Questions tagged [freeswitch]
FreeSWITCH is an open-source telephony platform designed to facilitate the creation of voice and chat driven products.
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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 ...
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1
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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 ...
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All traffic on port 5060 gets lost in whole network
I am having issues forwarding port 5060 in my firewall/router to a FreeSWITCH server.
Our firewall/router is an Uniquiti Edge Router X. The server's local IP is 10.0.0.216, the router's 10.0.0.1. ...
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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 ...
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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 ...
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FreeSWITCH Dual Stack IPv4/IPv6
I'm currently trying to understand how I can enable my freeSWITCH to talk both IPv6 as well as IPv4. Currently, I thought it was going to be easiest to first create a set-up which works on IPv4 and ...
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275
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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, ...
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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 ...
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Freeswitch/FusionPBX trying to list all inbound routes/destinations and their associated actions
I mainly work with Asterisk/FreePBX but just recently started working with Freeswitch on a large scale so forgive me as I'm only now starting to explore a lot of the interface.
Issue: User wants to ...
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Using Asterisk as a gateway to Provider
I have 2 sip servers on different LANs. Freeswitch and another is Asterisk.
Asterisk sits on a VPN with a provider and who has provided DIDs. All users register on Freeswitch. How can I route calls ...
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Freeswitch internal/external profile calling
On a dual NIC Freeswitch SIP Server, how can I enable calls between internal profile and external profile? I have eth0 192.168.1.10 , and eth1 public IP 41.x.x.x
sofia status profile internal
--------...
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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 ...
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533
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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 ...
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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 ...
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Least Costing Routing in Freeswitch - Cannot Read Config file
We have a Freeswitch installation as part of a FusionPBX GUI system.
FusionPBX version 3.7.1, Freeswitch version 1.4.9. Database used by FusionPBX is Postgres.
The install compiles the Least Cost ...
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Best practices for transcoding OPUS/G711 using freeswitch
Can someone share their experience of transcoding OPUS/G711 ans vice versa using Freeswitch? I am getting call quality issues even if there is a single call on the server. I am getting crackling noise ...
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freeswitch received audio delayed for 20-30 seconds
I manage to make a very simple configuration for basic phone to phone calls using FreeSwitch, When i Calls A -> B (and B answered), A can hear B instantly, but B cannot hear A, after waiting for 20-30 ...
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Nagios plugin to monitor FreeSWITCH
I am trying to configure Nagios to monitor FreeSWITCH as mentioned at https://github.com/kjhosein/nagios-freeswitch-plugin .
I have downloaded the script from git and followed listed steps.
On ...
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freeswitch call getting abandoned
i have a freeswitch box that successfully handles incoming calls to a certain DID number, but when i call in from another DID number (which happens to be from Asterisk) i get a message that the call ...
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FusionPBX Failed(405) on Cisco 303 IP phone
Since yesterday, my phone started throwing a Failed(405) error when trying to register to a FusionPBX server. I have tried resetting to factory settings and reinstalling the FusionPBX server on ...
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cant connect esl to freeswitch
im trying to connect an application to FreeSWITCh using ESL. the code is ok (it works on other machines) but on one installation its not connecting.
im trying to connect to localhost:8021
but the ...
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FusionPBX: SIP/2.0 405 Method Not Allowed, SIP Phones not registering
I followed below steps to install FUsionPBX/Freeswitch. Got to the point where phones are registering with SIP extension created.
But when I changed the default gateway of the freewsitch box and ...
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Hot to integrate sipJs and freeswitch?
I am trying to integrate sipjs with freeswitch. below is the config I am using
var config = {
// Replace this IP address with your FreeSWITCH IP address
uri: 'sip:1002@***.***.1.170',
// Replace ...
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0
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5k
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How to Properly Route FreeSWITCH incoming calls to external SIP URI
I am very very new to Freeswitch. I am running FreeSWITCH Version
1.5.15b+git~20141120T035109Z~79de78a0fb~64bit (git 79de78a 2014-11-20
03:51:09Z 64bit) on a CentOS 6.6 64-bit Virtual Machine.
I am ...
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0
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769
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Unable to connect Sip softphone to Freeswitch
I just started freeswitch 1.4.5 on ubuntu 14.04 x64 and tried to connect SIP phone to freeswitch, but it is not registering. I am using default configuration.
