M. Ranganathan wrote:
On Wed, Feb 25, 2009 at 5:59 AM, Chaitra <[email protected]> wrote:
Hi All,
This is regarding a Multi SIP Trunk Gateway scenario with the following
configuration:
We have 2 Bandwidth.com directory numbers 9497776945 and 9497776946.
Setup:
------
1. Create a SIP trunk (1) , with the 1st directory number 9497776945
specified as the Caller ID on the 'Caller ID' screen
2. Create a SIP trunk (2) , with the 2nd directory number 9497776946
specified as the Caller ID on the 'Caller ID' screen
3. Create a custom dial plan to dial out through SIP trunk (1) as --->
Prefix 00 and 'Any number of digits' , Resulting call dial, + and append
'matched suffix' , Assign SIP trunk (1) as gateway
4. Create a custom dial plan to dial out through SIP trunk (2) as --->
Prefix 500 and 'Any number of digits' , Resulting call dial, + and append
'matched suffix' , Assign SIP trunk (2) as gateway
Now from an SCS user,
Case1: dial 500 followed by 919886216883-----> the caller id displayed on my
PSTN phone is 9497776946
Case2: dial 00 followed by 919886216883----->the caller id displayed on my
PSTN phone is again 9497776946 (this should have been 9497776945)
Hi Chaitra,
Do you have the Caller-ID set up correctly? If so, it should pick up
the right account. For bandwidth.com the caller Id should be
usern...@public-address. Each trunk gateway should of course have a
different caller-id. If you have configured correctly, the intent is
that the right account will be used. If not, there is a bug. Please
file an issue and attach a log with DEBUG enabled ( and also turn on
fine logging in log4j.properties ). I can look at the logs then and
get a better idea if this is a problem.
Thanks
Ranga
Although the Caller-ID has been specified for each of the SIP trunks as
9497776...@public address for SIP trunk (1) and
9497776...@public-address for SIP trunk (2) and both these SIP trunks
have different dial plans, the wrong Caller-ID is displayed at far end.
(called number)
Reported http://track.sipfoundry.org/browse/XECS-2283 .
Thanks,
Chaitra
That is, the Caller id of the latest added SIP trunk is displayed on the
PSTN phone irrespective of the dial plan used to dial the PSTN number. In
XECS-1685 (Multi Sip Trunk Gateway support is needed from same
ISP/provider) the closing comments say "this seems to be intent: SipXbridge
simply uses the first account that matches domain if it cannot find one with
specific clid "
Also if this is the case , wont the wrong account be billed each time a call
is made with the setup mentioned above? Please correct me if I am wrong.
In Case2, also checked the logs to verify the account through which the call
was placed. This says,
Original-From: \"301\" <sip:[email protected]>;tag=B5E00800-AD8104BF
Aliased-From: <sip:[email protected]>;tag=B5E00800-AD8104BF
Thanks in advance,
Chaitra
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