September 2008 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Hangup cause return false
by msp 18 Sep '08

18 Sep '08

Hi all, I bridge the call using bridge(session, newsession); After hangup i use the following line to get start time of the call: starttime=session.getVariable("start_stamp") ; This works fine in all case, except one, When called party hangup, "start_stamp" becomes false. is there any other method to do so?

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Lua Version
by Robert Clayton 17 Sep '08

17 Sep '08

All, Is Lua in FreeSwitch derived from 4.0, 5.0, 5.1, etc. I am hoping to implement LuaCOM and it differs with versions. Bob

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directory and mod_xml_curl
by Cristian Talle 17 Sep '08

17 Sep '08

Hi, I'm new to FS. Can anyone shed some more light or point me to a place to read on how one can use mod_xml_curl for dynamic directory lookup? My scenario is: I define a new contact somewhere outside of FS, I have a SIP client attempting to register with FS using the new contact info -> that in turn invokes a custom url binding but the information returned in the HTTP reply doesn't seem to make any difference at runtime: mod_xml_curl.c:194 xml_url_fetch() XML response is in /tmp/3da0973a-84ec-11dd-b435-211ade423d94.tmp.xml 2008-09-17 15:10:07 [WARNING] sofia_reg.c:1061 sofia_reg_parse_auth() can't find user [XXXX(a)a.b.c.d] The xml returned by the url is: <document type="freeswitch/xml"> <section name="directory"> <domain name="a.b.c.d"> <user id="XXXX"> <params> <param name="password" value="1234" /> </params> <variables> <variable name="user_context" value="default"/> </variables> </user> </domain> </section> </document> How does the xml new xml node get loaded in the directory structure in memory?

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Adding LuaCom
by Robert Clayton 17 Sep '08

17 Sep '08

All, What would be the process for adding LuaCOM (a windows COM library used by Lua) to the FreeSwitch build? Bob

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Re: [Freeswitch-users] How to get the content of SIP headers and sdp in dialplan?
by Jon Bruel 17 Sep '08

17 Sep '08

Mike, in my test setup, no tags are added to the To-header by the FreeSWITCH. The reason why I want to control it? Well, many phones uses the tag to relate an existing call to a NOTIFY message (send after the INVITE has been sent) with the same headers (and the same to-tag) as the INVITE. I'm not aware of how this functionality is described in the RFC, though. My aim is the check if I can use the FreeSWITCH to replicate a function known in BroadWorks. In this system, there are a number of plug-ins and web-interfaces which are used to control the SIP phone: Answer, put on hold and resume. The plug-in controls the application server through a XML interface, which in turn sends the NOTIFY to the phone. Thus the phone can be controlled without connectivity between the PC and the phone. These three states are controlled by a Notify message sent from the application server. So it would be useful to find a way of sending a Notify message to the phone from the FreeSWITCH. Other NOTIFY messages could be used for other useful things such as resetting the phone.

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Debugging
by Christian Jensen 17 Sep '08

17 Sep '08

I am about to embark on a sip debugging mission (did not coming in correctly). I am on windows. Other than setting "sofia loglevel 9" what other switches can I throw to get a butload of debut info? I can hop over to ubuntu if need be. Thanks!

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voicemail
by Jair Santos 17 Sep '08

17 Sep '08

Hello Can anybody explain how to activate the voice mail ? When calling one of the extensions it times out and send a busy signal. 2008-09-17 08:55:45 [INFO] mod_dptools.c:1789 audio_bridge_function() Originate Failed. Cause: NO_ANSWER thanks Jair Santos

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

17 Sep '08

Well, let me be more precise: In general, I want to be able to modify the SIP headers to my liking. I would like to add a tag to the To-header sent to the B-phone. This tag information can be used at a later stage to put the phone on/off hold from the switch by sending a NOTIFY with an header: Event: hold or Event:talk.

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re gexp help
by xbipin 17 Sep '08

17 Sep '08

with the following expression i cant dial any sipbroker number, eg: *011188888 "^(^\*\d+)$" -- View this message in context: http://www.nabble.com/regexp-help-tp19529525p19529525.html Sent from the Freeswitch-users mailing list archive at Nabble.com.

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profile problem
by Jonas Gauffin 17 Sep '08

17 Sep '08

Hello I got the following setup: Phone1 and FS behind NAT1. Phone2 behind NAT2 Phone1 -> FS -> NAT1 -> INTERNET <- NAT2 <- Phone2 Both phone1 and phone2 registers to FS. Phone1 on the internal profile and Phone2 on a profile that is identical to internal, except that external sip/rtp ips are set. Everything works when Phone2 calls Phone1 (speech in both directions) When Phone1 calls Phone2 I get only speech in one direction (from phone1 to phone2). Anyone got any idea why? //Jonas

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