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.
i have create i trunk Sip between 2 servers in the same network
when i call a number (inbound calls) in the first server i can forward this number to my sip 222 in the second server
exten => 0522xxxxxx,1,Dial(SIP/222@trunk_created,30)
my question if there is any possibility to GOTO a context in the second server after like below
exten => 0522xxxxxx,1,Dial(SIP/222@ trunk_created,30)
same => 0522xxxxxx,n,GoTo (context in the second server)
thanks and regards
OK, it's been a while since I drank from the pool of wisdom hear on the list.
After cracking my head against the wall for a few days trying to figure this out, I have decided to swallow my pride and take the drink.
So, on to my question:
I have some agents/operators setup in sip.conf which point to a context where I have just about disabled outbound calls (only specific numbers can be dialed).
The purpose of this is to allow the inbound calls to come in, then if the operator has a need, they transfer the call to a pre-defined extension which…
Hello, I have noticed some occasional one way audio on a specific sub set of calls in my system. First, let me be more specific about what I mean about occasional one way audio. Unlike most of the posts I've seen (where the end fix was either NAT'ing or RTP issues) the calls in question will begin and progress just fine, and then in one direction (always the same direction) audio stops for a second or two. When this occurs, it seems like every other, or every third word is cut out for 5-15 seconds (estimate, I have not timed…
Converting this customer from a MiTel system to asterisk. Discovered that the inbound calls from the T1 are going to extension 366. (This was mapped in the MiTel for some arcane purpose.) The dial plan I am currently using is shown below. When loading the dial plan, I get this warning:
WARNING: pbx_config.c:1561 pbx_load_config: The use of '_.' for an extension is strongly discouraged and can have unexpected behavior. Please use '_X.' instead at line 331 of extensions.conf
Question: Do I need to worry about this warning?
I'm a little leery of just using 366 in the dialplan, since the company we…
If this is the wrong place to post this I'm sure someone will let me know. :-)
I'm looking for a reliable, inexpensive call termination service (SIP). The one I am presently with does not seem to know what IPs they send inbound calls from, and it is maddening to keep up with the FW changes necessary, not to mention the calls which are not connected.
Kind regards, Chris