- FreeSWITCH-users - FreeSWITCH Mailing Lists

jitter buffer usage
by savoipguy＠mailbox.co.za 01 Aug '08

01 Aug '08

Hi I have a situation is which I need to do jitter buffering. The setup is as follows: (local sip user)---->freeswitch------->(remote gateway) The leg that needs de-jittering is the RTP travelling from the remote gateway to the local sip user. I only want Freeswitch to enable jitterbuffer on these RTP packets received from the gateway, going to the local sip user. No other de-jittering required. Would the correct usage be: <action application="export" data="nolocal:jitterbuffer_msec=180"/> <action application="bridge" data="sofia/gateway/$1(a)gateway.com"/> Any help will be appreciated. Thanks Clive ------------------------------------------- South Africas premier free email service - www.webmail.co.za ------------------------------------------------------------------ For super low premiums, click here http://www.webmail.co.za/dd.pwm

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Event Socket: Get Via list
by Alois Komenda 01 Aug '08

01 Aug '08

Hello Freeswitchers, is it possible to get a list of all hosts, which are conained in the Via header, over the Event Socket? Most important for me is to know the first proxy, which forwarded the call (i. e. outbound proxy of the caller). But a complete list of involved forwarding hosts would be nice as well. What host is displayed in the variable_sip_via_host? Thanks in advance! Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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Module Load Error
by Shehzad Pankhawala 01 Aug '08

01 Aug '08

I have installed FS from svn trunk. It works fine but some modules are not loaded even if they compiled, and loaded in modules.conf.xml While start freeswitch and load that module the following error displayed: 2008-08-01 10:58:43 [CRIT] switch_loadable_module.c:756 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/<name of modules listed below>.so **/usr/lib/libnssutil3.so: undefined symbol: PR_GetEnv** Above Error is displayed in the below all modules: mod_xml_curl mod_xml_cdr mod_spidermonkey mod_shout I need to use javascript and xml cdr it is important for me to load these two modules. Is there any dependency missing. Thanks ----- Shehzad

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Complete seperation of VOIP app from FS
by Erol Akarsu 01 Aug '08

01 Aug '08

Still searching event_socket module. If we set up dialplan, all received calls will routed based on dialplan rules. But at the same time, if there is remote process connected to FS through socket_module, dialplan or remotes process will have priority to control FS behavior. Can remote process interfere dialplan rules.? Is it possible to define a dialplan so that one extension will be routed to a remote process and other extension will be routed to different process? I am thinking to abstract out IVR or any VOIP application from FS box completely and to run application on different host or process. Regards Erol Akarsu

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FS hanging up call with reason UNKNOWN after 10 seconds
by Cesar Cepeda 31 Jul '08

31 Jul '08

Hi, I'm experimenting with mod_fifo, I'm having a problem when I insert a consumer into the FIFO. The scenario is the following: . Consumer enters the FIFO with <action application="fifo" data="myFIFO out wait undef undef"/> . Caller enters the FIFO with <action application="fifo" data="myFIFO in undef undef"/> . The consumer and the caller are bridged. . The caller hangs up the call . The consumer reenters the FIFO. . After exactly 10 seconds the consumer channel is hung up by FS with reason UNKNOWN. If I configure music_on_hold the call is not hung. The issue is also present if I use "fifo_consumer_wrapup_key" and wait more than 10 seconds to press the key. It would seem that the problem is that sofia detects that I'm not sending any audio and hangs up. How can I avoid this? Thanks, Cesar Cepeda.

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PBX features
by Simon Shaw 31 Jul '08

31 Jul '08

In the list of features on the spec-sheet in the FS Wiki, one of the features is "Basic IP/PBX features". Do you have a list of specific features that you support? Thanks, Simon

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New user problems (Nokia E60)
by Martin Joseph 31 Jul '08

