- FreeSWITCH-users - FreeSWITCH Mailing Lists

Odd sonus warning
by Greg Thoen 27 Aug '09

27 Aug '09

Hi, I have some DIDs from Bandwidth.com and when they call in I see this in the console: 2009-08-27 11:01:47 [WARNING] sofia_glue.c:2701 sofia_glue_negotiate_sdp() Hello, I see you have a Sonus! FYI, Sonus cannot follow the RFC on the proper way to send DTMF. Sadly, my creator had to spend several hours figuring this out so I thought you'd like to know that! Don't worry, DTMF will work but you may want to ask them to fix it...... Anything I should worry about? -- Greg

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need help with mod_xml_odbc
by Juan Backson 27 Aug '09

27 Aug '09

Hi, I tried to make install mod_xml_odbc and load it in freeswitch, but I am getting: 2009-08-28 00:46:55.848087 [CRIT] switch_loadable_module.c:871 Error Loading module /usr/local/freeswitch/mod/mod_xml_odbc.so **/usr/local/freeswitch/mod/mod_xml_odbc.so: invalid ELF header** I had to manually copy the .so file from the mod_xml_odbc dir to /usr/local/freeswitch/mod Does anyone know what is wrong? Thanks, JB

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upstream Registrar / Mirror proxy
by sylver_b 27 Aug '09

27 Aug '09

Hello, We would like to use FreeSwitch to test interoperability with our sip registrar platform - as our registrar doesnt handle NAT Traversal we would like to use FS as a mirror proxy . If a call goes through, FS should transparently handle nat traversal functionalities while forwarding the REGISTER/INVITE requests to our registrar server. Below some detailed requirements: Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards our upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request. Please let us know the best way to configure FS to achieve this type of configuration. Thank you

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Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
by Hristo Benev 27 Aug '09

27 Aug '09

Bojnour, I'll send a trace ASAP. What I see is that SIP header does not get updated -> regex is not true then it does not go to the peer. (I assume that is coming from kamailio config) I'm really interested to see the updates of the project. Thank you for the good tutorial. Hristo >-------- Оригинално писмо -------- >От: rod >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >До: freeswitch-users(a)lists.freeswitch.org >Изпратено на: Четвъртък, 2009, Август 27 09:41:05 EEST >Hi Hristo, > >I'm the author of this setup and wiki page. I did a lot of modifications >on this setup (alternative routing if failure essentially) but don't >have too much time to update the wiki. > >May you please send me an ngrep trace when you call 1000: > >ngrep -d any -nn -i '1000' port 5060 -W byline > >I will check what's happening. >Do you have an entry for 1000 in your mysql database ? > >regards, >rod > >Hristo Benev a écrit : >> >> It seems that the problem is on kamailio configuration. >> Will ask on their list. >> >> But to test i try to connect to my asterisk server and i receive 407 proxy authentication required. >> >> I have it setup as friend in asterisk, but still ??? >> >> Any ideas? >> >> Thanks, >> >> >> >-------- Оригинално писмо -------- >> >От: Hristo Benev >> >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >> >До: freeswitch-users(a)lists.freeswitch.org >> >Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST >> >> > I think that the problem is here: >> >------------------------- >> >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING >> >Dialplan: sofia/internal/1001(a)209.71.254.33 parsing [ROUTING->PEER_01] continue=false >> >Dialplan: sofia/internal/1001(a)209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false >> >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting >> >-------------------------- >> > >> >Actually Regex FAIL >> > >> >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success? >> >Here is my default.xml: >> >---------------- >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> >-------------------------- >> > >> > >> > >> > >-------- Оригинално писмо -------- >> > >От: Brian West >> > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >> > >До: freeswitch-users(a)lists.freeswitch.org >> > >Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST >> > >> > >We do not blindly follow 302's as that is a dangerous thing to do. >> >You have to process all 302's in the dialplan. >> >Set this on your sofia profile >> >You can set these variables sip_redirect_profile, >> >sip_redirect_context, >> >sip_redirect_dialplan, >> >When a redirect happens you get these variables - sip_redirect_contact_%d, >> >sip_redirected_to, >> >sip_redirect_contact_user_%d, >> >sip_redirect_contact_host_%d, >> >sip_redirect_contact_params_%d, >> >sip_redirect_dialstring_%d, >> >sip_redirect_dialstring, >> >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the >> >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems? >> > > >> > >> >_______________________________________________ >> >FreeSWITCH-users mailing list >> >FreeSWITCH-users(a)lists.freeswitch.org >> >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >http://www.freeswitch.org >> > >> >> _______________________________________________ >> FreeSWITCH-users mailing list >> FreeSWITCH-users(a)lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org >

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Can an Instant Message (IM) be sent to a cell phone with FS?
by Merle J. Ebbert 27 Aug '09

27 Aug '09

Can an Instant Message (IM) be sent to a cell phone with FS? Any of these: AOL Instant Messenger (AIM) Yahoo (Y!) Blackberry Messenger Google Talk Windows Live Messenger Thanks, Merle

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Timers/DTMFs During a Call
by delianSPAM 27 Aug '09

