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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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Multiple (3) NICs
by Тарас Гірник 16 Jul '10

16 Jul '10

Hi, guys. Sorry for posting suppose such a frequently asked question, but anyway it's still problem for me to handle Freeswitch on 3 different subnetworks. In fact i was able to run, but with one discfuntion - on one of the instances there was no sound in one direction. I know you say this problem is related to NAT and so on, but let me ask anyway. It should mentioned that first i was run Asterisk and it was fine with that setup. Another important thing is Freeswitch runs under guest OS Debian/Lenny in OpenVZ environment (kernel 2.6.18 with ovz patch) Network: venet0:0 192.168.1.128/24 venet0:1 public_ip + default route here eth2: 172.20.252.6/30 + route 172.20.252.0/24 via 172.20.252.5 (complete interface derived to guest OS, nothing to deal with HN) My peers is 192.168.1.1 (asterisk runs here, under my administration) 192.168.1.204 (gsm gateway, under my administration) x.x.x.x (remote peer under my administration) 172.20.252.2 (voip providers proxy, no authentication here, not under my administration) I successfully installed profiles belong to first 3 ones and everything was ok. I substituted external-sip external-rtp and others with my public ip. The internal ones - with my internal. The problem was with the last one. I have tried to force Freeswitch run on that one but have no luck. Finally i copied profile "internal.xml" to "internal-2.xml" (actually i tried the same with external.xml) in "sip_profiles" dir and change it to parse files from "sip_profiles/internal-2" dir. I have no idea is this right way, but anyway it works and Freeswitch start to listen on 172.20.252.6. I also used inbound-acl and proxy-acl to handle with no authorization. Freeswitch was able to handle incoming calls on 172.20.252.6 and even it was sound in both directions. But if you try to add a dialplan which will use 172.20.252.2 as an outbound proxy you will see that call is handled correctly but no voice there. I examined outgoing packets send by my phone to Freeswitch and from Freeswitch to 172.20.252.2 and found no rtp was transmitted over eth2 (from freeswitch to proxy) only from my phone to freeswitch. There is no FW on guest OS, i tried to stop in on HN but nothing changed. Packets come to venet0 ok, but then for some reasons are not transmitted to eth2. I'm sure that is something with nat-acl as far as 172.16/16 is also private space addresses, but i'm not sure. Please, help me with that and if you know the right way to handle such 3 subnets config correctly, tell me. Freeswitch was build from the git. Corresponding conf from Asterisk, works [172.20.252.2] type=friend context=*** canreinvite=no host=172.20.252.2 disallow=all allow=alaw videosupport=no outboundproxy=172.20.252.2 fromuser=*** from freeswitch (no audio during outgoing calls) http://www.privatepaste.com/bebc7e8988 sip_profiles/internal-2.xml and http://www.privatepaste.com/4df33fc7a5 sip_profiles/internal-2/int. xml

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