- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH not sending Notify to presence subscribers of Callee
by sagar malam 15 Oct '17

15 Oct '17

Hello, I am working with FS 1.6.19. I am facing a situation when FS is not sending Notify packets to SIP endpoints which have subscribed for callee presence. To explain scenario, I have 3 SIP endpoints : EP 1 :1000 EP2 : 1001 EP3 : 1002 EP3 has subscribed for 1000 and 1002 presence information. When i call from EP1 to EP2,FS sends Notify packet for EP1 to EP3 but not for EP2. Should FS send Notify for EP2 as well ? or this is the functionality of FS itself ?

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http://files.freeswitch.org/ down ?
by Yuriy Nasida 13 Oct '17

13 Oct '17

Hi, anybody able to open?

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record_session without 3rd party from eavesdrop
by Marcin S 13 Oct '17

13 Oct '17

Hello, I have an C ESL application, that makes calls to two parties (bgapi originate ...), then connects them together (uuid_bridge) and record (record_session). This works fine, but sometimes there is an 3rd leg attached to this bridge via eavesdrop or threeway. The problem is, that when eavesdrop is switched to WHISPER or THREEWAY mode, this 3rd party is also recorded. How can I make recordings without this 3rd party? I have also tried making two separate record_session calls for each of two legs (with RECORD_READ_ONLY), but it doesn't help.

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doing an att_xfer from within a uuid_standby session (mod_callcenter)
by Khalil Khamlichi 13 Oct '17

13 Oct '17

Hi, I am trying to make an att_xfer from api. con.execute('att_xfer','sofia/gateway/provider/%s' % dstNumber, uuid) all is working, except when agent presses feature key 0 to bridge both parties, his uuid_standby call hangs-up too and so he is out of the queue and needs to log in again. Is there any solution to this problem? Thanks in advance. Khalil

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Choppy and fragmented return-audio problem using Unicast
by Dayton Turner 13 Oct '17

13 Oct '17

Hello list, I've got a question about using the unicast call command via ESL. I realize there are other ways to do this, such as writing a native C module, but with a lack of developers with the necessary C skills, we're trying to take a "simpler to implement" approach at first just to get things rolling. The objective is to send and receive audio to a 3rd party application. In this case, we're using a UDP transport via unicast, running a small application written in golang on the FS server itself. As a proof of concept, we're simply capturing received RTP on the UDP socket, and piping it back to the UDP port FS is listening on - basically an echo test. Audio does come in, we are able to send audio back, but the return audio as we hear it sounds incredibly choppy, and like some of the audio frames are being clipped or dropped entirely. We also tested merely saving the received audio to a file and wrote it to disk. Loading it up in audacity and importing raw audio, it plays back properly when we import it as ULAW 8khz mono. Additionally, we also tried skipping the echo test and instead simply opening the saved file and bitstreaming it back to FS on the listening unicast UDP port. Despite playing back properly in audacity, its all choppy and cut up when played back via the UDP stream. Nothing fancy going on, literally just opening the file and bitstreaming it to the UDP socket. We also tried a couple example 8khz ulaw files, just streaming those back and they behave similarly. In fact, they sound sped up (in playback "speed", but not pitch, which is interesting). Am I missing something about what format the UDP unicast listening port in FS expects to receive, differently than what it sent out in the first place? As we receive 8khz ulaw, we're sending that back in return. flags:native is specified on the ESL call-command. We also tried transcoding to L16 before sending it back, no luck there either (completely unintelligible stream of fuzz) I thought about trying another method of allowing a 3rd party application to access the audio stream (both receive and send back) like mod_tts_commandline but it wasnt clear wether this route permits for audio in both directions to be involved or if it was merely return audio. Additionally, via ESL if we execute the playback application and just ask FS to play one of its own soundfiles, it sounds perfectly fine, so Im confident that the setup up until the unicast test is working correctly. Testing is on FS 1.6.15 on Centos 7. Any thoughts? Thanks in advance :) Dayton <http://www.getpostbox.com><http://www.getpostbox.com>

