3CX V.12 & GrandStream HT503: no pstn access

[Gerard_Smith]

Hello all,

I am having some issues with getting a GrandStream HT503 to access the pstn line from an extension.
The HT503 is setup as a VOIP/PSTN Gateway with port of 10000. I have just setup a quick 'n dirty outgoing rule for 7 digits to use the HT503. (in Caribbean, we can use 7 digits locally).

The HT503 has a single FXS and a single FXO port, the FXO & FXS Ports are registering correctly.

I am connecting with a Blackberry Z10 (running Android version of 3CX for v12) via 3G, resolving to a "No-IP" DDNS into my PBX. I am also connecting on a iPhone 4S running the 3CX v12 via local WiFi.

Both Mobile phone are working 100% (presence, audio, accepting and originating calls etc)

I can make calls to the FXS port (Labelled Ext 107) from BB-Z10 (Ext 104)
I can also dial a 7 digit number from the BB-Z10 and it will ring on the 10000 port. The issue i get is that although the port says 10000, phone on the FXS side (Ext 107) is what rings. When i look at the extension status, Ext 107 is green (not ringing)

I believe this to be a setting in the HT503 that i am overlooking, where is pushing the call over to the FXS side. I just can't find the setting.

The HT503 can use the FXS line to answer calls coming into the FXO port, and you can also pickup the FXS line and press *99 to directly access the PSTN line as well. Using the phone on FXS via the *99 route does not engage ext 107 on the PBX.

Here is a dump:

01-Jun-2014 19:38:55.428 [CM505002]: Gateway:[GrandStream HT503 PSTN Gateway] Device info: Device Not Identified: User Agent not matched; Capabilities:[reinvite, replaces, able-no-sdp, recvonly] UserAgent: [Grandstream HT-503 V2.0A 1.0.10.9 chip V2.2] PBX contact: [sip:[email protected]:5060]
01-Jun-2014 19:38:55.030 [CM503025]: Call(C:26): Calling T:Line:10000>>4289164@[Dev:sip:[email protected]:20346] for L:26.1[Extn]
01-Jun-2014 19:38:54.997 [CM503027]: Call(C:26): From: Extn:104 ("BlackBerry Z10- Gerard" <sip:[email protected]:5060>) to T:Line:10000>>4289164@[Dev:sip:[email protected]:20346]
01-Jun-2014 19:38:54.997 [CM503004]: Call(C:26): Route 1: from L:26.1[Extn] to T:Line:10000>>4289164@[Dev:sip:[email protected]:20346]
01-Jun-2014 19:38:54.997 Line limit check: Current # of calls for line Lc:10000(@GrandStream HT503 PSTN Gateway[<sip:[email protected]:20346>]) is 0; limit is 2
01-Jun-2014 19:38:54.997 Call(C:26): Call from Extn:104 to 4289164 matches outbound rule 'Rule for GrandStream HT503 PSTN Gateway'
01-Jun-2014 19:38:54.994 [CM503001]: Call(C:26): Incoming call from Extn:104 to <sip:[email protected]:5060>


Dialing plan on HT503 was changed to { [4268]xxxxxx }.
Originally it was { x+ | *x+ | *xx*x+ }, but that didn't make a difference either.

Any help would be appreciated.

Regards

Gerard

 

[leejor]

Off hand it sounds like an option in the Grandstream that would normally allow the habndset plugged into the FXS port , to answer calls coming in on VoIP. I've used an older grandstream model that had the same features., but not your version. I'll have to see if i can find the menu options on-line. they should be in the section that also has something about selecting One, or Two stage dialling.
In the combination FXO/FXS units, threre are usually a number of "cross-over" options that have to be diabled when using the device with 3CX as a trunk and and extension.

 

[Gerard_Smith]

Hi Leejor,

I saw the setting about one and two stage dialing, it was on two stageoriginally, which was the default, but i changed it to one stage as that was recommended if using as a gateway. Didnt make a difference.

There was also another option about "Tel URI:" and i tried the "user=phone", "enabled" and "disabled" option on that as well.

Life line mode was on "Auto", i tried it on "Disconnected" as well

Regards

Gerard