September 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

freeswitch, mod_callcenter and heartbeat
by Evgeniy Kozhuhovskiy 05 Sep '13

05 Sep '13

We have almost the same story and problem as here: http://lists.freeswitch.org/pipermail/freeswitch-users/2013-April/094651.ht… We have two servers (fs1 and fs2), and one shared IP between them (heartbeat). Also, we've configured fs (core-db-dsn and odbc-dsn's in profiles) into MySQL, and moved mod_callcenter's DB into MySQL. And, ofcourse, we've enabled track-calls for each profile. FS listens only on shared IP with net.ipv4.ip_nonlocal_bind in sysctl.conf This configuration allows us to transfer calls between nodes almost without interrupt. The only one problem is mod_callcenter. On "passive" node it is trying to process calls from shared DB. Is it a possibility to fix this behavior? -- With best regards, Evgeniy Kozhuhovskiy

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play_and_get_digits - with no invalid entry
by Ali Pey 05 Sep '13

05 Sep '13

<#>Hello, I need to play a prompt and get any digits if entered, but if there is no digit entered, simply continue. Basically I don't have any minimum entry required and there is no invalid entry. I use playAndGetDigits in my perl code but it doesn't accept 0 for min or an empty string for invalid_file. How can I achieve this behaviour? Regards, Ali

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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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