January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] Freeswitch-users Digest, Vol 31, Issue 12
by Ben Holtsclaw 05 Jan '09

05 Jan '09

- Sent from my Windows Mobile Smart Phone - ------- Original Message ------- From: freeswitch-users-request(a)lists.freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Sent: 1/5/09, 9:12:15 AM Subject: Freeswitch-users Digest, Vol 31, Issue 12 Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP response code in Freeswitch (Ognjen Seslija) 2. [dingaling] Gtalk accept friend invite (kriko) 3. Re: VXML support in Freeswitch (Stefan Knoblich) 4. Re: VXML support in Freeswitch (Michael Jerris) 5. Re: [dingaling] Gtalk accept friend invite (Michael Jerris) 6. Idea/Suggestion required (shehzad p) 7. Re: [dingaling] Gtalk accept friend invite (Gonzalo Servat) 8. Re: Idea/Suggestion required (Anthony Minessale) 9. Re: LDAP Integration (Vinicius Kobashi) ---------------------------------------------------------------------- Message: 1 Date: Mon, 5 Jan 2009 10:30:42 +0100 From: "Ognjen Seslija" <oseslija(a)gmail.com> Subject: Re: [Freeswitch-users] SIP response code in Freeswitch To: freeswitch-users(a)lists.freeswitch.org Message-ID: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, there is proto_specific_hangup_cause switch variable you can use for the cdr i.e. You can also use SIP messages number for a continue_on_fail action like: <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,486,503"/> Regards, Ognjen On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N <saigop(a)gmail.com> wrote: > Hi, > Is there any possibilities that Freeswitch may detect the SIP response > code from the IP media gateway. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > 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] Freeswitch-users Digest, Vol 31, Issue 12
by Ben Holtsclaw 05 Jan '09

05 Jan '09

- Sent from my Windows Mobile Smart Phone - ------- Original Message ------- From: freeswitch-users-request(a)lists.freeswitch.org To: freeswitch-users(a)lists.freeswitch.org Sent: 1/5/09, 9:12:15 AM Subject: Freeswitch-users Digest, Vol 31, Issue 12 Send Freeswitch-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of Freeswitch-users digest..." Today's Topics: 1. Re: SIP response code in Freeswitch (Ognjen Seslija) 2. [dingaling] Gtalk accept friend invite (kriko) 3. Re: VXML support in Freeswitch (Stefan Knoblich) 4. Re: VXML support in Freeswitch (Michael Jerris) 5. Re: [dingaling] Gtalk accept friend invite (Michael Jerris) 6. Idea/Suggestion required (shehzad p) 7. Re: [dingaling] Gtalk accept friend invite (Gonzalo Servat) 8. Re: Idea/Suggestion required (Anthony Minessale) 9. Re: LDAP Integration (Vinicius Kobashi) ---------------------------------------------------------------------- Message: 1 Date: Mon, 5 Jan 2009 10:30:42 +0100 From: "Ognjen Seslija" <oseslija(a)gmail.com> Subject: Re: [Freeswitch-users] SIP response code in Freeswitch To: freeswitch-users(a)lists.freeswitch.org Message-ID: <4468a6770901050130u61ee7602x3b57b6d2495e6fdd(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Hi, there is proto_specific_hangup_cause switch variable you can use for the cdr i.e. You can also use SIP messages number for a continue_on_fail action like: <action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,486,503"/> Regards, Ognjen On Mon, Jan 5, 2009 at 8:53 AM, Gopalakrishnan A.N <saigop(a)gmail.com> wrote: > Hi, > Is there any possibilities that Freeswitch may detect the SIP response > code from the IP media gateway. > > -- > Thank you with regards, > Gopal, > > > _______________________________________________ > 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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busy tone detection
by Gopal krishnan 05 Jan '09

05 Jan '09

Hi, Is it possible to have busy tone detection thru console dialing? -- Thank you with regards, Gopal,

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VXML support in Freeswitch
by Gopalakrishnan A.N 05 Jan '09

05 Jan '09

Does freeswitch support VXML? Is there any separate module for this. -- Thank you with regards, Gopal,

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(no subject)
by fidibus83 05 Jan '09

05 Jan '09

Hello! Can FS do something like an ACD? I have configure a Supportqueue like this: <extension name=supportqueue> <condition field=destination_number expression=^991$> <action application=set data=$${hold_music}/> <action application=answer/> <action application=fifo data=supportqueue in </condition> </extension> And my Agents are in a Call Group. How can I transfer the call to a free agent? I saw the example in mod_fifo. But my Agent cant wait in the queue all the time. Is it possible that the agents are called back when a call is waiting in the queue? Thanks. Regards, fidibus

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Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value.
by mszlazak＠aol.com 05 Jan '09

