP"/> </condition> </extension> </context> </include> Now let's consider this couple of commands issued through fs console (I hope I'm using % vs @ properly here): 1) originate {ringback=/usr/local/freeswitch/sounds/calling.wav}sofia/default/1000%my FsBox &bridge(sofia/default/0287654321@MySswIP) in this case when I answer the first ringing phone (softphone) while the second phone is ringing I hear "calling.wav": wonderful! 2)originate {ringback=/usr/local/freeswitch/sounds/calling.wav}sofia/default/0212345 678@MySswIP &bridge(sofia/default/0287654321@MySswIP) in this case when I answer the first ringing phone I don't hear calling.wav, instead I hear normal ringback tone while the second phone is ringing. What could be the cause of these different behaviours? Thanks, Claudio 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. -----------------------------------------------------------------------------------------------------

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equivalent of realtime
by Jonathan K. Creasy 13 Jun '08

13 Jun '08

Can I load SIP peers and extensions from a MYSQL database with Freeswitch? -Jonathan No virus found in this outgoing message. Checked by AVG. Version: 7.5.524 / Virus Database: 270.3.0/1501 - Release Date: 6/13/2008 6:33 AM

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Installation report, a python crash and a bottleneck
by Krešimir Tonković 13 Jun '08

13 Jun '08

Hi! I'm new to freeswitch and I like to report my success with it and a few failures. I'll be a little bit vague on some details because I must protect some business details. Sorry for that. I have no experience with asterisk, so many concepts were new to me. We run a hosted IVR system with a few hundred lines. We have a few servers running a SIP/VoiceXML application server and connect to the network with SIP/ISDN gateways. Recently we started an IVR with very short calls and very high CPS. Our existing software doesn't handle this scenario very well, so I started looking into alternatives. FreeSwitch caught my eye because of its support for multiple scripting languages. I love python and this feature put FS into the evaluation list. So I started on friday. I installed FS from the debian repositories on my ubuntu 8.04 laptop and tried some examples from the wiki ("Some thing to try out!"). I was very impressed that everything worked right out of the box. I was a little disappointed that mod_python wasn't included in the distribution so I checked out the source and compiled everything. An hour later I had another installation of FS. It took me a few hous to get the dialplan right. Because our service only runs IVRs and uses no switching, I removed everything from the default dialplan (mainly because it conflicted with the ANI numbers we get from the gateways). Another hour later, I had a simple IVR in python done: use a web service for a database lookup and play an appropriate prompt. I didn't use the database directly because I wanted the best possible comparison to what our current system does, and (our) VoiceXML can't use databases directly, but can use web services. In less than 1 working day I had everything running. Quite good. Time for load testing :-) Our old software handles around 20 CPS on my laptop. I inceased max_sessions to 5000 and sessions-per-second to 100 and started sipp. The result was quite bad - I could not get over 8 CPS! The processor barely noticed that FS was running, so I had no idea what the bottleneck was. I still don't. After fiddling with this for a while, I gave up and decided to try it on one of our production machines. Weekends are not very busy, so I took one offline. It's a HP proliant server with 1 quad-core xeon on 2 GHz, 2G ecc ram and 10krpm disks. The server is also running ubuntu 8.04 (server edition) so I just copied the binaries. I ran sipp from another machine, with the uac scenario and limiting the call duration to 4secs: sipp <FS server ip> -sn uac -d 4000 -s <ivr_number> Theoretically, as each call lasts 4 seconds, the total calls number should never exceed 4x current CPS. These are the results: With up to 27 CPS everything was stable. The calls count was almost exactly 4 timee the CPS, indicating that new calls were ansewered immediately. This I also verified by calling in. Up to 30 CPS everything was stable for a while, but then the total calls number exploded to the limit set by sipp. The processor load was very reasonable, so I again I ran into the bottleneck mentioned above. After sipp hits the total call limit, it will not create new calls until some are released. So CPS oscillated between 0 and 30 as shown by sipp. CDRs show that there was an average of 27 CPS. At this point, when I called