July 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Conference without members
by Samuel Macedo 18 Jul '10

18 Jul '10

Hi, I've experienced a strange behaviour in one of my conferences rooms. It's a conference with a lot of users and monitors - that can kick the users - witch was working fine. What is happening now is that anyone can hear anything in this conference. And when everybody leaves the conference, the conference still "alive" in the freeswitch - like above: freeswitch@internal> conference 1_conference xml_list <?xml version="1.0"?> <conferences> <conference name="1_conference" member-count="0" rate="8000" destruct="true" running="true" answered="true" enforce_min="true" dynamic="true"> <members></members> </conference> </conferences> There is no member in the conference, and when two members got into there none can hear. Do you have any idea about what is happening? Thanks, -- Samuel Macedo

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How to change SIP To header
by Eugene Prokopiev 17 Jul '10

17 Jul '10

Hi, Is it possible to change SIP To header from default: To: <sip:79185847476@multifon.ru> to: To: <tel:+79185847476> -- Thanks, Eugene Prokopiev

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FreeSWITCH as an SBC
by Code Ghar 17 Jul '10

17 Jul '10

FreeSWITCH, since it is a B2BUA, can be deployed as an SBC (Session Border Controller). I understand that SBC itself is a fairly complicated product and needs to be more security oriented rather than application oriented. For this reason many proprietary SBCs are sold and deployed all over the world. However, most of the things they claim accomplish can already be done in FS, such as SIP header or SDP manipulation/normalization, transcoding, hide topology, limit sessions and calls, etc. This thread is mainly to discuss some of the non-obvious things which other SBCs might have implemented and users would like to replicate using FS. A couple days ago I was working with a carrier which said they ignore OPTIONS and REGISTER packets. Is there somehow we can do the same in FS? This would be beneficial to carrier-type users who only do wholesale traffic from trusted IPs and do not want to mix user registration in it. I don't know why they would ignore OPTIONS. The first solution that comes to my mind is to block all IPs and allow only trusted IPs. This cuts down on the REGISTER attempts from untrusted IPs, but does not deal with the OPTIONS (or even REGISTER) packets from trusted IPs. A proprietary SBC my company is evaluating these days, the name of the provider or the product I would prefer not to mention, has the ability to forward REGISTER packets. A few days ago I saw in the FS mailing list archive (http://old.nabble.com/FreeSWITCH-as-pure-SIP-proxy-td20096367.html) that FS, being a B2BUA, does not simply forward REGISTER packets. Instead, it consumes them by either deeming them to be valid and registering user or rejecting them. From the standpoint of an SBC, it should simply be routing the packet rather than consuming it; basically act as a proxy for REGISTER packets only. Has anything changed? Is there anything you guys have done, something non-obvious, that could be used to implement FS as an SBC? Please share. Thanks.

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Sending SIP MESSAGE in javascript
by Storm Demo 17 Jul '10

17 Jul '10

Dear All, I am trying to send a SIP MESSAGE when someone call FS, however I can never get it work The below is what I am doing session.answer(); var msg = "Hello, welcome to the FreeSWITCH e = new Event("custom", "message"); e.addBody(msg); session.sendEvent(e); I found that my SIP phone never receive any SIP MESSAGE message Is that I have done something wrong? Thanks Jasper

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Segmentation fault (core dumped)
by Tim King 16 Jul '10

16 Jul '10

I am new to freeswitch and I am not certain where to begin to track down this issue. I have basic LCR setup and when testing the unit I keep getting segfaults. Can anyone point me in the right direction, or let me know which configuration files would be relevant to post to track the source of this. I am using SIPp to send calls to the Freeswitch at a rate of 20 calls per second. Thanks in advance Tim

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Retrieve Callee User Variables in Dialplan
by Roman_ 16 Jul '10

16 Jul '10

Dear all, I have been looking through the documentation and the forum but could not find what I was looking for. Basically, I was wondering if it is possible, that when somebody calls an internal number, in the dialplan I retrieve a variable associated with the user of the called number and then evaluate conditions according to the contents of that variable. Just as an example, let's assume I have a user in the directory with the following entry: <include> <user id="1000" mailbox="1000"> <params> .... </params> <variables> <variable name="some_name" value="some_value"/> .... </variables> </user> </include> Now, if somebody calls extension 1000, when evaluating the extension in the dialplan, I would like to be able to retrieve the variable "some_name" of the user with extension 1000 (i.e., the callee) and then take some action depending on the value "some_value" of that variable. Is there a mechanism to do that, or am I trying to do something the wrong way? Basically what I would like to do is be able to add a variable to a user and then, when that user is called, take different action depending on the value of that user. Thanks a lot. Regards, Roman -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Retrieve-Callee-User-Variable… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Video IVR on FS ?
by Anita Hall 16 Jul '10

16 Jul '10

Hi Is it possible to receive video calls using Free-SWITCH and then process them as an IVR ? One of our clients wants to set-up a video IVR system in the US and we are evaluation possible options. Also, what is the bandwidth of receiving a video call in US ? What protocols and codecs are supported and does it work on DID numbers ? Can I rent a hosted solution for this ? Thanks in anticipation of your valuable inputs. regards, Anita Hall, Simmortel.

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pre_answer problem
by Christian Löschenkohl 16 Jul '10

16 Jul '10

hello i try to pre_answer an inbound sip call. the inbound invite has the following codec order - g729, alaw. of course the pre_answer should be using alaw when answering but i only get 2010-07-16 16:27:59.708698 [ERR] sofia_glue.c:2271 No audio codec available the production system is running: 1.0.6 i get the same result with current trunk from git. why wouldn't the pre_answer'ing try harder and choose alaw? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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EventConsumer and multi-threading
by babak yakhchali 16 Jul '10

16 Jul '10

Hi I'm using mod_managed and I've used EventConsumer on a thread to listen for incomming events but whenever I wanna reload the .dll file of my script it complains about thread aborting. I used a dummy custom:dead event to end the thread but next time my dll loaded the custom:dead event is still in events and the tread ends. Is there any workaround to flush EventConsumer?? Thanks

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Sip chat messages
by Meftah Tayeb 16 Jul '10

16 Jul '10

hello i am using from console: chat sip|1003@external|883510000000094@XXX.XXX.XXX.XXX|hello but is not sending it i see unvalid profile X.X.X.X why? is not pocible to send chat msg to external uri? thanks -- Meftah Tayeb algérie télécom SPA phone: +21321761805 phone (INUM): +883510001289101 mobile : +213660347746 mobile (INUM: +883510001289110 http://www.algerietelecom.dz

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