January 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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ACD - Queue and Call Group
by fidibus83 02 Jan '09

02 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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encryption infos needed
by excelsio＠gmx.net 02 Jan '09

02 Jan '09

Hi, we want to enhance our old Siemens Hicom 300 and replace it step by step. Therefore we decided to try out opensource solutions ourselves. One requirement is that the solution has to encrypt all data. So try let´s look at Asterisk was our first thought. Well, there seem to be unoffical patches for Asterisk 1.4.x with SRTP/SIPS support. So, unofficial. With 1.6.x the support for it hasn´t been fully integrated, yet. So, what´s next out there? => freeswitch But what about encryption support? SRTP is end to end encryption between phones, SIPS is used for the encryption of signaling "hop-by-hop", well which hop? Talking about encryption, it seems there are many different scenarios to consider, which looks like they couldn´t encrypted? Let´s look at our planed setup public telefon network <--ISDN/S2M--> Patton 4960 <--ISDN/S2M--> Siemens Hicom 300 Patton 4960 <--IP--> freeswitch <--IP--> Snom 320 SIP Provider <--IP-- freeswitch <--IP-- Snom 320 1. Incoming calls shoud be reached via landline: [e.g. telefon network --ISDN/S2M--> Patton 4960 --IP--> freeswitch --IP--> Snom 320 users] So, what about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole path? Well, how? Is it supported with freeswitch? 2. Outcoming calls should go to a SIP provider which supports sip trunking and DDI, well SIPconnect: [e.g. SIP Provider <--SIP trunk-- freeswitch <--IP-- Snom 320 users] Same question here: What about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole path? Well, how? Is it supported with freeswitch? 2.1 Outcoming calls should be forwarded locally, if the SIP trunk between the SIP provider and the freeswitch server fails [e.g. telefon network <-- ISDN/S2M-- Patton 4960 <--IP-- freeswitch <--IP-- Snom 320 users] Same question here: What about encryption between the Patton 4960, the freeswitch and der Snom 320? Is it possible to encrypt the whole IP path? Well, how? Is it supported with freeswitch? 3. The next thing is the encryption of voice and signaling data in general. Does the freeswitch solution support this? I think it´s an end to end encryption between the users? As freeswitch seems to play a proxy part, I guess yes? [e.g. freeswitch <--IP--> Snom 320 users <--SRTP/SIPS --> Snom 320] 4. Another problem is the encryption of the voice and signaling data between our LAN and the SIP provider. Is it possible to encrypt all data between those with the freeswitch solution? Do I need something additionally? [e.g SIP Provider <--encrypted SIP trunk ??? --> freeswitch] So what can be done with freeswitch? What else can be done support all scenarios above? -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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uuid_playback
by Peter P GMX 01 Jan '09

01 Jan '09

As I see on the Wiki page uuid_playback seems to be implemented, however it doesn't work on the console or via event_socket. Also in the code I could not find it (svn 10438). So for now I use uuid_brodcast to play announcements to one or both parties. Question: What is the status of uuid_playback? Best regards Peter

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Re: [Freeswitch-users] Direct inward dialling
by excelsio＠gmx.net 01 Jan '09

01 Jan '09

Hallo Peter, could you please post your complete working config for DID and QSC? thanks in advance Tim -- Psssst! Schon vom neuen GMX MultiMessenger gehört? Der kann`s mit allen: http://www.gmx.net/de/go/multimessenger

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