- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to get the content of SIP headers and sdp in dialplan?
by Jon Bruel 17 Sep '08

17 Sep '08

How do I read the contents of the SIP headers sdp in a dialplan?

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uPnP/NAT-PMP
by R. Eric Bennett 17 Sep '08

17 Sep '08

I don't recall seeing anything anywhere (while compiling, on the Wiki, etc) that would suggest that FreeSWITCH does uPnP (or NAT-PMP for that matter; this question is without regard to NAT Traversal religion) for port mapping. Is this a correct assumption? Does anyone know if Asterisk does do uPnP or NAT-PMP to map incoming ports to itself? I haven't touched Asterisk in a couple years so... This is mostly just my idle curiosity working, triggered by another question asked on the list... thanks, eric

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Port Forwarding
by Jair Santos 16 Sep '08

16 Sep '08

Hi, Can anybody inform where can I find in the wiki port forwarding configuration. I've been trying to find it with no success so far. thanks Jair Santos

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cant get session variable in javascript
by Gayatri Kulkarni 16 Sep '08

16 Sep '08

var cid = session.getVariable("caller_id_number"); returns: 2008-09-16 08:31:57 [ERR] readFile.js:36 mod_spidermonkey() TypeError: session. getVariable is not a function var cid = self.session.getVariable("caller_id_number"); returns: 2008-09-16 08:31:57 [ERR] readFile.js:36 mod_spidermonkey() TypeError: session. self is not defined. what would I be doing wrong? Regards, Gayatri Kulkarni ----- Whenever you find yourself on the side of the majority, it is time to pause and reflect.

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cant dial through registered gateway in windows 2k3
by xbipin 16 Sep '08

16 Sep '08

i have the following in my gateway.msl file kept under conf/directory/default/ <include> <user id="id"> <gateways> <gateway name="sip.server.com"> </gateway> </gateways> <params> </params> </user> </include> on freeswitch start it registers also and in my dialplan i have: <extension name="india"> <condition field="destination_number" expression="^919825967120$"> <action application="bridge" data="sofia/gateway/sip.server.com/$1"/> </condition> </extension> but when i call that number, its unable to bridge to the gateway and it says route not available -- View this message in context: http://www.nabble.com/cant-dial-through-registered-gateway-in-windows-2k3-t… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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H323 trunk between avaya acm and freeswitch
by sambasivarao Vemula 16 Sep '08

16 Sep '08

Hi , I want make a h323 trunk between avaya acm and freeswitch. Anybody have information for this trunk . Please let me know. Thanks&Regards Samba DISCLAIMER ========== This e-mail may contain privileged and confidential information which is the property of Persistent Systems Ltd. It is intended only for the use of the individual or entity to which it is addressed. If you are not the intended recipient, you are not authorized to read, retain, copy, print, distribute or use this message. If you have received this communication in error, please notify the sender and delete all copies of this message. Persistent Systems Ltd. does not accept any liability for virus infected mails.

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Sofia profiles
by Jonas Gauffin 16 Sep '08

16 Sep '08

Hello I haven't touched my sofia profiles in a long time. I now need to have two profiles for phones registering against FreeSWITCH: * One for phones that are on the same LAN as FS * One for phones that is outside the lan (having external IPs). I've made an attempt, but I cant get RTP working, so I've missed a setting or two. Can someone be kind and show me what the profiles should look like? Regards, Jonas

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Liverpie is not sending no parameters
by Adeel Ansari 16 Sep '08

16 Sep '08

Hi folks, I am trying Liverpie. Its working fine, but sends nothing to my webapp. Below is the output from Liverpie console. === *Liverpie FreeSWITCH PROXY - Now Accepts Connections. Mon Sep 15 02:38:37 UTC 2008: New Connection Mon Sep 15 02:38:37 UTC 2008: Resetting webapp state machine... Mon Sep 15 02:38:37 UTC 2008: Calling webapp... Mon Sep 15 02:38:37 UTC 2008: PROBLEM: Didn't get any response from webapp UNBIND* === Yes, thats true. I am not sending any response back to Liverpie. But should I not expect any information as request parameter from Liverpie, when it is making the first request to my web app? Thanks. -- Best, Adeel Ansari http://www.linkedin.com/in/adeelansari

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Google Talk
by Christian Jensen 16 Sep '08

16 Sep '08

Hi everyone, I am thinking of making it so that anyone could leave me voicemail just by adding my gtalk account as a buddy and calling me. Freeswitch would automatically accept the new buddy as well as record anything that came in. Anyone know how to do this? I plan on using Liverpie connected to C# via Generic HTTP Handler. The part I need help with is: a) can mod_jingle do the notification of a buddy request b) can mod_jingle alert me of a call and pick up and start recording. If you have a config (especially for a google for domains config) I would be forever in your debt! (figure of speech of course :) Thanks! Christian

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VAD and CN question
by Tamas 15 Sep '08

15 Sep '08

Hello, I've talked about CNG, VAD with Anthony and Brian already but I still have a mess in it and don't really know how it works. Could anybody explain the practical side of these magic options? 1. What does suppress_cng=true do exactly? (yep, I know it suppresses CNG, but what does this mean? Won't send out the CN/8000 rtpmap in SDP?) 2. What do VAD settings do in sofia profile? I've read http://wiki.freeswitch.org/wiki/Sofia.conf.xml    What do these options do in practice? Can FS detect silence (e.g. no RTP?) and than what can it do with? 3. I found undocumented options rtp_enable_vad_in, rtp_enable_vad_out for sofia which might be the per call settings for point 2. ? If this is correct, what will this do? <action application="set" data="rtp_enable_vad_in=true"/> <action application="set" data="rtp_enable_vad_out=true"/> <action application="bridge" data="sofia/....."/> 4. What will FS do when one side of the bridge supports CN but the other doesn't? E.g. softphone->FS->GW (sends CN packets) the softphone does not support CN Would it be possible to turn on comfort noise generation in FS for the side which cannot handle CN? 5. Is it possible to let FS detect silence and on one leg send CN packets insted of RTP? E.g. softphone->FS->GW (sends CN packets) the softphone does not support CN, but for saving bandwidth it would be good to have silence detection and in case of silence FS lets send CN instead of RTP. Sorry for my dump questions, I would like to understand the whole picture. I would like to add the clarification to FS wiki to lets other have less troubles (with endpoints on one side supporting VAD/CNG and endpoints on other side not supporting VAD/CNG, e.g. CallWeaver) Thanks in advance! Regards, Tamas

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