[quote=“dpctech”]ok done now what?
Where I can see the log you need?[/quote]
autoquote
Solution was already posted. Just direct both trunks into the same context and replace my digits with yours.
Until you have contact= set in the peer definition all the incoming INVITE messages will be addressed to the s extension. If you want to see something specific then you either set what you want in contact= or at the end of the register= statement.
As I demonstrated earlier even the invite was sent to s we still have the ability to extract the necessary information from the To: field of INVITE.
Dear Andrew as I told you I have asterisk’s GUI and files are used in a different way.
This are my 3 conf files:
ns301086.ovh.net/asterisk_files/sip.conf
ns301086.ovh.net/asterisk_files/extensions.conf
ns301086.ovh.net/asterisk_files/users.conf
can someone please tell me HOW I got to do with this kind of files?
ok I’ve solved the problem this way without the gui:
[trunk_1]
context = DID_trunk_1
contact = 0707777777
…
[trunk_2]
context = DID_trunk_1
contact = 0708888888
[DID_trunk_1]
include = default
exten = _0707777777,1,Goto(default|200|1)
exten = _070888888,1,Goto(default|100|1)
but now I got 2 more problems:
1 - how to make calling rules for each trunk?
for example if I want to call just using trunk 0707777777?
2 - I’m loosing connection to asterisk, my ip phones loose connection after some while. My asterisk is placed in a remote server with firewall and my ipphones too.