April 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_skinny - Building with VS 2008 Express
by Dave Stevenson 27 Apr '10

27 Apr '10

Hi, I'm trying to use mod_skinny under windows but can't get the Visual Studio 2008 Express build configuration working. I know that Mathieu does not run Windows, but can anyone else modify the VS build solution in source to include building mod_skinny please ? regards Dave

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Header url encoding in Free SWITCH Event
by M. Ranganathan 27 Apr '10

27 Apr '10

Hello, When SIP Headers are passed in FreeSWITCH Events, is the entire header expected to be URL encoded (including things like Display Name) ? Thanks Ranga -- M. Ranganathan

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Freeswitch and dynamic application
by Martrès Jean-Jacques 27 Apr '10

27 Apr '10

Hi guys, I need to build dynamic application using freeswitch. Based on your experiences what is the best solution ? Using a LUA script or use the event_socket ? Thanks for your help. Regards. -- MARTRES Jean-Jacques email : jjmartres(a)networkconcordance.com website : http://www.networkconcordance.com

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need help on IVR
by false 27 Apr '10

27 Apr '10

Hi all my network topology: endpoint 1(100)-----sip server ---IVR(Freeswitch) | | endpoint2(101) endpoint1 + endpoint2 are registered to sip server Freeswitch is regsitered to sip server with 103 my wish is when endpoint 1 calls to freeswitch then endpoint 1 hear IVR and RTP from endpoint 1 --> media proxy---> FS then endpoint1 press 101, freeswitch will send INVITE 101 to sip server via call transfer feature of FS and RTP from endpoint1--> media proxy -->endpoint1, RTP will not go through the FS after FS transfer call to 101 can FS do it Thank you __________________________________________________ Bạn Có Sử Dụng Yahoo! Không? Mệt mỏi vì thư rác? Yahoo! Thư có chương trình bảo vệ chống thư rác hữu hiệu nhất trên mạng http://vn.mail.yahoo.com

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understanding effective_caller_id_number and other variables before bridge
by RR 27 Apr '10

27 Apr '10

Hello All, just want to clarify my understanding re: variables etc that can be set before a call bridge/transfer etc. It appears that we need to set these variables immediately before the bridge without any other operation occurring from when they are setup and when the bridge happens. Is that true? I have a very simple dialplan like so: <include> <extension name="DIDtest"> <condition field="ani" expression="^\+?|+1?|1?(\d+).*$"/> <action application="export" data="effective_caller_id_number=$1"/> <condition field="destination_number" expression="^\+?+1?1?(.*)$"> <action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/blade2/$1"/> </condition> </extension> </include> But when the call is sent to gateway blade2, the callerID on the call is whatever the original callerID was with all the 1s and '+'s and whatever prefixes the call came in with. Questions: a) can the variable effective_caller_id_number not be set dynamically from a temp variable such as $1. I have also tried forcing it to a number like effective_caller_id_number=20239388383 and even then it doesn't work b) I have also tried <action application="bridge" data="{effective_caller_id_number=20239388383}{sip_invite_domain=${sip_from_host}}sofia/gateway/blade9/$1"/> and that doesn't work either. Any ideas? Thanks RR

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Spanish sound voices on freeswitch 1.0.6
by joseba 27 Apr '10

27 Apr '10

I see that on information about freeswitch 1.06, but I dont find this. Only EN and RU voices. Is this an error? How can I find spanish voices? Thx JuanjoA

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What to do after 1.0.6 has been released?
by Yehavi Bourvine 27 Apr '10

27 Apr '10

Hello, I would like to know what other people are doing in this situation: We've upgraded to 1.0.6 which is very solid. At the last 2-3 months before 1.0.6 has been released I did a weekely upgrade to the latest trunk. What to do now? The stability is very important, but I do not want to open too large gap from 1.0.6 to the deveopment version. What do you suggest? Thanks! __Yehavi:

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sofia.c:896 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer
by Armin Theis 27 Apr '10

27 Apr '10

During a FreeSwitch call a sofia media error occured (sofia.c:896 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer) I am using FreeSwitch Stable Release 1.0.6 and want to use it for production. Does anyone know why this error comes? 2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:3585 Audio Codec Compare [PCMA:8:8000:20]/[PCMA:8:8000:20] 2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:2292 Already using PCMA 2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:3524 Set 2833 dtmf send/recv payload to 101 2010-04-27 10:02:38.421875 [DEBUG] sofia.c:4604 Processing updated SDP 2010-04-27 10:02:38.421875 [DEBUG] sofia_glue.c:2579 Audio params are unchanged for sofia/internal/1010(a)192.168.1.234. 2010-04-27 10:03:39.328125 [DEBUG] sofia.c:4153 Channel sofia/internal/1010(a)192.168.1.234 entering state [calling][0] 2010-04-27 10:03:39.484375 [DEBUG] sofia.c:896 nua_i_media_error: unknown event 22: 988 Incomplete offer/answer 2010-04-27 10:03:39.484375 [DEBUG] sofia.c:4153 Channel sofia/internal/1010(a)192.168.1.234 entering state [terminating][200] 2010-04-27 10:03:39.484375 [NOTICE] sofia.c:4789 Hangup sofia/internal/1010(a)192.168.1.234 [CS_EXECUTE] [NORMAL_CLEARING] 2010-04-27 10:03:39.484375 [DEBUG] switch_channel.c:2102 Send signal sofia/internal/1010(a)192.168.1.234 [KILL] 2010-04-27 10:03:39.484375 [DEBUG] switch_core_session.c:1021 Send signal sofia/internal/1010(a)192.168.1.234 [BREAK] 2010-04-27 10:03:39.500000 [ERR] switch_rtp.c:2395 switch_rtp.c if (switch_rtp_ready(rtp_session)) ...

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Add a diversion header?
by Ken Fulmer 27 Apr '10

27 Apr '10

We have customers using a sipX internal PBX and a Freeswitch device as a softswitch to route calls to our upstream provider, PaeTec. However, the provider doesn't support the P-Asserted Identity for external call forwarding. PaeTec wants to see a SIP Diversion header instead. How can we add this header in the Freeswitch device? Thanks, Ken Fulmer

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Calling Web service from Freeswitch
by Alain MELIOT 27 Apr '10

27 Apr '10

Hi All New to freeswitch i rewrite an asterisk application to freeswitch. I have a javascript application where i collect some user data and i must send the data to a web service (wsdl) for processing. The problem is that i have no idea how to call the web service from curl or javascript. Any help will be welcome. Thank in advance Alain MELIOT

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