[Serusers] Re: checking called number
Kamran Ahmad
p_kami at yahoo.com
Fri Aug 19 20:48:45 CEST 2005
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 at 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 at 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 at 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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