I am using 1001 as username. 1234 as ...
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560
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Voicemails cutting off in freeswitch
We are using a freeswitch PBX. Any incoming voicemail getting left is being cut off after 60 seconds - i.e. the call is actually getting disconnected at that point. Regular calls are not, just ...
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how to install lafilefixer in ubuntu
I am getting following error while installing freeswitch on Ubuntu.
libtool: link: cannot find the library /usr/src/freeswitch/libs/apr-util/xml/expat/lib/libexpat.la' or unhandled argument/usr/src/...
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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 - ...
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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....
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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
...
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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,...
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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 ...
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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 '...
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Best Guest OS for running FreeSWITCH under Proxmox
We have a lot of Asterisk systems running on dedicated machines and would like to use FreeSWITCH to replace a number of them. One of the advantages of FreeSWITCH is supposed to be that it is doing ...
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How to improve call quality freeswitch?
Currently I have a system that makes use of FreeSWITCH for outbound calls via SIP External with flowroute and works well, but some users complain about the quality of the call. The system is running ...
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location of sofia profile
I am not able to connect to freeswitch server because it is not configured correctly. I get an error...
call failed: forbidden
The freeswitch log says...
2013-01-14 02:33:33.014079 [WARNING] ...
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How to stop freeswitch by the command line?
freeswitch has it's own CLI and it is possible to shut it down with shutdown command issued within this CLI.
But is it possible to shut it down bu separate command, for example, by shell script?
user102132
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Why are SIP calls via my server silent?
I have FreeSWITCH SIP server up and running. It has public IP and sits behind 1-to-1 NAT (it's Amazon EC2 instance actually).
I can connect to it, make a call to other endpoint (namely, my android ...
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Is it possible to change freeswitch user password on hot?
Is it possible to change freeswitch user password on hot? I mean I have freeswitch running and wish to change 1000's password from 1234 to something else without restarting freeswitch?
I an changing ...
user102132
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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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257
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Using an extension to block a caller
I have a couple of SIP phones and use callcentric. I get a lot of junk calls. I'd like to implement the following feature and would like some suggestions on how to do this:
Once I get a junk call, I ...
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How can I call my voicemail in Bluebox Freeswitch?
I'm using Bluebox to manage Freeswitch. How can I call my voicemail? If I dial my extension it takes my to the part where I need to record a voicemail.
If I dial 4000, *89 or 879 its just doesn't ...
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Cannot connect softphone as a FreeSwitch Extension
Having successfully configured and maintained few Asterisk based installations, I have now been provided a task to configure FreeSwitch SIP server.
ISO downloaded from
http://wiki.fusionpbx.com/...
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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?
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How to pass/process extension numbers while calling Freeswitch from outside SIP gateway?
Suppose I have Freeswitch, which has SIP gateway configured. So, the local users of this Freeswitch can call outside via
<extension name="russia">
<condition field="destination_number" ...
user102132
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Can you please explain the difference between PBX and soft switch in less technical terms?
Asterisk is an open source PBX and FreeSWITCH is an open source soft switch.
Source: http://www.freeswitch.org/node/117
Can you please explain the difference between PBX and soft switch in less ...
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Are there parallels to Asterisk AMI and Asterisk AGI in FreeSWITCH?
Asterisk has Asterisk Manager Interface (AMI) and Asterisk Gateway Interface (AGI), using which one can trigger PHP scripts at certain events from Asterisk. Using the same PHP scripts can also ...
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How can I record a call transfer with freeSwitch?
I am using freeSwitch to terminate calls that originate on the local network from an IVR system. I have it working with several different voip terminators, and I can record sessions successfully, ...
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FreeSWITCH & OpenSIPS - How do I avoid duplicating extensions in multiple FreeSWITCH servers?
I want to use FreeSWITCH instead of Asterisk because of it's performance compared to Asterisk. I know that FreeSWITCH can be a full PBX or just run parts (modules) to do only the things I want it to.....
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1
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2k
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asterisk/freeswitch in nat/no-nat setup
my current setup - i use bunch of sip hard-phones around few offices. all devices have two sip accounts configured - one on internal sip proxy [for calls between the branches], another - at 3rd party ...
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