July 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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

31 Jul '08

Both FS and the asterisk box I would like to trunk are in the same private network, I have no reason or ability, (IT guy will take weeks to open up a port on the firewall) to connect to a Stun server in my test environment. 1) Is there a way to disable this? I tried to disable it by changing internal.xml as shown below but even after reset it seem to have no effect. 2) Is UDP port 3478 the only port I need to open to get this to work in a real environment? This is why. On Jul 31, 2008, at 9:57 AM, Simon Shaw wrote: > sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org: > 3478 [Timeout] Brian West sip:brian at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>

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

31 Jul '08

Made the suggested change but the call is still failing. Here are the traces: 2008-07-31 17:54:51 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)10.10.4.204 [554b908a-7b88 -7e42-b4ec-6f9c5a57eac1] 2008-07-31 17:54:51 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Simon FS->58661@default 2008-07-31 17:54:52 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/nat/$1(a)10.10.7.95 [7ec08549-dcf2-b045-87f7-fff2d91061c5] 2008-07-31 17:55:08 [ERR] sofia_glue.c:480 sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org:3478 [Timeout] 2008-07-31 17:55:08 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 34 (sofia/nat/$1(a)10.10.7.95) Ended 2008-07-31 17:55:08 [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 17:55:17 [ERR] sofia_glue.c:480 sofia_glue_ext_address_lookup() Stun Failed! stun.freeswitch.org:3478 [Timeout] 2008-07-31 17:55:17 [NOTICE] mod_sofia.c:100 sofia_on_init() Hangup sofia/nat/$1(a)10.10.7.95 [CS_INIT] [DESTINATION_OUT_OF_ORDER] 2008-07-31 17:55:17 [INFO] mod_dptools.c:1794 audio_bridge_function() Originate Failed. Cause: DESTINATION_OUT_OF_ORDER 2008-07-31 17:55:17 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 36 (sofia/nat/$1(a)10.10.7.95) Ended 2008-07-31 17:55:17 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/nat/$1(a)10.10.7.95 [CS_HANGUP] 2008-07-31 17:55:17 [NOTICE] mod_dptools.c:1821 audio_bridge_function() Hangup sofia/internal/1000(a)10.10.4.204 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] 2008-07-31 17:55:17 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 35 (sofia/internal/1000(a)10.10.4.204) Ended 2008-07-31 17:55:17 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.10.4.204 [CS_HANGUP] ________________________________ Try "^5\d{4}$" /b On Jul 31, 2008, at 8:55 AM, Simon Shaw wrote: > "^5d{4}$" Brian West sip:brian at freeswitch.org <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users> From: Simon Shaw Sent: Thursday, July 31, 2008 4:55 PM To: 'Freeswitch-users(a)lists.freeswitch.org' Subject: Simple gateway config I am running FS of a windows box IP 10.10.4.204 and have successfully managed to demo most of the default features. I attempted to extend the configuration by adding a gateway as follows in $(prefix)\conf\dialplan\default.xml, where 10.10.7.95 is the address of a working asterisk test box I have setup (I ran the reloadxml command from the console after updating the file):  <extension name="Staging"> <condition field="destination_number" expression="^5d{4}$"> <action application="bridge" data="sofia/${use_profile}/$1(a)10.10.7.95"/> </condition> </extension> When I dial 58661 (8661 being the extension of a working phone on the asterisk box) I get the following traces in the console, however no INVITE message is sent to 10.10.7.95. 2008-07-31 16:50:44 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)10.10.4.204 [54b22243-26c5 -1c41-a59d-27ae7547df1e] 2008-07-31 16:50:44 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Simon FS->58661@default 2008-07-31 16:50:45 [NOTICE] switch_ivr.c:1097 switch_ivr_session_transfer() Transfer sofia/internal/1000(a)10.10.4.204 to enum[58661@default] 2008-07-31 16:50:55 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-31 16:50:55 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/1000(a)10.10.4.204 [CS_ROUTING] [NO_ROUTE_DES TINATION] 2008-07-31 16:50:55 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 45 (sofia/internal/1000(a)10.10.4.204) Ended 2008-07-31 16:50:55 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.10.4.204 [CS_HANGUP] What am I missing, Thanks, Simon

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Simple gateway config
by Simon Shaw 31 Jul '08

31 Jul '08

I am running FS of a windows box IP 10.10.4.204 and have successfully managed to demo most of the default features. I attempted to extend the configuration by adding a gateway as follows in $(prefix)\conf\dialplan\default.xml, where 10.10.7.95 is the address of a working asterisk test box I have setup (I ran the reloadxml command from the console after updating the file):  <extension name="Staging"> <condition field="destination_number" expression="^5d{4}$"> <action application="bridge" data="sofia/${use_profile}/$1(a)10.10.7.95"/> </condition> </extension> When I dial 58661 (8661 being the extension of a working phone on the asterisk box) I get the following traces in the console, however no INVITE message is sent to 10.10.7.95. 2008-07-31 16:50:44 [NOTICE] switch_channel.c:534 switch_channel_set_name() New Channel sofia/internal/1000(a)10.10.4.204 [54b22243-26c5 -1c41-a59d-27ae7547df1e] 2008-07-31 16:50:44 [INFO] mod_dialplan_xml.c:222 dialplan_hunt() Processing Simon FS->58661@default 2008-07-31 16:50:45 [NOTICE] switch_ivr.c:1097 switch_ivr_session_transfer() Transfer sofia/internal/1000(a)10.10.4.204 to enum[58661@default] 2008-07-31 16:50:55 [INFO] switch_core_state_machine.c:114 switch_core_standard_on_routing() No Route, Aborting 2008-07-31 16:50:55 [NOTICE] switch_core_state_machine.c:115 switch_core_standard_on_routing() Hangup sofia/internal/1000(a)10.10.4.204 [CS_ROUTING] [NO_ROUTE_DES TINATION] 2008-07-31 16:50:55 [NOTICE] switch_core_session.c:807 switch_core_session_thread() Session 45 (sofia/internal/1000(a)10.10.4.204) Ended 2008-07-31 16:50:55 [NOTICE] switch_core_session.c:809 switch_core_session_thread() Close Channel sofia/internal/1000(a)10.10.4.204 [CS_HANGUP] What am I missing, Thanks, Simon

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FreeSWITCH 1.0.1 "Phoenix" Released!
by Anthony Minessale 31 Jul '08

31 Jul '08

The first round of improvements are in from our 1.0 line! There are several new features including: Open Source ASR with mod_pocketsphinx Open Source TTS with mod_flite Improved ACL with user matching. Enhanced SNOM support. And many more! Thanks for all the beta testing and feedback! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ 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

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MPL in simple terms
by Simon Shaw 31 Jul '08

31 Jul '08

Is there a simple explanation, in terms that a mere software engineer can understand, what the MPL license means? 1) Can I sell FS under my own guarantee and support? 2) If I write code that connects FS to an external application that I have written, does the external application have to be MPL as well? 3) Extension to previous question: What if the external application is Active Directory? Thanks, Simon

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