31 Jul '08

Hello FreeSWITCHers! I am new to FreeSWITCH, having played with/used asterisk for a couple of years and being based on the mac platform, I was looking for a superior choice and FreeSWITCH seems to fit the bill. I have built and am running Freeswitch for learning purposes on a PowerPC mac running OSX 10.4.11. This seems to be running ok. I initially configured an SIP softphone (SJphone) as extension 1000 which seems to register and work ok as far as playing MOH. I have been struggling to configure my Nokia E60 as a second extension (1001) . I have gotten that working to some extent by following the directions for the Nokia N series over at the WIKI (ie setting the realm on the handset to the IP address of the server). Now I am able to call the Nokia from the softphone and audio passes fine. When I attempt to call the softphone from e60, it fails with a no route. I see also that when it registers it appears to be deregistered because it doesn't respond to the notify? There is a similar item in the Jira, and Brian says to ignore it, which I am trying to do ;~) I am very familiar with the E60, so I don't feel that is the problem. Below are the messages from registration through the call: 2008-07-30 11:49:54 [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify 2008-07-30 11:50:56 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1001(a)10.0.1.18 [d8e095c4-ccb4-42de-bcc0-d57d0d24ac28] 2008-07-30 11:50:56 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1001->@default 2008-07-30 11:50:56 [WARNING] mod_limit.c:412 db_function() USAGE: db [insert|delete]/<realm>/<key>/<val> 2008-07-30 11:50:56 [NOTICE] switch_ivr.c:1097 switch_ivr_session_transfer() Transfer sofia/internal/1001(a)10.0.1.18 to XML[enum@default] 2008-07-30 11:50:56 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1001->enum@default 2008-07-30 11:50:57 [NOTICE] switch_ivr.c:1097 switch_ivr_session_transfer() Transfer sofia/internal/1001(a)10.0.1.18 to enum[enum@default] 2008-07-30 11:50:58 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-30 11:50:58 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/ 1001(a)10.0.1.18 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-07-30 11:50:58 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 28 (sofia/internal/ 1001(a)10.0.1.18) Ended 2008-07-30 11:50:58 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/ 1001(a)10.0.1.18 [CS_HANGUP] 2008-07-30 11:51:01 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1001(a)10.0.1.18 [5b4473aa-51c7-457b-a054-29ad9891e6ba] 2008-07-30 11:51:01 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1001->@default 2008-07-30 11:51:01 [WARNING] mod_limit.c:412 db_function() USAGE: db [insert|delete]/<realm>/<key>/<val> 2008-07-30 11:51:01 [NOTICE] switch_ivr.c:1097 switch_ivr_session_transfer() Transfer sofia/internal/1001(a)10.0.1.18 to XML[enum@default] 2008-07-30 11:51:02 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing 1001->enum@default 2008-07-30 11:51:02 [NOTICE] switch_ivr.c:1097 switch_ivr_session_transfer() Transfer sofia/internal/1001(a)10.0.1.18 to enum[enum@default] 2008-07-30 11:51:02 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-30 11:51:02 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/ 1001(a)10.0.1.18 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2008-07-30 11:51:02 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 29 (sofia/internal/ 1001(a)10.0.1.18) Ended 2008-07-30 11:51:02 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/ 1001(a)10.0.1.18 [CS_HANGUP] I am not really understanding why the E60 can't dial through to MOH or to the other extension? Any thoughts or ideas on this, or further troubleshooting measures? Thanks, Marty

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Re: [Freeswitch-users] Simple gateway config
by Simon Shaw 31 Jul '08

31 Jul '08

Thanks for that, got it working now. I notice that the SDP in the outgoing INVITE message is changed so that the media will flow via the trunk, is there any way to override this? Simon

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Re: [Freeswitch-users] Simple gateway config
by Simon Shaw 31 Jul '08

31 Jul '08

OK, commented out the lines of code in sofia_glue.c that run the stun test and now I see an INVITE message being sent to the trunk, however the To field is $1(a)10.10.7.95 instead of 58661 as I would have expected. 2008-07-31 19:26:19 [WARNING] sofia.c:75 sofia_handle_sip_r_notify() delete subscriptions for failed notify 2008-07-31 19:26:50 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)10.10.4.204 [bd3a342f-9c90-5842-9399-cb276b6146e7] 2008-07-31 19:26:50 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Simon FS->58661@default 2008-07-31 19:26:51 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/nat/$1(a)10.10.7.95 [e7312a7e-2b59-1449-8266-c5f098ae93f0] 2008-07-31 19:26:51 [NOTICE] sofia.c:2550 sofia_handle_sip_i_state() Hangup sofia/nat/$1(a)10.10.7.95 [CS_CONSUME_MEDIA] [NO_ROUTE_DESTINATION] 2008-07-31 19:26:51 [INFO] mod_dptools.c:1794 audio_bridge_function() Originate Failed. Cause: NO_ROUTE_DESTINATION 2008-07-31 19:26:51 [NOTICE] mod_dptools.c:1821 audio_bridge_function() Hangup sofia/internal/1000(a)10.10.4.204 [CS_EXECUTE] [NO_ROUTE_DESTINATION] 2008-07-31 19:26:51 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 2 (sofia/nat/$1(a)10.10.7.95) Ended 2008-07-31 19:26:51 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/nat/$1(a)10.10.7.95 [CS_HANGUP] 2008-07-31 19:26:51 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 1 (sofia/internal/1000(a)10.10.4.204) Ended 2008-07-31 19:26:51 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.10.4.204 [CS_HANGUP]

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Reaction on incoming SIP IM
by Alois Komenda 31 Jul '08

31 Jul '08

Is there any way to inform a module of incoming SIP Instant Messages (e. g. sent to a special user)? -- Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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