12 Mar
2004
12 Mar
'04
8:40 p.m.
I would do it after the enum lookups. You don't have to send to a
certain IP address - just rewrite the request URI e.g.
4071234567(a)yourdoain.com to 1234567(a)fwd.pulver.com and then send the
message as usual (t_relay).
I'm not sure if this will work fine, as the To: header is still the
same, e.g. 4071234567(a)yourdomain.com and if the SIP client of the FWD
user checks the To: header before accpeting the call it won't match.
How is iptel solving the problem of interconnecting
iptel/fwd/sipphone/... Are there problems caused by the To: header?
regards,
Klaus
Doug R wrote:
I'm using the script from
http://lists.iptel.org/pipermail/serusers/2004-February/005996.html
(the first ser.cfg). I'm trying to figure out where to include my SIP outgoing calls
to my Asterisk PSTN server. For example, let's say I want to transfer a SIP call for
a local XXX-XXXX number to pstn.xyz.com port 5050. How would I set that up?
Additionally, I'm rather confused as to which section I would put a dialplan into for
say connecting to FWD users, etc. Again, like before if I wanted to send a certain call
to a certain termination point, let's say 407-xxx-xxxx, I want to send that to one IP,
and 954-xxx-xxxx I want to sent to a different one. Where are those statements supposed
to be?
Thanks a lot...
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