February 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Re: [Freeswitch-users] can vm settings go in conf\directory\default\default.xml?
by Paul Levin 14 Feb '10

14 Feb '10

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Fw: inbound outbound transparent proxy question
by Madovsky 14 Feb '10

14 Feb '10

Hi, I'm new to freeswitch world and apologize if my question is not relevant. I need to install freeswitch as inbound outbound transparent proxy (media only if possible, but not sure it's the right terms to use) example : Caller A (he can be anonymous with his own provider) connect with a softphone we provide in my network (nat and public) so he sets the softphone as user callerA, domain iptel.org (which is not my domain) proxy 192.168.0.1 (which is my FS proxy). So he calls an external user (who is unknown) like callerB(a)bobo.dot. So callerB receive the invite (until now he can see the caller user name but with proxy ip subsituted from the domain (but I want the original caller domain kept intact) and the call is established. But, for the contrary (callerB wants to call callerA), it doesn't work. Is anyone have idea or example of this kind of configuration ? Best Regards Franck Chionna

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inbound outbound transparent proxy question
by Madovsky 14 Feb '10

14 Feb '10

Hi, I'm new to freeswitch world and apologize if my question is not relevant. I need to install freeswitch as inbound outbound transparent proxy (media only if possible, but not sure it's the right terms to use) example : Caller A (he can be anonymous with his own provider) connect with a softphone we provide in my network (nat and public) so he sets the softphone as user callerA, domain iptel.org (which is not my domain) proxy 192.168.0.1 (which is my FS proxy). So he calls an external user (who is unknown) like callerB(a)bobo.dot. So callerB receive the invite (until now he can see the caller user name but with proxy ip subsituted from the domain (but I want the original caller domain kept intact) and the call is established. But, for the contrary (callerB wants to call callerA), it doesn't work. Is anyone have idea or example of this kind of configuration ? Best Regards Franck Chionna

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Re: [Freeswitch-users] Bypass-media and REFER method
by Anthony Minessale 14 Feb '10

14 Feb '10

We don't support the series of 100,180,200 in the notifies that is typically a pure sip pbx feature. We are a b2b and protocol agnostic softswitch. When you do refer to a sofia leg you are talking to that leg independantly. We do not track the progress of what the channel does once you transfer it, instead we send the channel back to the dialplan and accept the refer. This operation if sucessful will generate 202. The resume media on hold and bypass after att xfer sofia options or bypass_media_after_bridge var may be what you need if you want no media on fs. We have to make some sacrifices on sip madness to gain some general flexibility and call volume. =/ On Feb 14, 2010 3:39 AM, "Victor Maruani" <vmaruani(a)interwise.com> wrote: Hi, I would say it fails in 2 points: First in the fact that a "NOTIFY 200 OK" (line 1070) is sent right after FS gets the REFER. Then in the REINVITE (line 1094) sent to A (10.10.5.19) just after this NOTIFY, This REINVITE contains the SDP of the FS (10.10.5.92) causing the A side to send media to FS. There will be no REINVITE with SDP of C (10.10.5.48) But as you say, just afterwards, the REFER action is actually done and C is invited by FS with the SDP of A. Conclusion : 1) B is notified of success just after it sent the REFER and is disconnected. B may be notified of every step of the connection to C (100 trying… 200 OK) when these actually happen. What if C is down? Can't FS notify a failure? (didn't test that.) 2) 'A' gets to send media to FS Because of a REINVITE which disconnect him from B (we are in bypass media mode) . during the process of REFER, A should be still connected to B from a media perspective. The REINVITE is not done at the right time with the right params. Here, a pseudo bridge (on way voice) is established when C gets the INVITE and is sending media to A. A can hear C but C can't hear A after the REFER. If C was down, A would be "lost" in FS… I believe the correct behavior would be: B sends REFER. FS INVITE C C replies 100, 180… 200 and FS notifies B in accordance. Once C has sent 200 OK with its SDP. B is disconnected and A is updated (REINVITE) with C's SDP. Please share your thoughts, I still don't know if it's a bug or if I configured something wrong although I don't think so. Hasn't anyone done that before? Thank you. *From:* freeswitch-users-bounces(a)lists.freeswitch.org [mailto: freeswitch-users-bounces(a)lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Friday, February 12, 2010 1:02 AM To: freeswitch-users(a)lists.freeswitch.org Subject: Re: [Freeswitch-users] Bypass-media and REFER me... Where do you think it's failing? that log shows it get refer, go back to dp, invite to 3341 and... _______________________________________________ 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] can vm settings go in conf\directory\default\default.xml?
by Paul Levin 14 Feb '10

