19 Aug
2005
19 Aug
'05
2:21 p.m.
i m only sending invite to asterisk one when i try
bindaddr=0.0.0.0
calls are not working. now i changed sip.conf
bindaddr=0.0.0.0
and in ser.cfg
port=5060
with these changes now register messages are stoped
but still getting 482 "Loop Detected".
--- Iqbal <iqbal(a)gigo.co.uk> wrote:
Hi
The register messages what username are they for,
and from what IP
address, do a sip debug in asterisk for this.
Also why are you sending register messages to
asterisk, just send your
INVITES there.
As for debug run ngrep, to pick up your messages ,
and see what is going
on, and to look at the uri problem, remove 2 out of
3 of the routes, and
see what happens, if call fails then matching is not
correctly being
done, or the uri is not correct.
iqbal
Kamran Ahmad wrote:
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i tried to do debug (put messages in
/tmp/call.log)
but there is no Invite having only one zero(like
06999786) all call were 0097937223 or 46364736.
this
means that only third else part is always active
but
if third part is active then there must be some
invite
starting with only one zero. it means second time
invite call is not comming here .
My from main invites are only comming here
route(3). i
think all messages are going to asterisk because
there
is only one statement in ser.cfg having port 5970
(this is for asterisk) and all my register messages
are also going there to asterisk.
there are too many register messages on my
asterisk. i
dont know why they are comming to asterisk as
this
port is not available for any user nobody uses this
port only ser is routing calls to asterisk
route {
if (uri==myself) {
if (method=="INVITE") {
route(3);
break;
}
}
}
route[3] {
exec_msg("cat >> /tmp/call.log");
if(uri=="^sip:00[1-9].*@.*") {
route(4);
break;
} else if (uri=="^sip:0[1-9].*@.*") {
strip(1);
route(5);
break;
} else {
route(4);
break;
}
}
--- Iqbal <iqbal(a)gigo.co.uk> wrote:
>dump a call trace showing your uri
>
>Iqbal
>
>Kamran Ahmad wrote:
>
>
>
>>hello
>>
>>i m using ser-0.9.0 on 5060 and asterisk-1.0.9 on
>>
>>
>5970
>
>
>>but problem is that i cannot send my ser back to
>>asterisk.
>>the problem is that it always goto route(4)
second
else if is not properly checked how to check this
condition. i m adding 0 when asterisk is sending
call
back.
if(uri=="^sip:00[1-9]+@.*") {
route(4);
break;
} else if (uri=="^sip:0[1-9]+@.*") {
strip(1);
route(5);
break;
} else {
route(4);
break;
}
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