The company I work for just contracted with an answering service that handles our overflow and after-hours calls. We’re running FreePBX 14.0.13.4 and Asterisk 13.22. The answering service also runs Asterisk. (I’m adding spaces in the domains in this post since they this forum thinks they’re links and new users aren’t allowed to post links). They gave us a URI to send calls to, which is 1000@asterisk. domain. com (replacing “domain” with their actual domain). They use our IP address for authentication, so we don’t have a username or password. They also only accept the ULAW codec. I’ve spent countless hours trying to figure out how to get this to work.
I’ve tried adding a line to the extensions_custom.conf file, but get a “Your call cannot be completed as dialed” message when trying to call extension 1000:
exten => 1000,1,Dial(SIP/asterisk. domain. com,30)
I also get the following in the Asterisk log:
[2019-08-27 09:02:51] VERBOSE[12096][C-000000ca] netsock2.c: Using SIP RTP TOS bits 184
[2019-08-27 09:02:51] VERBOSE[12096][C-000000ca] netsock2.c: Using SIP RTP CoS mark 5
[2019-08-27 09:02:51] VERBOSE[18422][C-000000ca] pbx.c: Executing [1000@from-internal:1] Dial(“SIP/101-00000045”, “SIP/asterisk. domain. com,30”) in new stack
[2019-08-27 09:02:51] VERBOSE[18422][C-000000ca] netsock2.c: Using SIP RTP TOS bits 184
[2019-08-27 09:02:51] VERBOSE[18422][C-000000ca] netsock2.c: Using SIP RTP CoS mark 5
[2019-08-27 09:02:51] VERBOSE[18422][C-000000ca] app_dial.c: Called SIP/asterisk. domain. com
[2019-08-27 09:02:51] WARNING[12096][C-000000ca] chan_sip.c: Received response: “Forbidden” from ‘“Ryan Test” <sip:101@[masked IP address]:5070>;tag=as5370cc75’
[2019-08-27 09:02:51] VERBOSE[18422][C-000000ca] app_dial.c: Everyone is busy/congested at this time (1:0/0/1)
They won’t be making any calls into our PBX. We will only be sending calls to them. To make things a little more confusing, we use PJSIP and they use CHAN_SIP. I have both enabled on our server and run CHAN_SIP on port 5070. But their server knows we’re at port 5070 and responds correctly to that port, so I don’t think that’s an issue? The phone I’m using to test this has a CHAN_SIP extension. Do I need to create a trunk and/or outbound route? This is what they have on their server for us:
[genesis]
host=[masked IP address]
port=5070
type=friend
allow=ulaw
directmedia=no
insecure=invite,port
context=genesis
dtmfmode=auto
deny=0.0.0.0/0.0.0.0
permit=[masked IP]/255.255.255.255
For the time being, we have a workaround in place… They gave us a DID to forward calls to. But that’s not ideal since they then get our caller ID info (not the customer’s), it uses 2 of our 8 channels with our VOIP provider, and we get charged long distance for each call. Any help would be greatly appreciated!!