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How to use freeswitch to play mp4 video file?
by Thomas Lee 05 Sep '13

05 Sep '13

Hi, How to use freeswitch to play mp4 video file? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/How-to-use-freeswitch-to-play… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Mod_callcenter - recording calls - template - to use final agent number on filename
by Ravi 05 Sep '13

05 Sep '13

Hello Everyone ! I am using mod_callcenter application to queue the incoming calls. I also want the calls to be recorded. The line to use to record incoming calls, as I have found in documentation is : <param name="record-template" value="$${base_dir}/recordings/callntrade/${strftime(%Y-%m-%d-%H-%M-%S)}_${caller_id_number}_${destination_number}.wav"/> When a caller calls, s/he is greeted by the IVR. On selecting one of the options the caller is passed to the queue. When I use this, the destination number is the number that indicates the queue. The number that I have assigned is 001 for the queue. If there are multiple agents, the call gets routed to one of them. The filename always has 001 as the destination number and not the number of the agent who handles the call. How can I include the name/number of the agent who finally handles the call ? Please let me know if you happen to know how this needs to be done. Thanks. Ravi

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mod_opus codec on Windows
by Jeff Leung 05 Sep '13

05 Sep '13

As I heard on last Wednesday's conference call, mod_opus isn't "available" on Windows, basically meaning that FreeSWITCH users can't fully implement WebRTC on Windows Platforms just yet. I have taken the steps to have the codec successfully build on this platform; However it's a pain in the neck to build as the opus library generates multiple libs which includes CELT and SILK codecs (Go figure). Shoot me a message off-list if you would like to use the binary I've built for Win64 platforms. Before anybody asks for a 32-bit binary, the short answer is going to be no. I won't be providing a 32-bit binary as I intend this binary to be used in a production environment.

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Re: [Freeswitch-users] Event socket and python
by Johan Brannlund 05 Sep '13

05 Sep '13

Sami Montour wrote: > Hi, > Try this modified version of your code. It should work. You can also > do "while(con.connected())" instead of "while(1)" Hi. Thank you for replying. I'm afraid your version of the Python code does exactly the same thing as mine: if I connect to a closed port, con.connected is still True and I get an unending stream of SERVER_DISCONNECTED events. By the way, this is with Freeswitch version 1.5.2b+git~20130526T122251Z~076a7706db.

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Event socket and python
by Johan Brannlund 04 Sep '13

04 Sep '13

Hi. I've been trying to adapt the Python event socket example at https://github.com/FreeSWITCH/FreeSWITCH/blob/master/libs/esl/python/events… to do what I want but haven't quite succeeded. In particular, it seems like con.connected is True even if con is not actually connected to anything, like if I try to connect to a port where nothing is listening. The program ------------------------- #!/usr/bin/python import ESL # nothing at all listening on port 8041 con = ESL.ESLconnection("localhost","8041","fakepassword") if con.connected: print "con.connected says we are connected!" con.events("plain", "all"); e = con.recvEvent() if e: print "Event: ",e.serialize() ------------------ prints con.connected says we are connected! Event: Event-Name: SERVER_DISCONNECTED Is this really the intended behaviour? If it is, how would I check that there *actually* is a connection without using con.connected? Look for SERVER_DISCONNECTED events? Any guidance would be much appreciated.

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EXCHANGE_ROUTING_ERROR with transfer and nested IVRs
by Carlo Dimaggio 04 Sep '13

04 Sep '13

Hi All, I have 2 IVR (ext 1200 and 1201)in which I don't use menu-sub action but menu-app-exec with transfer. (<entry action="menu-exec-app" digits="1" param="transfer 1201 XML domain"/> I would like to navigate the tree in this way: external phone (SIP Trunk) -> IVR 1200 -> (option 1) -> IVR 1201 -> (option 1) -> extension 201. But there is a problem when IVR 1202 try to transfer the call to extension 201: EXECUTE sofia/external/<EXT_Number>@<ITSP> bridge(user/201@domain) [...] 2013-09-04 14:02:54.450559 [NOTICE] switch_ivr_originate.c:2651 Cannot create outgoing channel of type [user] cause: [EXCHANGE_ROUTING_ERROR] 2013-09-04 14:02:54.450559 [DEBUG] switch_ivr_originate.c:3622 Originate Resulted in Error Cause: 25 [EXCHANGE_ROUTING_ERROR] 2013-09-04 14:02:54.450559 [INFO] mod_dptools.c:3096 Originate Failed. Cause: EXCHANGE_ROUTING_ERROR It is strange because when I navigate the tree froma local extension, all is fine(ext 200 -> IVR -> IVR -> ext 201). What could be the problem? Some variable should be set/unset? Best regards, Carlo Dimaggio

