September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Freeswitch with Audiocode Mediant 2000
by Gopal krishnan 23 Sep '08

23 Sep '08

Hi, I followed the below link to configure the Audiocode Mediant 2000 with Freeswitch http://wiki.freeswitch.org/index.php?title=Configuring_AudioCodes_MP-114/11… but the above link is for FXO line, where I am using digital PRI line. when I try to dial I am getting call failed, the traffic from freeswitch were hitting audiocode the log as follows, attached with this email, *some sample SIP header as follows,* d:2h:17m:7s INVITE sip:9894929942@172.20.176.254<sip%3A9894929942(a)172.20.176.254>SIP/2.0 Via: SIP/2.0/UDP 172.20.176.31;rport;branch=z9hG4bKKmB9HrNr22HZQ Max-Forwards: 69 From: "Extension 1002" <sip:9894929942@172.20.176.31<sip%3A9894929942(a)172.20.176.31> >;tag=j9a4e9Q4ycvtr To: <sip:9894929942@172.20.176.254 <sip%3A9894929942(a)172.20.176.254>> Call-ID: 7702517d-0413-122c-efab-0019d150d051 CSeq: 104969298 INVITE Contact: <sip:mod_sofia@172.20.176.31:5060> User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9596M Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, PRACK, MESSAGE, SUBSCRIBE, NOTIFY, REFER, UPDATE, REGISTER, INFO, PUBLISH Supported: 100rel, timer, precondition, path, replaces Allow-Events: talk, presence, dialog, call-info, sla, include-session-description, presence.winfo, message-summary Min-SE: 120 Content-Type: application/sdp Content-Disposition: session Content-Length: 347 Remote-Party-ID: "Extension 1002" <sip:9894929942@172.20.176.31<sip%3A9894929942(a)172.20.176.31> >;screen=yes;privacy=off 1d:2h:17m:7s ( sip_stack)(212 ) ?? [WARNING] AcSIPParser: Unrecognized Header was detected at line: 12 1d:2h:46m:9s ( lgr_TrnkGrp)(344 ) !! [ERROR] #1:TrunkGroup::AllocateEndPoint- Can't find EndPoint for phone number 9894929942 1d:2h:46m:9s ( lgr_psbrdif)(345 ) !! [ERROR] AcBoard::GetEndPoint- Can't find EndPoint for Dest:9894929942 Source:9894929942 SourceIp:ac14b01f 1d:2h:46m:9s ( lgr_psbrdif)(346 ) TrunkBoard::GetEndPoint- Current trunk status:0010 1d:2h:46m:9s ( lgr_call)(347 ) !! [ERROR] Call::GetEndPoint- Can't find endpoint for phone number 9894929942 *Freeswitch log* *as follows* http://pastebin.freeswitch.org/5635 So how to proceed in this stage. -- Thank you with regards, Gopal,

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Grandstream
by jocke eriksson 23 Sep '08

23 Sep '08

Hello Freeswitch users, I'm having some trouble with a Grandstream phone of mine. Each time i try to call it up and stream a file to it Freeswitch prints this ' Codec G722 Exists but not at the desired implementation. 16000hz 20ms '. Does anybody use Granstream GXP-2000 and now how to fix it, or can i modify Freeswitch to except this codec , all help is appreciated . Regards Jocke

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Running js file automatically when starting freeswitch
by preetha Ayyappan 23 Sep '08

23 Sep '08

Hi, I am trying to start the js file automatically when I start freeswitch.Instead of running the script like jsrun filename.js in the freeswitch console,I need to run the javascript along with the freeswitch.Is this possible?if yes please guide me to do this. Thanks

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Problems with configure script
by Jon Bruel 23 Sep '08

23 Sep '08

I have got the trunk version of FreeSWITCH, have run bootstrap.sh, but I get get stuck in configure, where I get the following message, which I don't know how to act on: ./configure: line 13837: AX_COMPILER_VENDOR: command not found configure: error: Please update configure.in with SOLINK values for your compiler Versions: Freeswitch: build 9609, automake: 1.10.1, autoconf: 2.61, libtool: 1.514 or 1.5.24. Operating system is CentOS, uname -r = 2.6.9.-67.0.4.plus.c4 or 2.6.25.9-40.fc8 (same problem on two different systems). What should I do? Jon

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Creating and destroying local_stream dynamically
by Cesar Cepeda 22 Sep '08

22 Sep '08

Hi, I need to create and destroy local_streams dynamically, that is, I need to be changing the MOH of several fifo's in real-time, since local_stream creates a thread that is broadcasting audio I would like to be able to create and destroy them as I need them. How can this be achieved? Cesar Cepeda.

