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

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-2
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0answers
20 views

sip from command line

I am definitely remember, that somewhere on freeswitch.org site there was an example of how to initiate a SIP call from command line. It was not by using sipcmd, but either some general linux commands ...
-4
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0answers
24 views

Sound pass only in one direction if two SIP telephones connected [on hold]

I have installed virtual Ubuntu 12.04 inside Windows 8 host. It has own separate IP address. Builded from source freeswitch version 1.5.13b runs inside Ubuntu. Two Jitsi sip telephones runs on hosting ...
0
votes
1answer
13 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 :)
0
votes
0answers
19 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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votes
0answers
11 views

Freeswitch: Registration Failed with status Service Unavailable [503] [closed]

While tryint to register with external SIP trunk, my FS gives the following error: Registration Failed with status Service Unavailable [503] May this mean network/firewall problem, or this is ...
0
votes
1answer
40 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 ...
1
vote
1answer
71 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 - ...
1
vote
0answers
218 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 ...
1
vote
0answers
66 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 ...
0
votes
3answers
100 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 ...
1
vote
2answers
140 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 ...
0
votes
2answers
215 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 ...
0
votes
1answer
153 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
votes
2answers
874 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 ...
0
votes
1answer
233 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] ...
0
votes
2answers
663 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?
0
votes
2answers
458 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 ...
0
votes
1answer
348 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 ...
1
vote
1answer
158 views

What sound formats mod_rtmp supports?

Does mod_rtmp (freeswitch module) supports all sound formats FLV does? Or some limitations exist?
4
votes
2answers
95 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 ...
0
votes
0answers
165 views

want freeswitch events to trigger curl or command

I'd like to trigger a CLI command to execute or call a URL when certain freeswitch events fire. I really really don't want to write an outbound socket daemon that does this - its just another ...
0
votes
1answer
607 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 ...
1
vote
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 ...
1
vote
0answers
56 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?
0
votes
1answer
723 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" ...
9
votes
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 ...
0
votes
1answer
411 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 ...
1
vote
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
votes
0answers
357 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, ...
4
votes
1answer
880 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
votes
1answer
822 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
votes
1answer
325 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 ...