I have this situation: I can make calls internally, I can make inbound calls but I can't make outbound calls.
Thanks in advance.
These are my devices: * asterisk 11.8.1 = 192.168.1.22 * sipphone grandstream gxp2160 = 192.168.1.5 * gateway audiocodes mp-114 2fxs-2fxo = 192.168.1.4 port 1 (FXS) connected to an analog phone port 3 (FXO) connected to the PSTN
These are my sip.conf and extensions.conf files:
Our environment is a register free setup, and our phones are set as host=dynamic. The problem we are experiencing is for inbound calls:
Name/username Host Dyn Forcerport ACL Port Status Realtime 222/222 (Unspecified) D N A 0 Unmonitored Cached RT
So when we DIAL 222 we get:
WARNING: app_dial.c:2198 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Unknown)
My question is how can we get Asterisk to fill in the gaps (ie, ipaddr, port) for a dynamic peer in a register free environment.
We've been having a weird issue... It is happening more often in the last few months...
Most inbound calls, we have in our dialplan before Queue():
So when the call rings a member, softphone will show this string ....
The issue is that sometimes the string showing in the softphone is not the same. Its a string from a past call, in the latest case I've seen, from about 40 days ago!!
User took a screenshot, I've searched for that uniqueid showing in softphone in cdr, and that string was valid for a different call 40 days ago!!
I searched full log, and…
I would like to develop a Call Center Dialer (outbound and inbound calls)
and it would use AMI method to communicate with Asterisk Server.
A daemon would need to run in the background, would you recommend coding in PHP or Node.js? which would be much faster and stable.