[PBX] VOIP PBX Options for a small office. The effort to install FreePBX or a similar Asterisk distribution on new hardware or a virtual server is small, compared with the work of planning. KBIT VoIP Asterisk FreePBX. (Media Gateway Control Protocol SCCP (Cisco® Skinny®) SIP (Session Initiation Protocol) UNIStim. SUPPORTED CODECS.

1. FreePBX is licensed under the GNU General Public License (GPL), an open source license. FreePBX can be installed manually or as part of the pre-configured FreePBX Distro that includes the system OS, Asterisk, FreePBX GUI and assorted dependencies.
2. Nortel non-SIP IP telephones uses a communication protocol called Unistim. This protocol has. Cd chan_unistim-1.0.0.5f make make install. FreePBX uses this for.
3. Ks16alt install which gives 2.5.2.2 of Freepbx and Asterisk 1.6.2.6 and Centos 5.2 We can confirm that neither DND and Voicemail work on the Unistim phones with this Config. We believe this is an issue with the UNISTIM as delivered with 1.6 of Asterisk and we had this working previously with 1.4.

System:
Ubuntu 12.04 LTS 64Bit
Asterisk 11.2.1
FreePBX 2.11
Hi Guys,
I have a problem to issue calls with SRTP.
The FreePBX configured to work with TLS and libSRTP installed well. I used this tutorial to configuration: https://wiki.asterisk.org/wiki/display/AST/Secure+Calling+Tutorial
When I am initiating new call with TLS only the calls working well, when SRTP added and initiate new call, I received this error:
Using SIP RTP TOS bits 184
Using SIP RTP CoS mark 5
[2013-04-21 10:39:00] WARNING[16542][C-00000352]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies

Install Freepbx On Ubuntu

[2013-04-21 10:39:00] WARNING[16542][C-00000352]: sip/sdp_crypto.c:170 sdp_crypto_activate: Could not set SRTP policies
[2013-04-21 10:39:00] WARNING[16542][C-00000352]: chan_sip.c:10427 process_sdp: Can't provide secure audio requested in SDP offer

Freepbx Setup

When I tried to Google it, I could not find any relevant posts to solve this issue.
If more information needed, just let me know and I will upload it.
Thanks,
Noy

I need help. I have some i2004 phones, an i2002, and an 1100e. I had all calls working between the sets and some SIP soft phones. I could not access voicemail and research led me to change the context from 'default' to 'from-internal'. Since then I cannot get the phones to work flawlessly. I stupidly didn't have a backup of my unistim.conf file. And now I've made so many changes over the past week I can't go back to the working config.

The current issues is that the UniSTIM phones do not appear to be sending the audio from the microphone. UniSTIM to UniSTIM phones are silent, though one can hear themselves in their handset indicating that the microphones work. UniSTIM to SIP phones also do not transmit audio from the UniSTIM set. I've tried all 4 rtp methods in the config file with no success. With '0' I get no audio at all in either direction. IIRC, I was using RTP method '1' originally. If you have any ideas, I'm open to trying anything.