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

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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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26 views

Freeswitch Database Hangup Causes

I am looking at our Freeswitch database for the VoIP service my company provides and trying to investigate an issue with a particular router. I see from a CDR (Call Detail Record) in the Freeswitch ...
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78 views

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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1answer
46 views

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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42 views

can't connect E1 digium te420 to freeswitch on CentOS

i'm experiencing an unknown problem for me, while i'm trying to connect digium te420 to freeswitch running on CentOS, digium card gives an alarm light while i'm connecting it. any ideas, can IRQ ...
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1answer
115 views

How to know freeswitch version?

If I print version in fs_cli, it says freeswitch@internal> version FreeSWITCH Version 1.5.7~32bit ( 32bit) But latest version is 1.2.23 :)
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57 views

SIP/2.0 400 Sip message body is shorter that is indicated by the content length

During SIP communicating between my application, Freeswitch, and nextiva trunk, I am getting the following error message from 208.73.146.95 (Nextiva server): SIP/2.0 400 Sip message body is shorter ...
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1answer
78 views

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 ...
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1answer
280 views

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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558 views

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 ...
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179 views

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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3answers
127 views

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 ...
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2answers
244 views

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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2answers
374 views

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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1answer
233 views

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 ...
2
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2answers
1k views

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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1answer
441 views

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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2answers
1k views

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?
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2answers
718 views

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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1answer
397 views

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 ...
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1answer
197 views

What sound formats mod_rtmp supports?

Does mod_rtmp (freeswitch module) supports all sound formats FLV does? Or some limitations exist?
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2answers
97 views

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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1answer
705 views

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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3answers
2k views

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 ...
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0answers
70 views

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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1answer
861 views

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" ...
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2answers
2k views

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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1answer
475 views

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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0answers
32 views

Is it possible to call user both by number and by name in Freeswitch?

How should I configure freeswitch directory entry so that it will be possible to call this user both by numeric number and by textual username?
2
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0answers
422 views

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, ...
5
votes
1answer
992 views

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 ...
2
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1answer
905 views

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 ...
2
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1answer
342 views

Call issue with Freeswitch

I am testing the following with Freeswitch and different devices (nokia n900, nokia e60, ekiga) and have similar results between them. On the Freeswitch server (1.0.4 in multi-tenant mode) I have ...