in, I got ringback tone for as long as the operator allows (60s) and then I was dropped. With a softphone I could reach the IVR after about 80 sec. When I set sipp to more than 30 CPS, the number of total calls exploded immediately. Experimenting some more, I found I could contain the explosion (and the instability in CPS) by limiting the number of total calls to 4x current cps when cps was up to 30. Thus, by starting sipp like this: sipp <FS server ip> -sn uac -d 4000 -s <ivr_number> -l 120 I could go up to 30CPS and get reasonably stable real 30CPS. When calling in with a real phone, I would reach the IVR after 2-3 seconds of ringback, which is acceptable. This simulation run for several hours without any other problems. With sipp <FS server ip> -sn uac -d 4000 -s <ivr_number> -l 160 and setting cps to 40, the total calls count obviously never passed 160, but the cps shown by sipp became unstable, oscillating between 0 and 40. These results are only slightly better than our current SIP server. Today I put FreeSwitch into production. The unexpected thing here was that when FreeSwitch talked to our gateways instead of sipp, it crashed a few times. I don't associate these crashes with load because it happened equally on low and high load. Here's the log: 2008-06-09 09:34:33 [NOTICE] switch_core_session.c:753 switch_core_session_thread() Session 166 (sofia/internal/---- deleted stuff -----) Ended 2008-06-09 09:34:33 [NOTICE] switch_core_session.c:755 switch_core_session_thread() Close Channel sofia/internal/---- deleted stuff ----- [CS_HANGUP] 2008-06-09 09:34:33 [CRIT] switch_core_state_machine.c:218 print_trace() Obtained 10 stack frames. /usr/local/freeswitch/lib/libfreeswitch.so.1 [0xb7e413b1] [0xb7f69420] /usr/local/freeswitch/mod/mod_python.so [0xb011b46a] /usr/lib/libpython2.5.so.1.0(PyCFunction_Call+0xfa) [0xb002a50a] /usr/lib/libpython2.5.so.1.0(PyObject_Call+0x37) [0xafff38d7] /usr/lib/libpython2.5.so.1.0(PyEval_EvalFrameEx+0x4067) [0xb0076907] /usr/lib/libpython2.5.so.1.0(PyEval_EvalCodeEx+0x748) [0xb007a368] /usr/lib/libpython2.5.so.1.0(PyEval_EvalFrameEx+0x601c) [0xb00788bc] /usr/lib/libpython2.5.so.1.0(PyEval_EvalCodeEx+0x748) [0xb007a368] /usr/lib/libpython2.5.so.1.0 [0xb001667f] 2008-06-09 09:34:33 [CRIT] switch_core_state_machine.c:319 switch_core_session_run() Thread has crashed for channel sofia/internal/---- deleted stuff ----- It seems like mod_python is not quite ready for production yet :-) I have a few core dumps available on demand, cca 2MB each. Turning crash protection on didn't help. FreeSwitch would reject new calls, and wouldn't shutdown completely. I had to kill it. I would still like to replace our existing system with FreeSwitch because I find way more comfortable to work with. It has great potential and I'm sure it is being succesully deployed in many places as I write this. The python crash is probably a simple bug. The invisible bottleneck is what troubles me more. Any help would be greatly appreciated. Finally, this is all with FreeSwitch Version 1.0.pre4 (8760). -- kresho

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Conference call June 11 2008
by Raymond Chandler 13 Jun '08

13 Jun '08

Here's the agenda for this week's conference call. We hope to see a larger turnout than last week. Looking forward to hearing from everyone at 1PM CDT... 2PM EDT ... 6PM GMT * 1 Brief review of progress on dead end page cleanup from last week's call <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#Brief_review_of_progre…> * 2 New Features To Doc <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#New_Features_To_Doc> o 2.1 MODAPP-100 <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#MODAPP-100> o 2.2 Polycom Presence <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#Polycom_Presence> o 2.3 Per-User acl <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#Per-User_acl> o 2.4 Voicemail Notification Email <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#Voicemail_Notification…> o 2.5 New channel variables added since 1.0.0 <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#New_channel_variables_…> o 2.6 Complete list of channel variables, how to reliably extract from source code? <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#Complete_list_of_chann…> o 2.7 undocumented variables <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#undocumented_variables> * 3 Recommend review of getting started guide <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#Recommend_review_of_ge…> * 4 Review app syntax in mod_dptools.c <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#Review_app_syntax_in_m…> * 5 New Call Time <http://wiki.freeswitch.org/wiki/Wiki_meet_2008_06_11#New_Call_Time> c u there, -Ray