27 Aug '09

Hello Everybody! 1. Scenario. I am writing an IVR in Python that gets a destination from the calling party (party A) and then connects to the destination (party B). When the call is CONNECTED, I want to: - Receive DTMFs - Have a timer that can call a certain function in my script. The script will have to play a message to party A. - Have a timer that can call a certain function in my script. The script will have to drop the call. Please notice that I want to do the things after the two parties are connected, and not after I send the Invite to party B. 2. Problem. I will be happy to receive help on: - Which methods should I look for to implement this. 3. Details Here is how I connect the call currently: session.execute("bridge","sofia/internal/" + destination_number + "@domain.com") I have tried to create a timer callback function "my_method()" using: ivr_timer =threading.Timer(30,my_method) This never called the function "my_method()". Maybe I am wrong in using threading.Timer and the "bridge" application? Maybe I need to create a new thread and a new timer using the API of freeswitch, plus to use the "session.setInputCallback", plus use a conference rather than a bridge? Can you please provide any suggestions or examples? Thank you! Best Regards, Delian Tashev

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Define Condition in FreeSwitch
by Ahmed Munir 27 Aug '09

27 Aug '09

Hi, I'm newbie. How can we translate the asterisk's condition in freeswitch as listed below; 1. NoOp ("Remote Conference Call") 2. GotoIf ($[${LEN(${DIALSTR})}=0]?3:4) 3. Hangup() 4. NoOp(Finish if-CONFERENCE-430) Kindly reply soon. -- Regards, Ahmed Munir

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Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route
by Hristo Benev 27 Aug '09

27 Aug '09

It seems that the problem is on kamailio configuration. Will ask on their list. But to test i try to connect to my asterisk server and i receive 407 proxy authentication required. I have it setup as friend in asterisk, but still ??? Any ideas? Thanks, >-------- Оригинално писмо -------- >От: Hristo Benev >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route >До: freeswitch-users(a)lists.freeswitch.org >Изпратено на: Сряда, 2009, Август 26 22:02:13 EEST > I think that the problem is here: >------------------------- >2009-08-26 22:56:52.725878 [INFO] mod_dialplan_xml.c:315 Processing 1001->1000 in context ROUTING >Dialplan: sofia/internal/1001(a)209.71.254.33 parsing [ROUTING->PEER_01] continue=false >Dialplan: sofia/internal/1001(a)209.71.254.33 Regex (FAIL) [PEER_01] ${sip_h_X-ROUTE}(LOOKUP) =~ /PEER_01/ break=on-false >2009-08-26 22:56:52.728289 [INFO] switch_core_state_machine.c:136 No Route, Aborting >-------------------------- > >Actually Regex FAIL > >I'm not familiar, but is this stating that ${sip_h_X-ROUTE} should be PEER_01 for success? >Here is my default.xml: >---------------- > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >-------------------------- > > > > >-------- Оригинално писмо -------- > >От: Brian West > >Относно: Re: [Freeswitch-users] freeswitch as SBC and kamailio - no route > >До: freeswitch-users(a)lists.freeswitch.org > >Изпратено на: Сряда, 2009, Август 26 19:47:37 EEST > > >We do not blindly follow 302's as that is a dangerous thing to do. >You have to process all 302's in the dialplan. >Set this on your sofia profile >You can set these variables sip_redirect_profile, >sip_redirect_context, >sip_redirect_dialplan, >When a redirect happens you get these variables - sip_redirect_contact_%d, >sip_redirected_to, >sip_redirect_contact_user_%d, >sip_redirect_contact_host_%d, >sip_redirect_contact_params_%d, >sip_redirect_dialstring_%d, >sip_redirect_dialstring, >sip_redirected_byThen its up to you to process the redirect in your dialplan, If you don't set the >sip_redirect_context then it'll default to redirected context and XML as the dialplan./bOn Aug 26, 2009, at 11:37 AM, Hristo Benev wrote:HelloI followed the tutorialhttp://wiki.freeswitch.org/wiki/SBC_SetupI have following problem when I dial 1000 Kamalio reports 302, but freeswitch does not routeWhere to look for problems? > > > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users(a)lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org >

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Re: [Freeswitch-users] Dial phone number + extension?
by Michael S Collins 27 Aug '09

27 Aug '09

Sent from my iPhone On Aug 26, 2009, at 6:00 PM, "Lars Zeb" <larclap(a)yahoo.com> wrote: > Is there a way to dial an external 10-digit phone number, wait a > second or two after connecting, and then dial a 4-digit extension? > > From the CLI: originate sofia/gateway/mygw/1234567890 1234 As for the short pause you could put that in the Dialplan entry for ext 1234 or you could even do an inline Dialplan instead of just 1234 as the second arg to originate. originate sofia/foo/1234567890 sleep:2000,transfer:1234 inline Search the wiki for "inline dialplan" to learn more about this handy feature -MC > > > Thanks, Lars > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org

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Dial phone number + extension?
by Lars Zeb 27 Aug '09

27 Aug '09

Is there a way to dial an external 10-digit phone number, wait a second or two after connecting, and then dial a 4-digit extension? Thanks, Lars

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