September 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

How can I configure TO and FROM in the invite message
by Tzury Bar Yochay 17 Sep '09

17 Sep '09

Hi, Currently, the invite message looks as follows INVITE sip:1002@CLIENT_IP:5060 SIP/2.0 Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: "Extension 1001" <sip:1001@SERVER_IP>;tag=2rH67Q3aa1rpe To: <sip:1002@CLIENT_IP:5060> Is there a way to configure FS so the message will look like this: INVITE sip:1002@CLIENT_IP:5060 SIP/2.0 Via: SIP/2.0/UDP SERVER_IP;rport;branch=z9hG4bKgvD702De7e0Se Max-Forwards: 69 From: "Extension 1001" <sip:1001@ACCOUNT_DOMAIN>;tag=2rH67Q3aa1rpe To: <sip:1002@ACCOUNT_DOMAIN:5060> That is, at "From" having the account's domain name (e.g. sip:1001@example.com) instead of the server's IP address. and having the same at "To" thanks @tzury

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DB table sip_dialogs always empty
by kokoska rokoska 17 Sep '09

17 Sep '09

Hello, I have setted-up odbc-dsn on all my FreeSWITCH sofia profiles and based on logs FS connected to the dsns correctly. But during the calls I can't see any rows in sip_dialogs table. When I run "show channels" from console it works fine. All other tables are "populated and maintained" as I expect :-) I'm running FreeSWITCH on 64bit Centos 5.3 with Postgresql 8.1 DB... Could you, please, point what I'm missing, or where I'm wrong? Thank you very much! Best regards, kokoska.rokoska

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switch_core_io.c:118 sofia/internal/XXXXX has no read codec.
by Yehavi Bourvine 17 Sep '09

17 Sep '09

Hello, I have a problem when trying to put a call on hold: I get the above message and the call is disconnected. Any idea where to look for the source of the problem? One thing I've tried is limiting all phones to use only one codec, but it doesn't help... Thanks! __Yehavi:

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Re: [Freeswitch-users] FS Presence Implementation
by Jerry Richards 17 Sep '09

17 Sep '09

Also, is presence conveyed as any string? Or is presence a predefined list of status? Best Regards, Jerry -----Original Message----- From: Jerry Richards [mailto:jerry.richards@teotech.com] Sent: Tuesday, September 15, 2009 8:46 AM To: 'freeswitch-users(a)lists.freeswitch.org' Subject: FS Presence Implementation I would like to modify my SIP phone and my gateway to convey/exchange presence information. Could someone point me toward the FS presence documentation? I've seen bits and pieces. Also, I think presence can be communicated via more than one protocol. Thanks And Best Regards, Jerry

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local stream problem - moh
by Christian Löschenkohl 17 Sep '09

17 Sep '09

hello since installing the latest trunk 14894 my local streams / moh don't work anymore no config file has changed, the files are in place ---- show_local_stream outputs default,/opt/freeswitch/sounds/music/8000 moh/16000,/opt/freeswitch/sounds/music/16000 moh/8000,/opt/freeswitch/sounds/music/8000 ---- console prints 2009-09-17 01:15:01.496277 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.516281 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.516281 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.536282 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.536282 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.556286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.556286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.576286 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.576286 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.596283 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.596283 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.616261 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.616261 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.636304 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.636304 [ERR] switch_core_file.c:152 File [(null)] not created! 2009-09-17 01:15:01.656273 [WARNING] mod_local_stream.c:318 Unknown source local_stream://moh, trying 'default' 2009-09-17 01:15:01.656273 [ERR] switch_core_file.c:152 File [(null)] not created! ----- and yes, files are there ls /opt/freeswitch/sounds/music/8000 danza-espanola-op-37-h-142-xii-arabesca.wav partita-no-3-in-e-major-bwv-1006-1-preludio.wav ponce-preludio-in-e-major.wav suite-espanola-op-47-leyenda.wav ----- br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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originate command sofia behaviour
by Alberto Escudero 16 Sep '09

