[Serusers] Re: checking called number

[Iqbal]

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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>