Using Grandstream FXO adapter to a Valcom paging system, asterisk doesnt complete the call, just rings

[ghurty]

I have a Valcom paging system that works by providing a line. For example, I can hook up an analog phone directly to it, it will get dial tone, and I dial “#11” and it will page.

I hooked it up via a grandstream FXO adapter. The FXO adapter is configured properly as I can use it to make calls over a POTS line. But what happens when I call the the valcom, the valcom answers, but the PBX continues to ring. It is as if Asterisk/Grandstream are not getting the signal that the line was answered. Any suggestions?

Thank you

[arielgrin]

Call log please

[Stewart1 (Stewart)]

In the Grandstream, set Enable Call Answer Supervision (or similar parameter, depending on model) to No.

If you already had this setting, or it doesn’t help, post a log including SIP trace, and the model of your FXO device.

[ghurty]

Enable call supression is currently set to ch1-4:N;.
When I changed it to ch1-4:Y;
the call doesnt even connect.

The 88 is me telling it to use that outbound route
Call Logs when I have it as N is:
[2019-08-04 18:18:57] VERBOSE[10759][C-00002bbb] app_dial.c: Called SIP/604040/#11
[2019-08-04 18:18:57] VERBOSE[10758][C-00002bbb] app_dial.c: SIP/Voip Innovations-0000234a requested media update control 26, passing it to Local/8811@from-internal-00000029;1
[2019-08-04 18:18:57] VERBOSE[10759][C-00002bbb] app_dial.c: Local/8811@from-internal-00000029;2 requested media update control 26, passing it to SIP/604040-0000234b
[2019-08-04 18:19:00] VERBOSE[10759][C-00002bbb] app_dial.c: SIP/604040-0000234b is ringing
[2019-08-04 18:19:00] VERBOSE[10758][C-00002bbb] app_dial.c: Local/8811@from-internal-00000029;1 is ringing

When I have it as Y they are
[2019-08-04 18:20:20] VERBOSE[10898][C-00002bbc] netsock2.c: Using SIP RTP TOS bits 184
[2019-08-04 18:20:20] VERBOSE[10898][C-00002bbc] netsock2.c: Using SIP RTP CoS mark 5
[2019-08-04 18:20:20] VERBOSE[10898][C-00002bbc] app_dial.c: Called SIP/604040/#11
[2019-08-04 18:20:20] VERBOSE[10898][C-00002bbc] app_dial.c: Local/8811@from-internal-0000002a;2 requested media update control 26, passing it to SIP/604040-0000234d

Thank you

[arielgrin]

That doesn’t seem to be a log of a call out of the FXO. You should provide more info, like screenshot of the trunk configuration, both on FreePBX and Grandstream and a log of an actual call from an extension to the pager. Also the outbound route used to call the pager.

[ghurty]

Hi,

That was a call in via me using a DISA. When I have the GXW410X hooked up to a regular landline instead of the valcom the call complete without any issues. Attached are screenshots from my pbx, and grandstream

Thank you

[arielgrin]

Make a call to the pager and post the log

[ghurty]

I am trying to send #11 do the Valcom. So I have an outbound route that when I dial 88XX it should strip the 88 and append a # before the 11.

Here is the log when I have the grandtream set to ch1-4:Y; by Enable call supervision.

Here is the log when I have the grandtream set to ch1-4:N; by Enable call supervision I hung up after letting it just ring and ring

Thank you

[Stewart1 (Stewart)]

It appears that the GXW default dial plan does not allow numbers beginning with # .
Try setting it to
{ [x#]+ | [x*]+ }
as shown in

gxw410x_usermanual_english.pdf

I believe that Call Answer Supervision should be No.

If you still have trouble, type
sip set debug peer 604040
at the Asterisk console, then make a test call. Post the log (which will now include a SIP trace).

[ghurty]

I adjusted the dial plan but it still didnt work. Here is the log with debug enabled.

Thank you