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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.

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17 views

FusionPBX/Freeswitch no ACL module installed

I have a FusionPBX box I inherited and I'm trying to configure the domain-based ACL controls, but the Fusion UI doesn't show any options for 'Access Controls' and no "mod_acl' in directory. The docs ...
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0answers
33 views

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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1answer
94 views

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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29 views

How to remote control a VOIP conferences - FreeSwitch mod_conference

Is there any standard way to remote control conferences in FreeSwitch (or generally in VOIP related standards)? I don't mean by DTMF because i know this method, i'm rather asking about some protocol ...
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0answers
19 views

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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1answer
78 views

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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78 views

Freeswitch: Some phones send their internal ip in recv while trying to register

My FS has external ip only, is not NATed. Some phones send REGISTER with their internal ip and Freeswitch send them 401 Unauthorized. Internal profile settings: Name internal ...
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1answer
296 views

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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46 views

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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1answer
165 views

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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1answer
399 views

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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1answer
250 views

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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1answer
2k views

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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1answer
456 views

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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1answer
381 views

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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0answers
818 views

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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1answer
176 views

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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1answer
833 views

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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2answers
3k views

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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1answer
547 views

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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0answers
3k views

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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0answers
551 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
356 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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1answer
226 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 argument/usr/src/...
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1answer
2k 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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0answers
2k 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 (192....
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0answers
887 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
472 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 wondering,...
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2answers
843 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
2k 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
757 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 ...
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2answers
3k 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
2k 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
7k 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
3k 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
900 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
382 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
156 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
1k 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
6k 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 http://wiki.fusionpbx.com/...
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0answers
132 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
2k 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
7k 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
1k 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
1k 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, ...
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1answer
1k 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 to.....
3
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1answer
2k 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
501 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 ...