14 Feb '10

2 2

Is it possible to repeat music in playback
by Jingwei Yang 14 Feb '10

14 Feb '10

Hello, I've defined a very simple dialplan like the one below and when the caller is connected to this plan, I hope to keep the call alive and repeat the music set by playback. How am I able to achieve this? <extension name="agent_hold"> <condition field="destination_number" expression="^33333$"> <action application="answer"/> <action application="playback" data="customized/test.wav"/> </condition> </extension> Thanks, -Jingwei

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Other party does not hear voice when connecting with X-Lite
by Vitalii Colosov 14 Feb '10

14 Feb '10

Hi, I have installed FS default configuration and setup an external gateway for international calls. When I call to international phone using YATE windows client, both parties able to hear voice... erevything is OK. But when I connect using X-Lite, it connects but other party cannot hear me. I can hear him well... Could you please hint in which direction should I look? Thank you, Vitalii Colosov

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Bypass-media and REFER method
by Victor Maruani 14 Feb '10

14 Feb '10

Hi, I'm trying to do a POC using FS, the goal is to have FS handle REFERs containing proprietary data. I want to have some logic on top of FS and also use the fail over mechanism. in short, I have something like this: (third party) A side --- FS ---- B side (IVR server) the IVR the sends a REFER to FS. I don't want A to deal with it. now say B refers to C, it would be considered as a "group" C1, C2 ... to which I want FS to failover. only when one has answered should A be updated (REINVITE) and B notified and disconnected. if all fails I would expect B to be notified of the failure and proceed as I wish without "losing" A. from what I've read FS should be OK for the job but I have a couple issues: 1 ) I have some issues getting FS handle a REFER while in bypass-media mode. (I tried with the release and some revisions including latest) first when I bridge A and B everything is fine and media is bypassed. When B sends REFER to C: - FS immediately NOTIFY B of success and send a reinvite to A with SDP containing its own media IP/port. - then it does INVITE C with A's SDP. - B gets disconnected. A is not updated with C's sdp. so at this point A sends RTP to FS and C sends RTP to A. ... I basically have one extension for B: (set bypass-media and bridge to B) and another extension to C which does the same actions. what do you think I do wrong? 2 ) how can I catch the REFER and set variables from it? (like ref-by or ref-to) in the dial plan I do catch the INVITE sent to C, but how to do it with the REFER itself? thanks for your help! Best Regards, Victor.

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Send DTMF after call bridge
by Eder Souza 14 Feb '10

14 Feb '10

Hi How i can send DTMF digits after call bridge ?? Example:: <action application="bridge" data="*openzap*/2/1/1831"/> After my ramal 1831 answer i want send DTMF's how i can make this?? Att, Eng Eder de Souza

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Re: [Freeswitch-users] Mod_Conference capacity....
by Joel Sisko 13 Feb '10

13 Feb '10

Brian/Gavin thanks for the input. But I agree with Brian, if were that easy I would have done it prior to the post. Just looking to find out what some of the communities success has been to see if this is a path we should go down for a conference solution platform. Joel ----- Original Message ----- From: "Brian West" <brian(a)freeswitch.org> To: freeswitch-users(a)lists.freeswitch.org Sent: Thursday, February 11, 2010 3:47:46 PM GMT -08:00 US/Canada Pacific Subject: Re: [Freeswitch-users] Mod_Conference capacity.... Not an optimal test scenario unless you know wtf you're doing! /b On Feb 11, 2010, at 5:22 PM, Gavin Henry wrote: > Why don't use script a test or use sipp and then dial in yourself to listen? > > Cheers. _______________________________________________ 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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