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execute a lua script from another script
by Snabel Kabiya 10 Nov '14

10 Nov '14

Hi, Is there a way to execute a lua script from another script? I've this script s1.lua i want to run s2.lua from it. I've tried s1.lua: dofile(s2.lua); but this didn't work. Thanks, Snabel

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ESL: Which event is generated when an api command completed?
by Grant Bagdasarian 10 Nov '14

10 Nov '14

Hello, When using Outbound ESL in async full mode and executing a API command like uuid_transfer or uuid_bridge using bgapi, which channel event is generated to notify the command execution completed? For dialplan applications it is the CHANNEL_EXECUTE_COMPLETED event. That I know. All I got is this: Command Executed: uuid_transfer 563c9750-64cc-11e4-8cd8-4fc1451152d2 -both park_extension XML park_ext Bgapi returned an ESLevent: { "Event-Name": "SOCKET_DATA", "Content-Type": "command/reply", "Reply-Text": "+OK Job-UUID: ede6c869-0bfa-4e07-8490-04973b6ba971", "Job-UUID": "ede6c869-0bfa-4e07-8490-04973b6ba971" } However, a channel event named SOCKET_DATA did not show up when listening for channel events using the RecvEvent method. What I'm trying to do is use bgapi or sendMSG to execute an application/command, and use the RecvEvent method to wait for a channel event that indicates the application/command completed execution. So if I were to use the playback application, I'd send it using the SendMSG method, and use RecvEvent to wait for the CHANNEL_EXECUTE_COMPLETED event containing the UUID set for this playback. This works fine for dialplan applications but I don't know which event to listen for when using api commands. I hope someone could point me into the right direction. Regards, Grant

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DTLS-SRTP decryption
by Luis F Urrea 10 Nov '14

10 Nov '14

Hi all, So I have been doing some research on the feasibility of decrypting SRTP traffic whose key has been negotiated through DTLS, such as WebRTC RTP traffic. It seems to me that even when having the private keys from the DTLS negotiation, the SRTP master key is obtained through a TLS-exporter mechanism as defined in RFC5705 <https://tools.ietf.org/html/rfc5705>, and therefore I haven't found any way to obtain this from a packet capture. It only seems reasonable that an endpoint in the negotiation would provide this. Is there anything that could be done in the FS side of the DTLS negotiation to obtain the SRTP master key? Thanks in advance for your input.

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hash_remote <list>|<kill> [name]|<rescan>
by Andre Demattia 10 Nov '14

10 Nov '14

Hi, What would the syntax look like to connect more than one FreeSwitch Limit hash_remote <list>|<kill> [name]|<rescan> Thanks Andre

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Json_cdr vs cdr_mongo
by Volkan Oransoy 09 Nov '14

09 Nov '14

Hi all, I am working on a CDR reporting tool and I am testing both mongo cdr and json cdr. With mongo cdr, there is no problem and I am getting all variables as expected. With json cdr, I am sending records to couchdb. I works well with a minor diffrence. Mongo shows all callflow in an array but couchdb (with json cdr), callflow variable shows only first flow element. Am I missing a point or is this a normal behaviour? Thanks, /Volkan

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Extension and carrier same ip problem
by Martin Snelgoedkoop 09 Nov '14

09 Nov '14

> Hi, we use Kazoo which loads acl and dial plan out a DB. > We run in to smth that seems freeswitch default behavior so that's why I write here. > > We have a carrier that needs to register a extension to our platform as well. > They use the same ip. And that creates a problem as when I place a call with the device using the extension it gets identified as carrier and the route is then not correct. > > The same happens the other way arround, if a call comes in via the carrier the ip is recognized as an extension ip and then it won't use the carrier route. > > I was wondering if I could use some flags or a way of changing the ip used or doing some magic to make it work. I would prefer not to use a proxy. > > The setup is kamailio>freeswitch > >

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unable to do "api uuid_getvar" in outbound socket created by originate command
by raj gopal 08 Nov '14

08 Nov '14

I am trying to "api" commands on outbound socket created by originate .... &(socket ..async full) and I am getting -ERR command not found. So I tried same thing on fs_cli and netcat. from fs_cli executed following command: originate sofia/internal/1000%192.168.1.94 &socket(192.168.1.94 9050 async full) I did a netcat: on 192.168.1.94 9050. when the extension 1000 answered, I did connect\n\n , myevents\n\n and sendmsg command to answer (i.e. answers\n\n). now if I execute api uuid_setvar uuid x 30 or api uuid_getvar uuid x; I am getting -ERR command not found. if I do same stuff on outbound socket created when extension 1000 dials in to extension 1021. and I setup dialplan to created outbound socket application=socket data="192.168.1.94L9050 async full" I could do api commands. What is the difference in both outbound sockets?? except one is created using originate other is created when dialed in. My intension is to dial a phone number, do play_and_get_digits and do get the digits pressed by user using uuid_getvar in outbound socket but the originate command it self is issued in inbound socket. appreciate any insights. Regards, Raj

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FreeSWITCH founder gets movie credit
by Ken Rice 08 Nov '14

08 Nov '14

New Post on freeswitch.org from anthm check it out at http://ift.tt/1ACmdcc FreeSWITCH founder gets movie credit I’m an extra in a movie but I’m not sure you ever see me. My son has a real part though as a kid named Sam and ends up in a sticky situation to say the least. Check out Available on iTunes and DVD! -Anthony Minessale

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Cannot Establish Audio Call First, Then Enable Video
by Jerry Richards 08 Nov '14

08 Nov '14

Using two Bria softphones, if I establish an audio call first, and then turn on video, the softphone sends a re-INVITE including the H264 codec, but Freeswitch does not forward the INVITE to the other endpoint. If I originate the call as a video call, then the video call is established successfully. I tried this with the latest Freeswitch as of yesterday. The only change I made to the default configuration was to add the H264 codec to <global_codec_prefs> and <outbound_codec_prefs> in vars.xml. Is there a configuration tag that would allow this? Or is it required that the video codec must be offered in the first INVITE for video to work (i.e. cannot be added later)? Regards, Jerry

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Newbie question: Set up to answer new phone number (please be gentle)
by T Fred Farmington 07 Nov '14

07 Nov '14

Since I am a total newbie to FreeSwitch please be patient and 'gentle' with me.... I have 'inherited' 2 FreeSwitch installations. One is working (version 1.0.7) and one (version 1.5.14) is a new test environment that is not working. I cannot get the new test environment to answer the inbound call and in the FreeSwitch log I find: Rejected by acl "domains" I have tried to search the FreeSwitch documentation for an answer and got lost in it. And I have tried to search the web for an answer, but did not find anything clearly spelled out. And, lastly, I tried to investigate the working installation, but could not find my answer. I have a dedicated SIP phone line which can call into the new test environment. And via FS_CLI I can see the call being received. However the connection is never established and in the log I find: Rejected by acl "domains" In both FreeSwitch environments I have a directory named: \Debug into which the previous individual installed all of the configuration parameters. Questions: 1. Where/How is FreeSwitch configured to recognize this directory and look for its parameters there? **** Maybe the non-working environment does not have that setting established and therefore is ignoring settings made there. 2. I have configured the new inbound IP address into the file: acl.conf.xml and have 'mirrored' the working environment in that regard. **** But the 'Rejected' message in the log suggests that somehow that is not being recognized. 3. Are there other files which I have missed in trying to set this up? Thanks

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