October 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

outbound route question
by alex 17 Oct '17

17 Oct '17

Hello: I want to make a loop test from port 1 to port 2. Now, the pri show OK and UP. But the dialplan does not work. the call flow is : SIP A->Port 1->Port 2-> SIP B. it looks SIP A directly call to SIP B without going E1 ports. The attached file are inbound route file 、outbound route file and log file. Is there a problem with outbound route? what should i do? I hope you can help me,thank you.

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newbie Question in two parts
by Shawn Wheeler 17 Oct '17

17 Oct '17

I have three separate FS systems I running. I would like to be able to make a Mad Boss call from server A to Servers B and C. I found this page that tells how to connect two boxes. https://wiki.freeswitch.org/wiki/Connect_Two_FreeSWITCH_Boxes Question 1 - on this page the two server have difference extension. Is there a way to do this if they similar e.g. 1001, 1002 etc.. Question 2 - Can I make a Mad Boss call from one to multiple servers? If yes, can you point me in a direction to learn how to accomplish this task? Thank you in advance. Shawn

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Connection problem between FreeSwitch Server and Asterisk Server
by 王聡 16 Oct '17

16 Oct '17

I had built a set of server which includes a FS server and an Asterisk Server, and I hope to use FS on application call, Asterisk on public call. Both 2 servers had installed 2 Ethernet controllers(one for public network, and one for LAN). During test I found the App-App call works, but App-public not. Asterisk and FreeSwitch can’t communicate well via LAN. I thought it might a firewall problem, but it still not work since I disabled the iptables on both server. FS is listening a public ip with 5080 and a local ip with 5060, while Asterisk sends the message by local ip with 5080. In this situation, FS can’t receive messages from Asterisk. After I set port to 5060 on Asterisk, it can’t receive messages from FS. Anyone has idea about settings? Do I need to make another sip profile to listen local:5080?

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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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