I have a Grandstream HT488.
The FXO is connected to the PSTN line. The FXS is connected to an analog phone.
The WAN is connected to the switch. The WAN is assigned the ip address 192.168.2.128
On this HT488, the default PSTN access code is *00
From the analog phone if I want to dial a local number 765-4321 then
I dial *007654321 and it works immediately every time.
The Asterisk system is on the same LAN with the ip address 192.168.2.126
The HT488 is the only analog interface to the Asterisk system.
The HT488 is configured to ‘Forward to VoIP’ so that
’Calls are unconditionally forwarded to the specified VoIP phone number for all incoming PSTN calls.'
So my incoming PSTN calls are forwarded to a specific extension (nnn) on the Asterisk system.
So far all this works:
- All incoming PSTN calls do arrive at the Asterisk nnn extension.
- Unanswered calls go to the voicemail for extension nnn.
- There is little or no echo.
- There is almost no noise.
The outbound calls from any of the extensions are a bit challenging.
The default 9 to get an outside line is still there in freePBX.
It uses the one and only (HT488) SIP trunk.
Besides the above analog phone from any extension, if I want to dial the same local number 765-4321 then
first I have to dial 9*00 to get to the HT488’s dial tone
(wait a little bit)
then I enter the number 7654321
Simply dialing 9*007654321 does not work.
Question - How do I combine the “9*00” and the “7654321” strings?
Such that there is a bit of a pause in the middle
to accomodate the HT488’s dialtone.
I would like to put the unified string as a Contact in my X-Lite.