05 Jan '09

I see the following in that file: ? <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/> and thought that's what might need changing to: ? <X-PRE-PROCESS cmd="set" data="bind_server_ip=10.0.0.3"/> But your looking at the same variable I was and I'm guessing something else might be in order like: <X-PRE-PROCESS cmd="set" data="local_ip_v4=10.0.0.3"/> I'll see if either of these work unless you have a different suggestion. You and Brian also suggested a more specific approach in sip_profiles/*.xml and input the sip-ip and rtp-ip for the sofia profile. ??? <param name="sip-ip" value="$${local_ip_v4}"/>????? to???? <param name="sip-ip" value="10.0.0.3"/> ??? <param name="rtp-ip" value="$${local_ip_v4}"/>?????? to???? <param name="rtp-ip" value="10.0.0.3"/> What's the advantage to doing it in the sip_profiles\internal.xml file over the general way in vars.xml? ? ? -----Original Message----- From: Jason White <jason(a)jasonjgw.net> To: freeswitch-users(a)lists.freeswitch.org Sent: Sun, 4 Jan 2009 3:15 pm Subject: Re: [Freeswitch-users] How do I set my FS internal ip address to a "static" value. mszlazak(a)aol.com <mszlazak(a)aol.com> wrote: > > Ok, so is it? $${local_ip_v4} that needs changing somewhere? I see stuff in > vars.xml for external address changes but not for internal ip address > changes?? If you set $${local_ip_v4} in vars.xml it will determine which address FreeSWITCH binds to, at least in the default configuration. Addresses can be configured more flexibly in the SIP profiles. _______________________________________________ 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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How do I set my FS internal ip address to a "static" value.
by mszlazak＠aol.com 04 Jan '09

04 Jan '09

I believe there is a variable $${domain} that is used by FreeSwitch for it's internal IP address. This address was 10.0.0.2 but changes when I connect or disconnect things to my computer and I wanted to make it "static." The "Getting Started Guide" suggests this is do-able but it seems I have to change things in several places besides vars.xml like \sip_profiles\internal.xml and maybe other places. Is there some general way to set my internal domain? Thanks. ?

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Forward calls that need to be transcoded to a Asterisk box
by Ron McCarthy 04 Jan '09

04 Jan '09

Hi List, I have ready in places that we can run regex on the SDP and then determine if the call need to be transcoded we can then send a redirect / 302 to the user and have them contact say a Asterisk box which can then run transcoding for us. Now my question is how do I do that? My thinking is the call hits the dialplan and run regex on the SDP and redirect if needed. I only need to send some calls to the Asterisk box for transcode and most of the carriers I use accept g729, but some do not. Could I take that value in the SDP and then pass it to the diaplan so I can then add that information to my LCR query that I run, I do all this with the curl dialplan not the built in in LCR module. Any help would be great. Thanks!

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Conference
by Baskar 03 Jan '09

03 Jan '09

Hi, *Just i want to call the 1002==>9841799874 and i want to call another no 9840544078 and add in to the conference room * *First method i tried :* api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874(a)172.20.191.228) ====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Both caller and extension will be transfer into conference room Then i call the another caller api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9840544078(a)172.20.191.228)====> 1002 call another caller 9840544078 api uuid_transfer c564f668-6778-4e78-85f6-a78c9c67cfa4 -bleg 3001 ====>Transfer only the caller 9840544078 to conference room 3001 while calling the second no 9840544078 the first line get hold after the transfer of bleg they all come in to the conference room but the first line caller is in hold so we cant able to unhold the call through api we have to manually go the soft phone and unhold the call . but i need the unhold should be done in api what is the solution . *i have tried in another method also * api originate sofia/internal/1002%172.20.191.227 &bridge(sofia/default/ 9841799874(a)172.20.191.228)====>1002 call 9841799874 api uuid_transfer 04120c08-19e7-4881-8873-0834a11524ff -both 3001 ====>Transfer only the caller 9841799874 to conference room 3001 api uuid_kill 04120c08-19e7-4881-8873-0834a11524ff before i hangup i have transfer both the caller in to conference room so when i hangup the extension why my extension got hangup Can any one suggest me how can i try the conference are else correct where i am wrong. -- Warm Regards, N.Baskar

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Enabling SIP traces from the command line interface
by Jason White 03 Jan '09

03 Jan '09

The sofia loglevel command is very useful. However, even with sofia loglevel 9, I still don't get a SIP trace. I know all about shutting down FreeSWITCH and setting the environment variables to obtain a SIP trace and voluminous debugging output: # source this file in the current shell, then # run FreeSWITCH. At the FreeSWITCH prompt, # press F8, then make the call that requires debugging, and # capture the output. # type shutdown to end the FreeSWITCH session export SOFIA_DEBUG=9 export NUA_DEBUG=9 export NTA_DEBUG=9 export TPORT_DEBUG=9 export TPORT_LOG=1 this is fine for me, but for people in production for whom uptime is significant, I'm wondering whether there are plans to allow tracing to be dynamically enabled/disabled from the CLI, or whether this is possible now with options that I haven't yet discovered. Please don't interpret this as a complaint or as one of those nagging feature request posts! As mentioned, it doesn't affect my usage scenario - testing and experimentation at home on my desktop Linux box.

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