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Attended transfer on Softphones not updating number
by Shaun Stokes 12 Oct '17

12 Oct '17

Hi, We're experiencing problems with the CLI not updating when performing attended transfer to softphones (Jitsi, Bria, Zoiper). This problem does not occur when performing attended transfer to a physical phone (Polycom, Cisco, Yealink), it also does not occur on other VoIP platforms. The SIP messaging is different when performing an attended transfer to different devices, for softphones such as Jitsi, Bria and Zoiper there is no update message sent once the transfer completes whereas for physical phones there is. We have narrowed this down to: /src/mod/endpoints/mod_sofia/mod_sofia.c There are a set of if\elseif statements between line 1943 and 2025 which provide different update messages depending on the receiving device. Can someone please confirm if this is a known problem? Do we need to add support for the softphones? If so, I will raise a JIRA and do further testing, perhaps we can come up with a patch. Thanks, Shaun ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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One to many w/o mod_conference
by Tom Hartnett 11 Oct '17

11 Oct '17

Greetings all, I have an application where I could use some expert advice. I'd like to set up an audio broadcast bridge, where the audio from one call would be distributed to multiple listeners. Listeners have no send audio. Think a corporate conference call with only one host. All listener calls are incoming. I could of course set this up with mod_conference and just mute all the listeners. But this requires a bit of overhead to convert everything to PCM then back to the listener's codecs individually. I'm working on a resource-strained embedded system, so I'd like to find a way to simply bridge the speaker audio to multiple endpoints simultaneously. I have very tight control over the codecs used, and can make sure they are all the same. Sort of like live MOH (I'd be using mod_portaudio for the sending endpoint in fact). Delay must be very low, so converting to a shoutcast stream and using the MOH function isn't an option either. Can anyone offer any advice? Maybe FS is just not suited to this?

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Re: [Freeswitch-users] RTP Loss with Freeswitch in Default Mode
by vikas sharma 11 Oct '17

11 Oct '17

Hi Can i get some suggestions to improve the video call quality and reduce packet loss with Freeswitch server. Thanks On Wed, Sep 27, 2017 at 11:00 AM, vikas sharma <vikas452(a)gmail.com> wrote: > Anything that can help me in this regard?? > > On Mon, Sep 25, 2017 at 4:03 PM, vikas sharma <vikas452(a)gmail.com> wrote: > >> Hi >> >> I am making a linphone to linphone call using Freeswitch server in >> Defualt mode but i am facing huge RTP loss and the video call is getting >> stuck in between.I have also found that there is a difference between the >> RTP packets received on A Leg and sent from B-leg and vice versa. >> >> Can somebody help me to understand what exactly happens with the RTP >> packets on server and why there is a difference in the RTP count?? >> >> How can we improve the video call quality in moderate network conditions. >> >> Thanks >> > >

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Sending keepalives on Verto websocket
by Chad Phillips 11 Oct '17

11 Oct '17

I've switched to using Nginx to proxy Verto websockets, and have run into a small snag: by default, if Nginx doesn't read any data from a proxy backend within 60 seconds, it closes the connection, even for websockets. It appears the recommended solution is to have the server send some kind of regular keepalive. I poked around in mod_verto.c and found a 'request.keepalive' variable, but I'm unclear how to set that in the request, and/or if it even accomplishes what I'm wanting. I've solved the issue for now by periodically sending a JSON RPC 'echo' request along the websocket every 50 seconds, and the reply from the server is enough to keep the connection open. This is fine, but I am curious if there's a way to do it just from the server side, and if not, if it's worth it to add a setting to enable that functionality?

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freetdm errors..
by Babak Yakhchali 11 Oct '17

11 Oct '17

Hi I'm using sangoma a200 with freetdm on freeswitch 1.6 stable branch. Sometimes I see errors like this in console: 2017-10-11 10:58:32.669207 [ERR] ftmod_sangoma_isdn_stack_out.c:574 [s2c31][2:16] Received frame of 323 bytes, exceeding max size of 300 bytes Is this a serious problem?

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