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RRC7007 sucks!
by Steve Underwood 04 Sep '13

04 Sep '13

Darn it. We added the DVI ADPCM codec to FS, just so we could say we are one of the very very few RTP platforms which are actually RFC3551 compliant, and now RFC7007 has removed the requirement. Steve

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Running FS on a heavy loaded VMWare host
by Grant Bagdasarian 04 Sep '13

04 Sep '13

Hello, I've been going through some threads about virtualization, some say it works for them, others say it doesn't. Multiple platforms are used like MS Hyper-V, VM Ware, etc. We currently run FS on a physical box, which works great. We have our own virtual environment, VMWare vSphere 5.0. Our 5.0 environment is running under heavy load and I was wondering would FS (configured properly for correct timing) still function properly and able to handle hundreds of concurrent calls on a heavy loaded VMWare host? The FS virtual will also process RTP in default mode. I was hoping if someone had any experience using this setup and could give me some insights on this matter. The FS virtuals will function only as SBC's, so no complex dial plans, no call recording, no conferencing. Just bridging the incoming call leg with an outgoing one. VMWare has a tech paper about running VoIP on their virtual machine, and they claim it works. (http://www.vmware.com/files/pdf/techpaper/voip-perf-vsphere5.pdf) In the paper they used Mitel vMCD, but I'm not sure if the software has been optimized to run on VMWare so the tests would come out better than running any other telephony software like FreeSwitch or Asterisk. Grant

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bleg app data question
by Stuart Mills 04 Sep '13

04 Sep '13

Hi, Is anyone able to tell me in which case app_log data will be written to the bleg xml-cdr of a call please. Does something trigger this off? Could it be something I am doing that causes this to happen? Thanks, Stuart

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Google voice call fails
by Terry Barnum 04 Sep '13

04 Sep '13

I've been reading the freeswitch book (great read!) and playing with my first freeswitch install on a Mac mini host with freeswitch installed via git into a Virtualbox guest CentOS 6.4. It's working with a couple X-Lite softphones and a Yealink T32G, internally between extensions, using a free DID to test incoming calls and a free but limited iptel account to test outbound. My next step was to try and play with real outbound calls so I followed these instructions for Google voice: <http://wiki.freeswitch.org/wiki/Google_Voice> but outbound calls fail and are hung up. I disabled the h264 codec in dingaling.conf.xml leaving just PCMU but no change. xmpp appears to be connecting and authenticating to google correctly. The log of an attempted call (edited for privacy) is at <http://pastebin.freeswitch.org/21338> Snippets of the log that looks suspicious to a freeswitch noob: 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1627 Accepted 0 of 0 rtp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1629 Accepted 0 of 0 rtcp candidates. 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1632 Accepted 0 of 0 video_rtp candidates 2013-08-26 20:45:33.568012 [DEBUG] mod_dingaling.c:1635 Accepted 0 of 0 video_rctp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 3 rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3597 candidate 74.125.141.127:19305 PASS ACL wan.auto 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3651 Acceptable rtp Candidate 74.125.141.127:19305 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 rtcp candidates 2013-08-26 20:45:34.649239 [DEBUG] mod_dingaling.c:3577 0 video_rtcp candidates 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4114 using Existing session for 2155444888 2013-08-26 20:45:35.208090 [DEBUG] mod_dingaling.c:4434 hungup dingaling/gtalk/+7609876543(a)voice.google.com Where should I be looking to try and fix? Freeswitch is behind the firewall with a private IP but STUN says it's working (or at least it looks that way to me). Thanks, -Terry

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