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Group Pickup in Lua, Please test
by Luis Jimenez 13 Jun '08

13 Jun '08

Guys, please try this, i did in my lab but just 5 phones. http://wiki.freeswitch.org/wiki/Group_Pickup_Using_lua Luis Jimenez

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Re: [Freeswitch-users] Gafachi Setup / Caller ID - Resolved
by Nick Temple 12 Jun '08

12 Jun '08

Brian, Updating to /trunk and changing {} to [] in the origination string has fixed the problem, calls are now going through w/ appropriate caller id. Should this be documented in the Wiki somewhere? I'll also add Gafachi to the appropriate place in the wiki. Thank you! Nick On Thu, Jun 12, 2008 at 1:32 PM, Brian West <brian(a)freeswitch.org> wrote: > I know change the {} to [] this has to be a bug introduced in a recent > patch in trunk. > /b > > On Jun 12, 2008, at 12:12 PM, Nick Temple wrote: > > Show me the INVITE: > > INVITE sip:1502XXXXXXX@sip.gafachi.com<sip%3A1502XXXXXXX(a)sip.gafachi.com>SIP/2.0 > Via: SIP/2.0/UDP 75.101.133.137:5080;rport;branch=z9hG4bKm4028av6cyD0F > Max-Forwards: 70 > From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com<sip%3Aa6031sTVQLeOhyIV(a)sip.gafachi.com> > ;transport=udp>;tag=j8BBv5jaycaFF > To: <sip:15025482003@sip.gafachi.com<sip%3A15025482003(a)sip.gafachi.com> > > > Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 > CSeq: 100525131 INVITE > Contact: <sip:a6031sTVQLeOhyIV@75.101.133.137:5080>;transport=udp > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.0.0-exported > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, > NOTIFY, REFER, UPDATE, REGISTER, INFO > Supported: 100rel, precondition, timer > Allow-Events: talk > Proxy-Authorization: Digest username="a6031sTVQLeOhyIV", realm=" > sip.gafachi.com", nonce="4fa20cd7b73a320d", algorithm=MD5, uri=" > sip:15025482003@sip.gafachi.com <sip%3A15025482003(a)sip.gafachi.com>", > response="617d7d2f882293a061ad27a72cd4ca60" > Min-SE: 120 > Content-Type: application/sdp > Content-Disposition: session > Content-Length: 325 > Remote-Party-ID: "" <sip:@sip.gafachi.com <sip%3A(a)sip.gafachi.com> > >;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 4716203835921160991 4794018502006774974 IN IP4 > 10.252.115.245 > s=FreeSWITCH > c=IN IP4 75.101.133.137 > t=0 0 > a=sendrecv > m=audio 28950 RTP/AVP 0 8 3 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:3 GSM/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > > > On Thu, Jun 12, 2008 at 1:05 PM, Brian West <brian(a)freeswitch.org> wrote: > >> Show me the INVITE. >> >> On Jun 12, 2008, at 11:50 AM, Nick Temple wrote: >> >> SIP/2.0 604 Invalid Calling Party Number >> Via: SIP/2.0/UDP 75.101.133.137:5080 >> ;branch=z9hG4bKm4028av6cyD0F;received=75.101.133.137;rport=5080 >> From: "" <sip:a6031sTVQLeOhyIV@sip.gafachi.com<sip%3Aa6031sTVQLeOhyIV(a)sip.gafachi.com> >> ;transport=udp>;tag=j8BBv5jaycaFF >> To: <sip:15025482003@sip.gafachi.com<sip%3A15025482003(a)sip.gafachi.com> >> >;tag=gss0c342e9c3ad2398e2f610f7ce613a089 >> Call-ID: b161c7c6-b33c-122b-33b2-123138006c07 >> CSeq: 100525131 INVITE >> Server: Gafachi UAS v110.08 >> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE >> Contact: <sip:15025482003@64.192.112.13<sip%3A15025482003(a)64.192.112.13> >> > >> Content-Length: 0 >> >> >> Brian West >> sip:brian@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 > > > Brian West > sip:brian@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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