[Serusers] Re: checking called number

[Kamran Ahmad]

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 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 at 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]+ at .*") {
> >>>               route(4);
> >>>               break;
> >>>       } else if (uri=="^sip:0[1-9]+ at .*") {
> >>>               strip(1);
> >>>               route(5);
> >>>               break;
> >>>       } else {
> >>>               route(4);
> >>>               break;
> >>>       }
> >>>
> >>>
> >>>
> >>>		
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> >>>
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> >>>
> >
> >
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> 



		
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