SIP trunk between TDE600 and asterisk
When a TDE extension calls an asterisk one, TDE TRUNK USERNAME is displayed as callerid on asterisk phone.
This is when TDE sip trunk "from header"->"user part" is set as User Name
When "from header"->"user part" is set as PBX-CLIP , caller is displayed on...
I simply dial sip trunk number (80x) plus Asterisk extension
As I have a customer with no ext. names (he doesn’t want them) I’ll go for filling ext. names with ext. numbers …
Anyway, it would be interesting to know why of this behavior.
A tde600 is connected to asterisk pbx via sip trunk .
When a TDE phone (analogic or digital) calls an asterisk one, its NAME is displayed on called ringing ip phone.
If I remove NAME from TDE extension field and leave it blank, the asterisk called phone DOESN'T show panasonic extension number...
Well, I solved it by adding 7 as prefix in asterisk dial out string.
Being 7 the TIE line code in TDE600, it is used to tell the rest of numbers to be dialed towards TDE600 PSTN line
So if 12345678 is the called number, asterisk outbound rule add 7 and route to SIP trunk ...
I set up a sip trunk between freepbx (asterisk) and TDE600
Internal call are working fine.
I need to use TDE600 as voice gateway to telco provider: how to ?
If from asterisk I dial 12345678@TDE600ipaddress I get call answered by TDE operator
The same if I add the prefix I dial locally for using...
I set up a trunk between TDE600 SIPGW-16 ad a Freepbx system (pjsip trunk)
All is working fine except calls drop after 30 seconds.
Sip connection is over a site to site VPN, where two sites have different subnet.
I'm able for now to get just the asterisk trace :
-- Channel PJSIP/TDE-0000000d...
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