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PROBLEM: Making Concurrent Calls
by Shan Randhawa 01 Aug '13

01 Aug '13

The problem, i m confronting right now, is that When I make 2 calls simultaneously from 4 phones,it acts like a Conference Call means everyone is listening to everyone, For Example if cell A calls cell B,AND cell C calls cell D,then cell C can hear the conversation between A and B. and similarly vice verse is happening. Here is the extension that i m using to make calls, ------------------------------------------- <extension name="local_call"> <condition field="${python(VBTS_DB_Get name|callerid|${destination_number})}" expression="IMSI\d{15}"/> <condition field="${python(VBTS_DB_Get callerid|name|${username})}" expression="\d{7,10}"> <action application="set" data="target=${python(VBTS_DB_Get name|callerid|${destination_number})}"/> <action application="set" data="port=${python(VBTS_DB_Get port|callerid|${destination_number})}"/> <action application="set" data="host=${python(VBTS_DB_Get ipaddr|callerid|${destination_number})}"/> <action application="set" data="effective_caller_id_number=${python(VBTS_DB_Get callerid|name|${username})}"/> <action application="set" data="hangup_after_bridge=true"/><action application="set" data="continue_on_fail=true"/> <action application="bridge" data="sofia/internal/${target}@ ${host}:${port}"/> -------------------------------------------- Does any one has any feedback on this why this is happening. Thanks in Advance

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switch_channel_wait_for_state_timeout
by Stephen Wilde 01 Aug '13

01 Aug '13

the call to: switch_channel_wait_for_state_timeout(channel, CS_REPORTING, 5000); in "switch_ivr_originate.c" is correct? What I mean is that: CS_REPORTING >= CS_HANGUP then the "switch_channel_wait_for_state_timeout" doesn't wait for CS_REPORTING state: I'm wrong? Stephen

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WebSocket for proxied outgoing calls
by mayamatakeshi 01 Aug '13

01 Aug '13

Hello, I have kamailio SIP proxy with WebSocket support in front of FS using plain UDP transport (not using ws-binding). It works fine for incoming calls: http://sipml5.org --> SIP over WebSocket --> kamailio --> SIP over UDP --> FS plays some prompt. Then, the WebSocket app registers with kamailio and when a call arrives to it at FS, FS sends the call to kamailio and the call is sent to the browser. However, when I try to answer the call, the web app refuses the call with this: SIP/2.0 603 Failed to get local SDP. From: "displayname.user3"<sip:user3@test1.com>;tag=6197r4g0Kar9H. To: <sip:user2@127.0.0.1>;tag=okDK7cd5n34HgaEM9dgs. Call-ID: 2d1ab230-7516-1231-969f-5254000fd208. CSeq: 47333138 INVITE. Content-Length: 0. I suppose this is happening because an SDP for a WebRTC call cannot be a plain one that we send on ordinary calls. Is this correct? So is it possible to instruct FS (some parameter when calling originate or bridge) to prepare an SDP for WebRTC? If not, I was thinking if asking FS to not send SDP in the INVITE would work (late negotiation) as it would receive the SDP in the "200 OK" and send its SDP in the ACK. But i could not find a way to force FS to do this either. regards, Takeshi

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List registered users
by Jose Suero 01 Aug '13

01 Aug '13

I've been searching on how to get a list of registered users, all I found was: "sofia status profile external" which says the amount of Registrations, but doesn't list the actual users. How do I do that?? thanks in advance Jose Suero

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Just curious, is there any standard for FS API command output?
by Rafal Gwizdala 01 Aug '13

01 Aug '13

Guys, I'm communicating with FS through an inbound event socket and almost every command has some different standard for output data structure. Some responses are CSV, some (like 'sofia status profile internal reg') use a 'vertical name:value format , i've seen also different formats I don't remember now. What's worse, the event returned after executing a command (SOCKET_DATA api/response) doesn't indicate if the command succeeded or not. Some commands are nice enough to return +OK for success but not all of them. Am I missing something here? Are there more programmer-friendly methods of getting command results from FS? Thanks RG

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What ports are really necessary?
by Jim Lynch 31 Jul '13

31 Jul '13

I'm running a simple switch. No phones are connected to the system from outside of my local network. The only wan connections I have are a couple of voip providers. None of my phones on the lan call outside of the lan. All they talk to is the switch. I see a bunch of ports listed at http://wiki.freeswitch.org/wiki/Firewall but don't know if I need them all. I suspect I don't. I want to close up the firewall as much as I can. Thanks, Jim.

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Consecutive bridge actions vs multiple targets in a single bridge action.
by Oleg Sharoyko 31 Jul '13

31 Jul '13

Hi, I was wondering if there exists a simple way of implementing behavior equivalent to <action application="bridge" data="sofia/gateway/gw1/$1|sofia/gateway/gw2/$1"/> with two consecutive bridge actions? I've tried this: <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="sofia/gateway/gw1/$1"/> <action application="bridge" data="sofia/gateway/gw2/$1"/> and it results in almost the same behavior except that if first call is rejected by destination (say I was calling a mobile phone and on that phone reject button was pushed) then second action will still be executed. I think with a single bridge action call is considered successful if early media state is reached while with two consecutive actions hangup_after_bridge=true only acts on a fully established call. I couldn't find any options to change this. Will appreciate your help. Thank you! Best regards, -- Oleg

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lcr cid maniluaption
by Miha 31 Jul '13

31 Jul '13

HI, I need a little help with regex for cid manipulation in lcr. On wiki I can see that I can do it like this: /search/replace/ I am getting sometime number like: 385*60001264567 and sometimes without * (385*60001264567) (from different carrier). How can I get rid of * with regex in LCR table? I can see that row CID is for this manipulation. tnx miha

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Weekly Conference Call Today
by Michael Collins 31 Jul '13

31 Jul '13

Hi folks, The ClueCon weekly conference call is coming up in about an hour. Here is our agenda: https://wiki.freeswitch.org/wiki/FS_weekly_2013_07_31 Today Chris Rienzo is going to highlight some of the interesting things that you can do with mod_spandsp - it does more than just faxing! Talk to you soon! -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org

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Record write part of all legs in a bridge call
by Doron Kruh 31 Jul '13

31 Jul '13

Hello, I am trying to perform a record of all legs in a bridge call(and maybe in the future conference of n users) . It is important to keep the audio quality of the audio as good as possible (8khz for non HD and 16 khz for HD). To be more precise , I am interesting in recording only the WRITE channel of each leg (the READ channel is not important for my purposes). Currently, the only solution I was able to find was using record_session with STEREO option on the incoming UUID . Although I got the bleg audio available on the read channel , I also got all the audio I played to the bridge call on the same channel. The best solution will be to perform a record for each call leg which will record only the WRITE channel (based on the terminology defined in the record_session API) - is there currently any way to achieve that? Best regards, Doron. -- Doron Kruh, Senior Developer Lexifone , Hapalyam 7 Haifa. Mobile: 972-52-3823361 Office: 972-4-9126792

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