March 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Help with ignore early media
by Leonardo Alves 23 Mar '08

23 Mar '08

I need some help using the ignore_early_media. I am using a javascript file to dial a number from a asterisk server and play a file to the person who answers the phone. The problem I am having is relatated to the early_media. I think I am doing something wrong. But I cannot discover what it is. here is my .js file: ---------------------------------------------------------------------------------------------------------------------------------------------------- var languageCode = "en"; var soundDir = "sounds/"; function playFile(fileName, callBack, callBackArgs) { new_session.streamFile(soundDir + languageCode + "/" + fileName, callBack, callBackArgs); } new_session = new Session(); new_session.setCallerData("caller_id_name", "6130356060"); new_session.originate(session, "{ignore_early_media=true}sofia/gateway/asterisk/9133803764"); new_session.waitForAnswer(); if(new_session.ready()){ playFile("HelloWorld.wav"); } exit(); ------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------ Here is the Freeswitch output: jsrun HelloWorld.js API CALL [jsrun(HelloWorld..js)] output: OK 2008-03-23 10:21:08 [NOTICE] switch_channel.c:531 switch_channel_set_name() New Channel sofia/default/9133803764 [ff67046e-f8db-11dc-ad03-7f3820494c31] freeswitch@debianfs01> 2008-03-23 10:21:08 [NOTICE] switch_channel.c:1380 switch_channel_perform_mark_pre_answered() Ring-Ready sofia/default/9133803764! 2008-03-23 10:21:08 [NOTICE] sofia_glue.c:1760 sofia_glue_tech_media() Pre-Answer sofia/default/9133803764! 2008-03-23 10:21:19 [NOTICE] sofia.c:1848 sofia_handle_sip_i_state() Channel [sofia/default/9133803764] has been answered 2008-03-23 10:22:02 [NOTICE] sofia.c:1946 sofia_handle_sip_i_state() Hangup sofia/default/9133803764 [CS_HOLD] [NORMAL_CLEARING] 2008-03-23 10:22:02 [ERR] HelloWorld.js:7 mod_spidermonkey() TypeError: callBack is not a function 2008-03-23 10:22:02 [NOTICE] switch_core_session.c:748 switch_core_session_thread() Session 24 (sofia/default/9133803764) Ended 2008-03-23 10:22:02 [NOTICE] switch_core_session.c:750 switch_core_session_thread() Close Channel sofia/default/9133803764 [CS_HANGUP] The problem is: If in the originate command I use the {ignore_early_media=true} option. The script to play the file is executed when the call is answered. But the person who answers the phone cannot hear the .wav file. If I do not use the {ignore_early_media=true} option. The script to play the file starts when the call is preAnswered. And when the person in the phone answer the call, can hear the sound of the .wav file, but the audio has already started (when the preanswer is executed). ps: the computer that is making the call is a asterisk server, connected with the FS by sip, and the call is made by zaptel whe pri isdn signalling.. I would like to know if I am doing something wrong or this is a bug in the system. Thanks for the help. Leonardo Alves Devenez un meilleur amigo grâce à Yahoo! Courriel http://cf.promos.yahoo..com/courriel/visiteguidee2/

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not able to login wikipbx.com
by ram 23 Mar '08

23 Mar '08

Hi \ iam not able to loging to wikipbx.com using same user email and password of freeswitch any one suggestion, how can i get in and look the documentation ram

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Re: [Freeswitch-users] Cannot Initialize [[error near line 1621]: missing >]
by Henk Oegema 22 Mar '08

22 Mar '08

Chris, you were NOT wrong. It pointed me exactly to the line where I made the mistake. After correcting it, everything was OK. :) Big lesson (for (yet to be) a total beginner). Thanks Rgds, Henk >Guys correct me if I am wrong here but, another tip for resolving this >problem can by looking in the file: > /usr/local/freeswitch/log/freeswitch.xml.fsxml >(this is the compiled version of all the files, this may contain your >error... on the line mentioned) > >Cheers, >Chris Henk Oegema wrote: >> My system is Kubuntu 7.10 >> I downloaded an compiled (without errors) freeswitch-1.0.rc1 >> >> (There was an earlier version installed before I compiled this one. >> There I got the same error as what I got now. That was actually the reason >>for >> the new install. but the error remains, also after the new install) >> > >When I want to start FS I get following error:

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Cannot Initialize [[error near line 1621]: missing >]
by Henk Oegema 21 Mar '08

21 Mar '08

My system is Kubuntu 7.10 I downloaded an compiled (without errors) freeswitch-1.0.rc1 (There was an earlier version installed before I compiled this one. There I got the same error as what I got now. That was actually the reason for the new install. but the error remains, also after the new install) When I want to start FS I get following error: henkoegema@kubuntu:/usr/local/freeswitch/bin$ sudo ./freeswitch Error including /usr/local/freeswitch/conf/dialplan/extensions/*.xml Cannot Initialize [[error near line 1621]: missing >] <------------ :( In which file does the error occur ? Rgds Henk

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new socket module
by Jonas Gauffin 21 Mar '08

