[Serusers] Re: checking called number

[Iqbal]

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 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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>>>>>http://lists.iptel.org/mailman/listinfo/serusers
>>>>>
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>>>>>
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>>>>>
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>
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