Server Fault Stack Exchange
Server Fault Stack Exchange

Questions tagged [freeswitch]

FreeSWITCH is an open-source telephony platform designed to facilitate the creation of voice and chat driven products.

FreeSWITCH is an open-source telephony platform designed to facilitate the creation of voice and chat driven products.

It was created by a core Asterisk developer frustrated by the quality of the Asterisk code base, but is not a fork; it was written from scratch.

FreeSWITCH is used as the core of the Barracuda CudaTel PBX appliance.

51 questions
1
vote
0 answers

Slow Down Mod_Flite TTS in Freeswitch

I am working with Freeswitch on a debian server. I have an XML file which Flite reads from when someone calls the phone line. The problem is that she reads it just a little too fast. Is there a configuration setting to slow down the speech engine…
Joe
  • 151
  • 10
1
vote
1 answer

connection Issue between FreeSwitch and softphone

I'm having a connection issue between my X-Lite softphone and FreeSwitch. I get an error from the softphone saying "Failed to establish call," however SIP registration succeeds. Here are the setup details: X-Lite softphone app is installed on my…
jack_bauer
  • 11
  • 1
1
vote
1 answer

Freeswitch configure caller-ID with mod_lcr

I am working to get LCR (Least Cost Routing) working with Freeswitch. Using mod_lcr in Freeswitch 1.4.9. I have set it up and it is working. My install is actually a FusionPBX instance, but this should not matter for the following discussion. The…
nepdev
  • 381
  • 1
  • 6
  • 19
1
vote
1 answer

Freeswitch : Audio handshake failure 1 error while making call using Sipml5

I just installed FreeSwitch and successfully connected to server with user 1001. Details -> OS - Ubuntu 12.04 LTS 64 bits FS - 1.5.13b+git~20140614T114905Z~fc7a74905b~64bit OpenSSL - 1.0.1 chrome - 31, 35 webrtc is enabled. websocket -…
Anurag Rana
  • 113
  • 2
  • 7
1
vote
0 answers

can't get freeswitch to redirect calls back to kamailio

Background Information: I'm trying to follow the tutorial found here: http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc The basic set up is as follows: Two Polycom phones (192.168.1.100 and 192.168.1.102) A Kamailio server…
dot
  • 187
  • 1
  • 3
  • 11
1
vote
0 answers

freeswitch forbidden gateway error

i've stuck into a strange problem, i have Ubuntu 12.04 with static IP, installed Freeswitch with FusionPBX with easy install script from here http://wiki.fusionpbx.com/index.php?title=Easy_FusionPBX everything is working Freeswitch , FusionPBX GUI,…
jtushar53
  • 11
  • 2
1
vote
3 answers

What is the use of echo canceler in ISDN boards if all my endpoints have echo canceling?

Most digital telephony boards (BRI/PRI) have optional hardware echo canceler presented as alternative to the cpu intensive software echo canceling available in Asterisk and FreeSWITCH. I'm wondering, in all-digital communications, why echo…
Mohammad Alhashash
  • 202
  • 2
  • 9
1
vote
2 answers

Can I set up an "involuntary" conference call with Freeswitch?

I am trying to set up a SIP/RTP public announcement infrastructure. Basically there are several slave user agents that are configured to answer automatically, and a master UA which should be able to call all of them and make announcements. A way to…
Atilla Filiz
  • 235
  • 4
  • 11
1
vote
2 answers

Nagios plugin to monitor health of FreeSWITCH?

Is there a Nagios plugin to monitor the health of a FreeSWITCH server? It could either be on the server side or installed on the client. The Nagios Exchange doesn't show any results for a search of 'freeswitch'.
KJH
  • 372
  • 1
  • 14
1
vote
1 answer

What sound formats mod_rtmp supports?

Does mod_rtmp (freeswitch module) supports all sound formats FLV does? Or some limitations exist?
user102132
1
vote
0 answers

Creating a static callgroup in freeswitch

In the default configuration in Freeswitch calling 81XX adds the caller to callgroup XX. I would prefer to configure this somewhere in the XML. Any ideas where?
leiflundgren
  • 238
  • 2
  • 9
1
vote
1 answer

How to hangup Freeswitch/Sofia SIP-Calls at a 3CX correctly?

We are running a FreeSwitch-instance at host A, that is placing outbound calls to a 3CX-instance running at host B, using the SOFIA module. Everything is working fine except for the hangup: The BYE packet sent by host A is not accepted by the…
Sebastian Schmitt
  • 111
  • 3
1
vote
0 answers

tel: URI support in Freeswitch(for outgoing calls)

I have integrated a Freeswitch instance with an external SIP server and calls are working without any issues. But now I need to change the SIP server which only supports tel: URI. In the case of incoming calls, Freeswitch is able to identify the…
uts9
  • 11
  • 1
0
votes
0 answers

Freeswitch/fusion 'ghost' calls bypassing F2B/domain acl

We have been receiving 'ghost' calls from non-existent extensions. I've run into this before on asterisk systems and usually just configured the sip profile to disable guest/anon calling. However, this is Freeswitch system which uses F2B as it's…
merz1v
  • 71
  • 8
0
votes
1 answer

Error with simpl5 on call to freeswitch (onGetUserMediaError)

I have been working on setting up an HTML5 client (sipml5 by doubango: https://www.doubango.org/). The infrastructure of my setup is shown below: Server 1: sipml5 client, served through ngnix and https. Server 2: webrtc2sip setup with doubango,…
Husk Rekoms
  • 217
  • 1
  • 4
  • 15
1
2
3 4