[Serusers] Re: checking called number

[Iqbal]

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
>>>>>          
>>>>>
>>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;
>>>>>>>     }
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>		
>>>>>>>__________________________________ 
>>>>>>>Do you Yahoo!? 
>>>>>>>Read only the mail you want - Yahoo! Mail
>>>>>>>              
>>>>>>>
>=== message truncated ===
>
>
>
>		
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>