- FreeSWITCH-users - FreeSWITCH Mailing Lists

SDP Renegotiation
by Jon Lederman 14 Jan '15

14 Jan '15

Hi, I am trying to get my client to work with Freeswitch in the following context to no avail. If I initiate a video call, the video works fine. However, if I start an audio call and then try to add video later in the call, no video is received. I tried adding this line to vars.xml: (<param name="renegotiate-codec-on-reinvite" value="true”/> That, however, did not fix the problem. Any help would be greatly appreciated. The SDP negotiation is indicated below. Thanks in advance. I do see in fs_cli that FS has received a new SDP and is trying to renegotiate as follows: INITIAL SDP (AUDIO ONLY): v=0 o=FreeSWITCH 1421171138 1421171140 IN IP4 146.148.54.28 s=FreeSWITCH c=IN IP4 146.148.54.28 t=0 0 m=audio 25746 RTP/AVP 124 101 a=rtpmap:124 opus/48000/2 a=fmtp:124 useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv m=video 26852 RTP/AVP 98 a=rtpmap:98 VP8/90000 NEW SDP WITH ATTEMPT TO ADD VIDEO TO EXISTING SESSIOn: 2015-01-14 00:55:29.401667 [DEBUG] sofia.c:5828 Remote SDP: v=0 o=1000 3968 2153 IN IP4 10.1.10.10 s=Talk c=IN IP4 10.1.10.10 b=AS:380 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 53368 RTP/AVP 124 101 a=rtpmap:124 opus/48000/2 a=fmtp:124 useinbandfec=1; stereo=0; sprop-stereo=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 m=video 9078 RTP/AVP 103 a=rtpmap:103 VP8/90000

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Multitenant subscription restriction
by Ricardo Barroetavena 14 Jan '15

14 Jan '15

Hi, In a multitenant environment, is there a way to restrict one tenant UA from subscribing to another tenant events? For example, if I've got bob@domain1 and alice@domain2, is there a way to prevent bob from subscribing to let's say alice message-summary events? Thanks for the hints

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Database query from dialplan
by John Nash 14 Jan '15

14 Jan '15

I need to run a DB query (Postgresql) from dialplan and store results in variables. I came across this link https://wiki.freeswitch.org/wiki/Mod_odbc_query and looks like this is what I need. But I was wondering if I can use direct (without odbc) query to postgresql. Is there any such application available? P.S. I read some posts about postgresql native support but have not been able to figure out what do I need to do to run a postgresql query from dialplan and store results with out odbc

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Cisco SPA + BLF behavior
by Gabriel Kuri 14 Jan '15

14 Jan '15

I noticed that on a Cisco SPA500 series phone, when configured to do BLF+SD+CP for an extension, the LED for that line on the SPA does not turn red when the line goes off-hook on another phone. It turns RED when the call is actually in progress (ie after the number has been dialed and it's ringing). This behavior is different than when the same line is configured as a shared line appearance on the SPA, the LED turns red immedialtely when the line goes off hook on another phone. It appears FreeSWITCH is sending the same SIP NOTIFY event in both cases with line showing seized (appearance-index=1;appearance-state=seized), so I'm think this might be a bug in the phone's firmware, which is running the latest version, 7.5.6a. I did try a couple older version and had the same results. Am I missing something or is this the behavior of this phone? Should I bother to open a TAC case with Cisco, the behavior seems buggy to me?

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freeswitch reuses RTP port when bridging second call to user
by Dave Horton 14 Jan '15

14 Jan '15

I have a problem relating the handling of a second incoming call to same extension/user. The scenario is this: - call from PSTN comes to freeswitch; freeswitch bridges the call to the identified registered device; call is set up fine (two-way audio) - a second call from the PSTN to the same DID comes to freeswitch; freeswitch bridges the call to the same identified registered device; call has no audio In looking at the traces, the problem stems from the fact that for the second call freeswitch is sending an INVITE offering an SDP with the same source ip and port as the first call. This causes the subsequent INVITE on hold from the phone (which is accepted by the freeswitch) to cause the phone not to send any RTP; meanwhile, the RTP sent from the freeswitch to the phone for the second call is rejected with an ICMP port unreachable since the phone has successfully invited that port on hold. In other words, by re-using the same port for call 1 and call 2, when call 1 is invited on hold the result means no media for call 2. Is this known, or expected behavior? Is this a bug? Has this been fixed in a more recent release (I'm on a very old release) Dave

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Freeswitch only records empty files
by bee 14 Jan '15

14 Jan '15

Hey, i got an script which makes a call to an outbound number (so only 1 channel is open) and then i want to record the input. But the record function only create empty 44kb files. When i join the channel over a softphone the recorded files have input. Any ideas? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-only-records-empty… Sent from the freeswitch-users mailing list archive at Nabble.com.

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How can i hangup call if there is no audio or dead air on channel.
by Aqs Younas 14 Jan '15

14 Jan '15

Hi, list I am playing stream and sometimes streams went down and i hear on audio on my soft phone. i want to hangup call on whenever this happens. But don't know how can i do so. Any help would be much appreciated. Thanks

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Fork( ) and Exec ( ) functions
by Abdul Hakeem 13 Jan '15

13 Jan '15

Hello, I understand FS makes system calls for sending mails, voicemail and fax. Can anyone guide me on how to mitigate the load of fork ( ) and exec( ) on system calls & also, a list of functions which require FS to make system calls ? Thanks in advance. Cheers, Abdul Hakeem

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FreeSWITCH employment opportunity in Brevard County, FL
by McKnight, Jeffrey 13 Jan '15

13 Jan '15

Brevard County, (in Florida) with about 3000 employees, has been evaluating FreeSWITCH as a replacement for county wide legacy telephones for almost a year now. The system is almost completely feature functional in comparison to our Nortel system, however our FreeSWITCH expert has found more profitable employment elsewhere. We are testing the waters to see if there is anyone in the local area that would be interested in completing the perfection of a couple of features, assisting in the rollout of this system county-wide, and maintaining this system. This would be a full time employment position, good benefits, and great longevity. Please email me directly at Jeffrey.mcknight(a)brevardcounty.us if there is any interest. Regards, Jeff McKnight ----------------------------------------- Under Florida Law, email addresses are Public Records. If you do not want your e-mail address released in response to public record requests, do not send electronic mail to this entity. Instead, contact this office by phone or in writing.

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Error: cannot find profile
by chun 13 Jan '15

13 Jan '15

Hello, We have a freeswitch installed behind opensip which handles registration. We'd like to turn on and off MWI and cisco phones. When we use the following lua code, it will fail. We also tried using fs_path but it did not work. We appreciate any help. freeswitch.console_log("info", "Lua in da house!!!\n"); local event = freeswitch.Event("message_waiting"); event:addHeader("MWI-Messages-Waiting", "no"); event:addHeader("MWI-Message-Account", "sip:sofia/users/801.24118@10.0.151.110"); event:fire(); freeswitch@internal> luarun /etc/freeswitch/scripts/lua/mwi_event.lua +OK 2015-01-12 16:03:19.991140 [INFO] switch_cpp.cpp:1275 Lua in da house!!! 2015-01-12 16:03:20.011145 [ERR] sofia_presence.c:542 Cannot find profile 10.0.151.110 Thank you, Chun-Ju

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