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tcapi in svn doesnt have theme folder
by xbipin 22 Sep '08

22 Sep '08

hi, i did a snv update for tcapi which is the web frontend for freeswitch but when i tried to make it all work on win2k3 platform, it did but then due to the theme folder missing, the web intwerface seems to be all messed up and not even a proepr interface. Anyone managing it here, can u add the theme folder so i can test and report any issues? Regards, Bipin -- View this message in context: http://www.nabble.com/tcapi-in-svn-doesnt-have-theme-folder-tp19611927p1961… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Some common extensions for testing
by henkoegema 22 Sep '08

22 Sep '08

Just installed FS 1.0.1 When testing extension 9999 (calling from extension 2000) it rings 3 times and than I get busy tone. freeswitch@ubuntu> 2008-09-22 11:02:10 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/2000(a)192.168.2.11 [240f78bc-8885-11dd-8238-95905a8f75b0] 2008-09-22 11:02:10 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 2000->9999@default 2008-09-22 11:02:10 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/2001 [24106614-8885-11dd-8238-95905a8f75b0] 2008-09-22 11:02:11 [NOTICE] sofia.c:2167 sofia_handle_sip_i_state() Ring-Ready sofia/internal/2001! 2008-09-22 11:02:11 [NOTICE] mod_sofia.c:1058 sofia_receive_message() Ring-Ready sofia/internal/2000(a)192.168.2.11! 2008-09-22 11:02:11 [NOTICE] switch_ivr_originate.c:1148 switch_ivr_originate() Ring Ready sofia/internal/2000(a)192.168.2.11! 2008-09-22 11:02:26 [NOTICE] sofia.c:2545 sofia_handle_sip_i_state() Hangup sofia/internal/2001 [CS_CONSUME_MEDIA] [USER_BUSY] 2008-09-22 11:02:26 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: USER_BUSY :-/ 2008-09-22 11:02:26 [NOTICE] mod_dptools.c:1816 audio_bridge_function() Hangup sofia/internal/2000(a)192.168.2.11 [CS_EXECUTE] [USER_BUSY] 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 1 (sofia/internal/2000(a)192.168.2.11) Ended 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/2000(a)192.168.2.11 [CS_HANGUP] 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 2 (sofia/internal/2001) Ended 2008-09-22 11:02:26 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/2001 [CS_HANGUP] Calling 9996 - standard echo test , it rings 10 times and than disconnects. 2008-09-22 11:07:04 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NO_ANSWER Everything was working fine initially in version 1.0.0. Then all of a sudden, I got the problems (mentioned above) in version 1.0.0, so I decided to install a new version (1.0.1) Unfortunately the problem persists. When calling 5000 (demo ivr), option 1 (conference) OK option 2 (echo test) NOK option 3 (moh) NOK option 4 (sample ivr submenu) OK option 5 (screaming monkey) NOK When calling internally from extension 2002 (SJPhone) to 2000 (Analog phone with Linksys SPA 3000) the moh is very cracky. Doing the same test via an external DID line gives a fine moh ! Henk -- View this message in context: http://www.nabble.com/Some-common-extensions-for-testing-tp19604479p1960447… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Using the Command API
by Klaus Teller 22 Sep '08

22 Sep '08

Hi, I'm trying to figure out how to use the command API via XML-RPC. I want to do the two following things one after the other: 1) place a call, and 2) play an audio on the newly created call. The Java code for placing a call (copied from wiki) works fine and is following: XmlRpcClientConfigImpl config = new XmlRpcClientConfigImpl(); XmlRpcClient client = new XmlRpcClient(); config.setServerURL(new URL("http://192.168.50.70:8080/RPC2")); config.setBasicUserName("freeswitch"); config.setBasicPassword("works"); client.setConfig(config); String response= client.execute("freeswitch.api", new Object[] { "originate", "sofia/internal/1003 & park()" }) .toString(); >From my understanding the code for playing the audio should be: client.execute("freeswitch.api", new Object[] { "playback", "32a5e11a-8649-11dd-bb78-fd02030a93ef /var/lib/freeswitch/sounds/hello.wav" }); Unfortunately, the latter doesn't work. I suspect it's because the call has been parked. What is the proper way to address this? Thanks, Klaus. -- GMX startet ShortView.de. Hier findest Du Leute mit Deinen Interessen! Jetzt dabei sein: http://www.shortview.de/wasistshortview.php?mc=sv_ext_mf@gmx

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Grandstream
by jocke eriksson 22 Sep '08

22 Sep '08

Hello Freeswitch users, I'm having some trouble with a Grandstream phone of mine. Each time i try to call it up and stream a file to it Freeswitch prints this ' Codec G722 Exists but not at the desired implementation. 16000hz 20ms '. Does anybody use Granstream GXP-2000 and now how to fix it, or can i modify Freeswitch to except this codec , all help is appreciated . Regards Jocke

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How to upgrade ?
by henkoegema 22 Sep '08

22 Sep '08

At the moment I'm using FS version 1.0.0. I want to upgrade to the latest version. Can I just type #make current (?) :working: without loosing my own (or edited standard) conf files ? -- View this message in context: http://www.nabble.com/How-to-upgrade---tp19589317p19589317.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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