Solved SIPtrunk to Cloud PBX 15.5

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Matt Waechter

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The SIPtrunk is registered to the system and we can make calls out. I have the trunk set to ring to a ring group but when the call comes in it rings for a split second then goes directly to a mailbox. If I change the destination to IVR, inbound calls flow to the IVR fine. If I change the destination to a known working extension the call rings for a split second then goes to voicemail. Also if I dial the code (800) for the ring group it does ring all phones and roll after 20 seconds to the IVR as programmed. I removed the SIP trunk and ring group and rebuilt them to no-avail. This setup did work until yesterday when the Google server became unresponsive and we had to reset it.
 
And the logs say....?
 
To ring group:

07/12/2017 2:10:22 PM - [CM503003]: Call(C:76): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
07/12/2017 2:10:22 PM - [CM503003]: Call(C:76): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
07/12/2017 2:10:22 PM - [CM503003]: Call(C:76): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from x.x.x.x:24032
07/12/2017 2:10:22 PM - [CM503003]: Call(C:76): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Cancelled/INVITE from x.x.x.x:55172

To 1 extension:
07/12/2017 2:11:49 PM - [CM503003]: Call(C:77): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5488
07/12/2017 2:11:49 PM - [CM503003]: Call(C:77): Call to <sip:[email protected]:5060> has failed; Cause: 487 Request Terminated/INVITE from 127.0.0.1:5080
 
Make sure that the extensions in the Ring group are set to available. Also check in case any extensions has ring my mobile simultaneously as the phone might be dropped by the carrier if the call is going out. If you have the ability to open a ticket please do as we will need to see a wireshark capture as well as the matching logs of the PBX to see why the call is being dropped.
 
Ticket opened:
#148508
 
I found the resolution here: https://support.siptrunk.com/hc/en-...X-IP-PBX-V15-SIPTRUNK-com-Trunk-Configuration
The default template sets the Host as: gw.siptrunk.com
At this stage, many of the configuration settings have automatically been filled in for you, but it is still necessary to configure your account based on your needs. Most specifically:
  • Registrar/Server/Gateway Hostname or IP: gw1.siptrunk.com and port 5060.
I changed to gw1.siptrunk.com and calls flow as expected.
 
Glad to hear the issue is resolved :)
 
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