March 2014 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Handling SIP 180 ringback to freetdm?
by Pete Ashdown 21 Mar '14

21 Mar '14

I'm using Freeswitch as a PSTN gateway behind a Kamailio SIP proxy. I get no ringback audio if call a phone in this direction: (PSTN freetdm) -> Freeswitch -> Kamailio -> Phone (registered to Kamailio) Debugging it, I see that SIP 180 ringing is sent from the phone to Kamailio, and then relayed properly to Freeswitch, which does recognize the ring, but because there is no RTP audio associated with it, has nothing to send into freetdm. So my question is how do I turn SIP 180 into ringback audio generated on the PSTN gateway itself?

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thoughts about call fraud and sip security
by Sandro 21 Mar '14

21 Mar '14

Hello all, out of curiosity, since i work in the voip field, i am playing with a freeswitch + fusionpbx setup from a few weeks. I put it on a linux machine with public ip address, no firewall. The scenario is quite simple: it has a gateway towards a VoIP provider and one extension that is a softphone behind NAT (in my LAN). All works smoothly. Of course I received a lot of scans in these days, and i configured my fail2ban to react properly. But yesterday I received a call attempt from a malicious users on the public context (where resides the gateway to my voip provider) and this has been processed. Of course the call hasn't got through since the called number was not my own number, and it has been dropped in the public context without being transferred to the default context. But, if the attacker would guess my public number, i think i could receive a unauthorized call inbound to my softphone. And since this kind of calls are processed, i think i am suitable to be DDoS-ed. There comes a more theoretical question about the SIP protocol (I have posted it on the sip-implementors ML without any reply): *Let's have a very common scenario, where a sip "client" A (my freeswitch) interacts with a sip server B* *(my voip provider).* *To keep things simple, the interactions are registrations (REGISTER) and calls (INVITE and ACK).* *The administrator configured A with credentials that have been registered on B, so REGISTERs and INVITEs* *incoming to B are authenticated.* *Now in this network a malicious user appears, let's call it C.* *Of course it will be not able to send any malicious request to B, since B will ask for credentials (that C does* *not know) and subsequentially drop the unauthorized requests.* *What about A? What if C sends an unsolicited call to A?* *At this point A has a "registration" up with B, so they are exchanging some informations.* *Is A able to recognize that the call C is sending it is "not related" to its "registration session" with B? (of* *course without dealing with network addresses)* *For example, A and B are exchanging some random tags.* *Are they brought (by protocol, and not as an proprietary extension) in the subsequent dialogs initiated by* *B towards A, so that A can recognize them?* *If that is the case, can you point me to the exact point of the RFC that states it?* Thank you to anyone that reached here to read :) Kind regards, Sandro.

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sangoma codec
by jay binks 21 Mar '14

21 Mar '14

Can someone confirm the behaviour of mod_sangoma_codec, in relation to G729 passthru. I just want to be 100% sure that a call with 2 x G729 legs does not leave the codec module. ( and also that this does not count towards licence channel usage ) Also further to this, if the SNGTC soap server is unavailable for whatever reason will this affect the setup of G729 passthru calls or will it only affect transcoded calls ? -- Sincerely Jay Binks

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Convert bridge into a conference on the fly
by Oleg Stolyar 21 Mar '14

21 Mar '14

Hi guys, what's the best way to do this if I need to include a third person in a conversation? Right now all my conversations are conferences but I'd like to start them out as bridges to save resources. Looks like a conference takes 50%-80% more CPU than a bridge even when there are only two participants and both use the same codec (PCMU)

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ZRTP SAS to non ZRTP call leg UA?
by Bill Ross 20 Mar '14

20 Mar '14

Hi; When a call is between a ZRTP UA and non ZRTP UA, with FS MITM, I want to give the user the SAS that the ZRTP call leg is using. I note there is a script: scriptd/lua/zrtp_sas_proxy.lua that is supposed to hook into the call flow by: <action application="export" data="nolocal:api_on_answer=luarun zrtp_sas_proxy.lua ${uuid}"/> This appears to display the SAS, and, I will add TTS for POTS phones. Q1: Where do I make the script install call above? Q2: Is there a better way, code? Thanks; Bill Ross

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To everyone testing git master...
by Brian West 20 Mar '14

20 Mar '14

With all the changes going in to the build system over the next week or two we’ll need everyone to assist in building and reporting issues back to us. Thanks, -- Brian West brian(a)freeswitch.org FreeSWITCH Solutions, LLC PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com T: +1.918.420.9001 | F: +1.918.420.9002 | M: +1.918.424.WEST iNUM: +883 5100 1420 9001 ISN: 410*543 Skype:briankwest PGP Key: http://www.bkw.org/key.txt (AB93356707C76CED)

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P-Asserted
by George Assaad 20 Mar '14

20 Mar '14

Hi, When freeswitch receives 200 OK in the packet we see the following: From: "company" <sip:+compnay phone #@IP;user=xxxx> P-Asserted-Identity:"vendor"<sip:+vendor phone #@IP;user=phone> Privecy: id When freeswitch sends the same 200 OK packet to the endpoint: P-Asserted-Identity: "vendor" <sip:+vendor phone #@IP> Ideally we need freeswitch to either stripped the identity or leave the *privacy id* header. How to do that? Regards, George

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Example of playing background music
by Alex Lake 20 Mar '14

20 Mar '14

Is it possible in Lua to do something like.... playback_uuid = GetTheUuidFrom(session:execute("bgapi uuid_broadcast <a-leg uuid> nicemusic.wav")) Then do some stuff on the b-leg Then session:execute("uuid_kill "..playback_uuid) When I've tried it, I get.... 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 exec_async uuid_broadcast 4b0faa1a-6898-438a-8099-8c09eff7d1b0 local_stream://moh... (dmpx_r27s1_bleg.lua uuid:ddde917b-d9df-44b1-9b33-2b46d639f100 [?]) 2014-03-20 12:27:07.474905 [DEBUG] switch_core_media_bug.c:532 Attaching BUG to sofia/internal/0001302(a)004-0095.sb12.dmclub.org 2014-03-20 12:27:07.474905 [DEBUG] switch_core_session.c:1151 Send signal sofia/internal/0001302(a)004-0095.sb12.dmclub.org [BREAK] 2014-03-20 12:27:07.474905 [INFO] switch_cpp.cpp:1288 ...exec_async returning +OK Job-UUID: 88727341-5d44-4631-a69a-368df85ebdf0 ... ... 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 dmExec uuid_kill 88727341-5d44-4631-a69a-368df85ebdf0 2014-03-20 12:27:11.244906 [INFO] switch_cpp.cpp:1288 ...dmExec returning -ERR No such channel! I can only presume I've got the wrong end of the stick on these asynchronous jobs! (BTW, dmExec is a wrapper around api:executeString)

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Using FreeSWITCH as a STUN server
by Oleg Stolyar 20 Mar '14

20 Mar '14

Hi guys, WebRTC requires STUN servers to get the external IP address for ICE candidates. I would prefer not to rely on external STUN servers (google, etc) in my production environment. Ideally I'd prefer to use the FreeSWITCH server I need to connect to as my STUN server. Is this possible? Can FS act as a STUN server?

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Preventing auto moh
by Alex Lake 20 Mar '14

20 Mar '14

I note that if I've got a bridged call and there's a bound keystroke, when the binding is activated, the other party is automatically given hold music. I wonder if there's a way to disable this feature?

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