now i am doing this but still loop detected.
can u tell me how to log i am trying to
log(1,"message") but there is no messages in
/var/log/messages
if (search("User-Agent: Asterisk PBX.*")) {
route(5);
break;
} else {
route(4);
break;
}
--- Iqbal <iqbal(a)gigo.co.uk> wrote:
Hi
I know this isn't the best solution but what if you
did
!search("User-Agent: Asterisk PBX.*")
that way you could see the request coming from
asterisk and process
differently. As for why the 0 does not match, once
you sort out the loop
problem you could look at the URI, and setup the
debug
Iqbal
Kamran Ahmad wrote:
Hello
i m getting loop detected
UA-------->SER
SER----->asterisk
Asterisk------(back TO SER after adding 0)----->SER
there is if condition that ll check if 0 is added
then
dont send back to asterisk but it is not checking
that
condition properly and sending request back to
asterisk.
--- Iqbal <iqbal(a)gigo.co.uk> wrote:
>loop detected, when the call gets to asterisk what
>are you telling
>asterisk to do, send it back to ser, or out
>somewhere else.
>Calls are not workinf after the change or before.
>
>iqbal
>
>Kamran Ahmad wrote:
>
>
>
>>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:
>>>
>>>
>>>
>>>
>>>
>>>>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
>>>>
>>>>
=== message truncated ===
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