21 Mar '08

Hello, I'm trying to do a new socket module and I got a problem that is driving me mad. It's probably a really stupid thing. I got a (tcp) server that the module connects to. Then it sends a "hello" message that the server responds to. The module reads the response OK, but after that everything fails with error 730038 ("An operation was attempted on something that is not a socket."). I can't understand why, the socket descriptor is the same all the time. Got any general hints on what might cause it? 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:859 ivr_socket() Connecting to: localhost:8123. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:639 socket_send_event() send 'event;session;voicemail;sofia/default/jonas@192.168.0.58:5060;acf42337-a1ad-f84e-af01-4b7c633840dc;XML;CS_EXECUTE;jonas;jonas gaff;192.168.0.58;;;8331; ', len 152 on 21485136 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:641 socket_send_event() Bytes sent 152 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:708 socket_read() Reading, buffersize 4195 bytes from 21485136. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:713 socket_read() Bytes received 15. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:775 socket_loop() Read: action;answer; 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:708 socket_read() Reading, buffersize 4195 bytes from 21485136. 2008-03-21 09:23:59 [DEBUG] mod_ivr_socket.c:713 socket_read() Bytes received 0. 2008-03-21 09:23:59 [ERR] mod_ivr_socket.c:717 socket_read() socket receive failed: 730038. //Jonas

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wrong playback speed on shout streams and mp3 files
by Pekka Kurki 21 Mar '08

21 Mar '08

with the latest svn version the mp3 streams and files are played with higher speed. this used to work ok with an older version. my extension definition: <extension name="hki"> <condition field="destination_number" expression="^hki$">  <action application="answer"/> <action application="set" data="stream_prebuffer=131072"/>  <action application="set" data="stream_speed=1"/> <action application="playback" data="shout://217.30.180.252:8000"/> </condition> </extension> any ideas? br, pekka //(Mr.) Pekka K. Kurki Intellectics GesmbH http://www.intellectics.com Helenenstrasse 44, A-2500 Baden, Austria Tel. +43 780 815811 (NEW AREA CODE!!!) Mobile +43 681 10358833 Fax +43 780 815819 (NEW AREA CODE!!!) (Tel. US: 360-968-1576 (NEW!!) UK: 0870 4787064 FIN: 02-48084133) sip: pekka.kurki(a)sip.intellectics.com enum: +432256815811, ISN: 1*743 skype: pkkurki, pkkurki-sip (sip gateway) gtalk: pekkis50(a)gmail.com (sip gateway - experimental) gizmo: pekkis50 email: Pekka.Kurki(a)intellectics.com www: http://pekka.kurki.at push-to-call: http://www.intellectics.com/call.htm//

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RC1 Has Arrived!
by Anthony Minessale 20 Mar '08

20 Mar '08

FreeSWITCH Community, After nearly 2 years of non-stop development, FreeSWITCH has evolved from an idea and an empty directory into a fully operational soft-switch capable of withstanding the loads 8-10 times as large as anything else they were able to deploy in the past with other software according to the reports from our early adopters. I am happy to announce the immediate availability of FreeSWITCH 1.0 “Phoenix“ (Release Candidate 1). The Phoenix symbolizes the rebirth of a new era and I hope this new release will be the first step in a new age of voice-over-IP development we can all benefit from. We anticipate the official FreeSWITCH 1.0 “Phoenix” release date to be Monday, May 26th 2008. We will have a formal release presentation at this year’s ClueCon 2008 August 5th-7th 2008. http://www.cluecon.com I would like to thank our community for the testing, bug reporting, code contributions, feedback and financial donations that helped us get this far, especially those who were involved from day 1. Please take this opportunity to test drive FreeSWITCH as we will be making it available as binary packages for the first time to practice for the official 1.0 release. The code and arrangement of the configuration will not change apart from bug fixes and tweaks from this point forward. As always, the code will still be available in svn at the usual place. For download instructions please stay tuned to: http://wiki.freeswitch.org/wiki/Installation_Guide The packages will be online soon. Thanks Everyone! Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org pstn:213-799-1400 ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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mod_limit fall through
by bsnipes 20 Mar '08

20 Mar '08

I am current jumping to a different extension when a limit is reached and the limit_exceeded condition is hit by changing the 'dialed_ext' and calling: <action application="bridge" data="${sofia_contact($${domain}/${dialed_ext}@$${domain})}"/> I hit a problem when it hits the 'second line' for the user. I would like it to check via mod_limit if that extension is on the phone and respond appropriately at that point but it doesn't seem to hit the mod_limit for that second extension when it is called that way. Is there another way to call an extension and have it go through the limit process again? Thanks, Brian -- View this message in context: http://www.nabble.com/mod_limit-fall-through-tp16128842p16128842.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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session.recordFile and only silence
by Jonas Gauffin 20 Mar '08

20 Mar '08

Hello How will freeswitch handle if a caller comes to the voicemail and is only silent and then hangs up? (hangs up before the silence_secs value is exceeded) //Jonas

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jira seems to be having some problems
by Kurt Marasco 19 Mar '08

19 Mar '08

I'm trying to update a bug report at http://jira.freeswitch.org/ but the server is not properly responding. Thought that maybe someone here might be able to resolve the issue.

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