FreeSWITCH is an open-source telephony platform designed to facilitate the creation of voice and chat driven products.
1
vote
1answer
121 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
78 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
1answer
132 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
181 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
162 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
105 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
82 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
114 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
401 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
973 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
42 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
427 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
1k 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
293 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
30 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
251 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, ...
3
votes
1answer
747 views
FreeSWITCH & OpenSIPS
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
723 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
290 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 ...
