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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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Gateway setup
by Ilan Perez 31 Jul '08

31 Jul '08

Dear All, I am not sure who is an expert in PBX/ FS and who isnt but I dont know that I certainly am not Jin light of that I really need some help getting things moving with this system. Just before I start, I want to say that the wiki isnt the be-all and end all in terms of answers. I have read the wiki many times and steps for setup are left out. So people like myself struggle and fiddle without actually understanding what they are doing. Here is what I have and what I need help with I have 3 computers. 1. Server @ 192.168.43.111 2. Softphone @ 192.168.43.5 ext 1207 3. Softphone @ 192.168.43.6 ext 1208 4. And a Linksys SPA-3120 @ 192.168.43.3 5. My router/ default gateway is at 192.168.43.1 I have setup the router and the server fine. The two softphones have x-lite installed and are registering correctly and I can make call between them. I have also configure the public.xml file in the dialplan directory to accept my custom extension of 1208, 1207 <extension name="public_extensions"> <condition field="destination_number" expression="^([1-9][0-9][0-9][0-9])$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> The next step for my project is to get setup incoming and outgoing calls for the Linksys SPA 3120 Here is what I have done To the directory I have created a custom extension 1207, 1208 and sippura Sippura looks like this <include> <user id="1200"> <gateway name="sippura"> <param name="username" value="1200"/> <param name="password" value="1200"/> <param name="proxy" value="192.168.43.3"/> </gateway> </user> </include> To the default.xml file in the dialplan directory I have added this <extension name="Outbound - Linksys"> <condition field="destination_number" expression="^(1{0,1}\d{10})$"> <action application="set" data="effective_caller_id_number=12223334444"/>   <action application="bridge" data="sofia/gateway/sippura/$1(a)192.168.43.3"/> </condition> </extension> To the public.xml file in the dialplan directory I have added this <extension name="sippura"> <condition field="destination_number" expression="^(0283472006)$"> <action application="transfer" data="$1 XML default"/> </condition> </extension> To the sip_profiles directory I have added a file call sippura.xml This file contains <include> <gateway name="sippura"> <param name="username" value="1200"/> <param name="password" value="1200"/> <param name="extension" value="sippura"/> <param name="proxy" value="192.168.43.3"/> </gateway> </include> è I know I have setup my gateway correctly as I know how to do thisfrom previous attempts with other PBX software, but just for your information the Username, password and User ID is all 1200 What I need is someone to comment on my steps by sayingdont need this or you left out line xand please add the line in. I promise once this is sorted I will submit a compelte walkthrough to the wiki for all hopeless newbies like myself J Looking forward to your help Ilan Perez Sydney Australia

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Turn on call recording
by Ilan Perez 31 Jul '08

31 Jul '08

I was wondering if someone can tell me the command to turn on call recording. Ilan Perez Webmaster

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Some little help with regexp and extensions
by Diego Viola 31 Jul '08

31 Jul '08

Hi everyone, I have this in my default dialplan for dialing out: <extension name="outbound"> <condition field="destination_number" expression="^9(\d+)$"> <action application="bridge" data="sofia/gateway/teliax/$1"/> </condition> </extension> It works fairly well... but when I try to dial other extensions that start with "9", like 9996 (echo extension) I get this: http://pastebin.freeswitch.org/5026 and in my softphone I get "service or option unavailable". If I remove the "outbound" extension works. I know it might be something simple... but I'm not very familiar with regular expressions yet, so if you can help me with this I will be really thankful. Thanks everyone... FreeSWITCH rocks =D Diego

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Calling card billing for FS
by Alex Kinch 30 Jul '08

30 Jul '08

Hi gang, Are there any calling card billing packages out there that are compatible with Freeswitch? Obviously I've heard of a2billing, but a recent forum thread suggests that it's going to be a while until it's ported to work with FS (if ever). Thanks, Alex

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Having FreeSwitch using UDP only
by afro.systems＠gmail.com 30 Jul '08

30 Jul '08

Hi all, A newbie question, Is it possible to configure FreeSwitch to be using UDP only even fore the signaling? -- Tzury Bar Yochay

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