16 Sep '09

I will like to update the wiki to spell out clearly the differences between this three commands I have a IVR running in 4600 and the FS box has IP address 192.168.46.15 originate sofia/192.168.46.15/1001 4600 originate sofia/internal/1001(a)192.168.46.15 4600 originate sofia/internal/1001%192.168.46.15 4600 The first originate places a call as a external gateway, not until registered phone 1001 answers the call is transfer to 4600 The second and third originate command triggers extension 4600 Javascript IVR although 1001 has not answer Can anyone clarify me if this is the intended behavior also including the difference between % and @ /aep

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mod_radius_cdr: Multiple RADIUS messages for single call?
by email lists 16 Sep '09

16 Sep '09

Hello, Using the mod_radius_cdr module w/ Freeswitch, I am seeing duplicate RADIUS messages being generated for individual calls (sample messages for one call below). Looking at the "Acct-Unique-Session-Id" and "Acct-Session-Id" fields, it would appear that perhaps each call leg results in a pair of start/stop RADIUS messages; is this the expected behavior? If so, is there a way to disable RADIUS messaging for what I presume is the "ingress" or A leg of the call? Any leads would be appreciated. Thanks in advance. Vladimir Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "097c8472ff7bcec7" Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:25 2009 Acct-Status-Type = Start Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226(a)x.x.x.x" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "53f729e173e8c0a9" Timestamp = 1252604245 Request-Authenticator = Verified Thu Sep 10 10:37:57 2009 Acct-Status-Type = Stop Acct-Session-Id = "b0f387c0-35bd-e32c-971a-d79d026a8004" Freeswitch-Hangupcause = Normal-Unspecified User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Lastapp = "bridge" Freeswitch-Billusec = 32029926 Freeswitch-Callstartdate = "2009-09-10T10:22:00.259042-0700" Freeswitch-Callanswerdate = "2009-09-10T10:22:00.319197-0700" Freeswitch-Callenddate = "2009-09-10T10:22:32.349123-0700" Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "097c8472ff7bcec7" Timestamp = 1252604277 Request-Authenticator = Verified Thu Sep 10 10:38:02 2009 Acct-Status-Type = Stop Acct-Session-Id = "584f4573-46f7-655f-91d7-84cd59c9ec12" Freeswitch-Hangupcause = Normal-Clearing User-Name = "8135793256" Freeswitch-Src = "8135793256" Freeswitch-CLID = "sipp" Freeswitch-Dst = "14043297226(a)x.x.x.x" Freeswitch-Dialplan = "XML" Framed-IP-Address = 50.46.50.55 Freeswitch-Context = "public" Freeswitch-Source = "mod_sofia" Freeswitch-Billusec = 32049973 Freeswitch-Callstartdate = "2009-09-10T10:22:00.279044-0700" Freeswitch-Callanswerdate = "2009-09-10T10:22:00.289136-0700" Freeswitch-Callenddate = "2009-09-10T10:22:32.339109-0700" Acct-Session-Time = 32 NAS-Port = 0 Acct-Delay-Time = 0 NAS-IP-Address = 1.1.1.1 Acct-Unique-Session-Id = "53f729e173e8c0a9" Timestamp = 1252604282 Request-Authenticator = Verified

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reloadxml question
by Tihomir Culjaga 16 Sep '09

16 Sep '09

hi, I've build a custom module for FS and everytihng work well except reloadxml command :P... m'I missing something in my module? ... i used mod_skeleton as a template when i started. When i start the FS without my module reloadxml works fine ... as soon as i include my module within modules.conf.xml and start FS .. it hangs. So, it is definitelly up to the custom module ... but what can it be? freeswitch@l01freeswitch1> freeswitch@l01freeswitch1> freeswitch@l01freeswitch1> reloadxml nothing happens ... i have to kill freeswitch (kill -9) to get the shell. T.

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ATTN: Debian gurus and enthusiasts
by Michael Collins 16 Sep '09

16 Sep '09

If you are a Debian person and have experience creating .debs then read on... Frank Carmickle has graciously volunteered to assist with creating a FS deb config. I've heard that others have been doing something similar or are interested. If you are such a person then please email Frank and me off list so that we can organize the volunteers. Thanks! -Michael

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How to Process Invalid extension in FS
by Ahmed Munir 16 Sep '09

16 Sep '09

Hi, I'm newbie in FS. I want to know how to process invalid extension in FS? Because I want to prompt the IVR if invalid extension is dialled. Kindly advice me. -- Regards, Ahmed Munir

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