THANK YOU.
That works.
Stephen
Subject: RE: [Serusers] voicemail question - dial plan example - pstngatewayexample
Steve,
Have you tried using "append_branch()" after "rewritehostport(<new uri>)"?
Jaime
From: Steve Dolloff sdolloff@noc.dls.net on 03/10/2003 15:04
To: Alexander Mayrhofer axelm@nic.at@SMTP@Exchange Jiri Kuthan jiri@iptel.org@SMTP@Exchange serusers@lists.iptel.org@SMTP@Exchange cc:
Subject: RE: [Serusers] voicemail question - dial plan example - pstngatewayexample
Any other suggestions?
Subject: Re: [Serusers] voicemail question - dial plan example - pstngateway example
On (02.10.03 23:28), Alexander Mayrhofer wrote: > replacing > > rewritehostport("219.242.10.153:5061"); > > with > > append_branch("sip:219.242.10.153:5061"); > > in the fialure_route should do the trick.
err, i hate replying to my own mails.
The above line will make you loose the user part of the URI, and is therefore not a working solution ... we'd need a function to reset the branch status value here, which afaik does currently not exist ...
axelm
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What table in the ser database should I insert a row, for the SIP server to forward the calls to a 3 digit extension. I have put this statement in the ser.cfg.
if (uri=="sip:466@iic.com") {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # };
save("location"); break; };
# native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; if (uri=~"^sip:466@iic.com") { forward( x.x.x.x, 5060 ); };
x.x.x.x is the SIP phone's IP.