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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.

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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
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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
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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
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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
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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
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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
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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
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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
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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
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What sound formats mod_rtmp supports?

Does mod_rtmp (freeswitch module) supports all sound formats FLV does? Or some limitations exist?
user102132
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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
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Unable to access port 8000 on my Debian server

I have a Debian server running fusionPBX, I installed it using this official script. It uses nginx to host the application, I tried adding another server (Django) in the config file of nginx to support another server that's running on the same…
Vaibhav Vishwakarma
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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
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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
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Freeswitch drops calls after 32 seconds

So I've installed Freeswitch on a raspberry PI 3 and it's dropping calls after 32 seconds. I've googled extensively and this appears to be a common problem but all of the people with the problem had complicated setups with external gateways, VPNs,…
MikeKulls
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