- FreeSWITCH-users - FreeSWITCH Mailing Lists

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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The transferred call is not hanged up.
by Sangwoo Jin 03 Jan '09

03 Jan '09

I have met the non-hanged up call after being transferred. The following is the test scenario. 1. A makes a call to B 2. B answers the call of A and holds the call of A. 3. B makes a call to C and transfers the call of A to C before C answers the call of B. 4. A hangs up the call before C answers the call of A. In this state, C's phone is ringing endless. Why is not the channel of C hanged up? Thanks for reading my mail. Sangwoo Jin.

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SCTP Support
by Calvin Paul 03 Jan '09

03 Jan '09

Hello, I need some help to run a sip or iax trunk between two freeswitch boxes using the SCTP transport protocol. I am looking at the documentation and this list and I am pretty sure there is nothing covering SCTP. I am interested in doing this using the freeswitch package in pfsense but I will take anything I can get. The only mention I can find of SCTP is in the features list, an instruction to compile freeswitch with SCTP support and a reference to use TLS with SCTP. Any and all help will be welcomed. PS Can encryption be done on IAX? Paul

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mod_event_multicast TTL?
by Kristian Kielhofner 02 Jan '09

02 Jan '09

Hey FS Devs/Users, Happy New Year! Is it just me or should mod_event_multicast allow you to specify a TTL on the outbound packet? I know the wiki says you should use VLANs (which I usually do anyway) but the more "standard" way of restricting multicast packets is to set the TTL extremely low (1, etc). I didn't want to be another "gimme gimme gimme" user; I tried looking through the code but I think even my neighbors could here the "whooshing" sound as it went over my head... Thoughts? -- Kristian Kielhofner http://blog.krisk.org http://www.submityoursip.com http://www.astlinux.org http://www.star2star.com

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bind_meta_app with sip info dtmf in bypass media mode
by Gaëtan Minet 02 Jan '09

02 Jan '09

Hi, It looks like that even when using sip info dtmf, the bind_meta_app doesn't react on them when rtp is in bypass media mode. Of course rfc2833/inband cannot be processed in bypass media mode, but doesn't sip info still go through freeswitch ?. Is this a limitation of the current implementation, or simply not possible due to some technical limitations ? Or is there any option to control this ? I'd like to implement (dtmf-triggered) server-side features but still use bypass media mode. Thank you Kind regards Gaetan

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MOH and bypass media
by Gaëtan Minet 02 Jan '09

02 Jan '09

Hi, As a newcomer on Freeswitch, first of all thanks you for all your work on it. I'm currently evaluating it as a replacement for * and really appreciate working (ok, playing) with it. One of our requirements is to optimize the network flows as much as possible, and as such avoid any unnecessary rtp proxying. When a call is in bypass media and one of the phones puts the call on hold, the remote phone doesn't play moh. Of course I guess this is expected in bypass media, and moh should be the responsibility of the UA just like in a pure sip proxy environment ( so the phone that holds the line should for example invite the moh server in a new dialog and swap sdp with the remote phone or something similar). Snom phones do this just fine when you configure the moh server uri in the phone, but others like Thomsons ST2030 do not have that setting (... I know, real crap, but we already deployed tons of them so we cannot ignore them...) . So when a Thomson puts another phone on hold, there is no music at all. I'd like to achieve something similar to what asterisk does with (re)invites during the hold. I read about and tried the resume-media- on-hold option and this works fine, the phone correctly gets the moh. However, when the call is unheld, the media stream continues in media mode and I'd like to avoid this as much as possible. Is there a way to return to bypass media mode automatically when the call is unheld ? BTW, is the resume-media-on-hold really implemented for server moh support, or as a more generic way to put a call back in media mode (so we can listen for inband/rfc dtmf for example) ? I also noticed that the remote phone only sees/display the hold state when in bypass media mode. I guess no reinvite (a=sendonly) at all is sent to the held party when in media mode ? Again, is there a solution for this ? Many thanks ! Kind regards Gaetan

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fs_cli help command
by Jason White 02 Jan '09

02 Jan '09

I just noticed, as confirmed by reading the code, that now when the user types "help" at the fs_cli prompt, the fs_cli help text is printed; but if what one really wants is to execute the API help command, there doesn't seem to be any way to do it. process_command() gets the help command first, and there's no way to have it passed to FreeSWITCH as an API command. Here are a few options for solving this (I'm sure there are others): 1. Make the API command processing into a separate function, and have the "help" command call it to execute "api help" after printing its own text. I suppose one could also use a goto for this, but that could be accused of inelegance. 2. Rename the fs_cli help command to "/help" - probably confusing to new users! 3. Add a /help command that runs "api help". I'm leaning toward option 3, but opinions may differ.

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