I have the following system up and working for the below:
- I have a FreePBX Server setup with SIP Trunking to a provider.
- I have a ShoreTel Phone System
- I have SIP Trunks setup between the ShoreTel and the FreePBX.
I can register a softphone with the FreePBX server and make calls to the VOIP Provider. I can also dial an extension from the softphone to the ShoreTel system and it works just fine. I Can dial from the ShoreTel system to the FreePBX system and it works just fine.
What is not working:
When I dial from the ShoreTel system to the VOIP provider, it makes it to FreePBX but I get an error back from the VOIP provider. See below. Here is what is happening and trying to figure out how to remove the /s.
When a call is made from the Soft Phone to the FreePBX system and out to the VOIP Provider is works and this is what it displays:
It appears not to append the calling number when it is from the ShoreTel to the FreePBX to the VOIP provider?
Any help would be appreciated.
– Executing [[email protected]:8] Dial(“SIP/Asterisk-0000010b”, “SIP/voip/s,300,”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
– Called voip/s
– Got SIP response 503 “No Circuit Available” back from XXX.XXX.XXX.XXX
– SIP/voip-0000010c is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [[email protected]:9] Set(“SIP/Asterisk-0000010b”, “CALLERID(number)=+1999999999”) in new stack
– Executing [[email protected]:10] Set(“SIP/Asterisk-0000010b”, “CALLERID(name)=Test User”) in new stack
– Executing [[email protected]:11] Hangup(“SIP/Asterisk-0000010b”, “”) in new stack
== Spawn extension (ext-trunk, tdial, 11) exited non-zero on 'SIP/Asterisk-0000010b’