[Asterisk-Users] Asterisk Registering with SER question

Ryan Pagquil rpagquil at philonline.com
Wed Feb 1 00:56:58 MST 2006


Hi Olle,
         Nice to know that. In my case I'm simulating a prepaid call 
from Asterisk to SER. On the Asterisk side, there are users 
registered with of course different extensions. Asterisk uses SER as 
the SIP trunk and SER will forward it to the PSTN gateway. Asterisk 
registers to SER with single username "asterisk", and assuming that 
the asterisk-registered-users placed calls simultateously, on my CDR 
database there will be multiple occurence of the asterisk username 
because of multiple calls. Now if total duration of all the calls 
placed by username " asterisk" greater than his credit, I will send a 
BYE to them for them to disconnect. Would it be usefull not setting 
up the extension on my register= parameter in sip.conf? If I used the 
sip:s at X.X.X.X will it distuinguish the asterisk registered user to 
disconnect? I'm using a perl script for me to monitor the calls on 
SER, also sipsak.

Thanks,
Ryan

At 03:28 PM 2/1/06, Olle E Johansson wrote:
>Ryan Pagquil wrote:
>>Hi,
>>         On asterisk console I enabled SIP debugging and I found 
>> out that asterisk is sending this:
>>Reliably Transmitting:
>>REGISTER sip:imydomain.com SIP/2.0
>>Via: SIP/2.0/UDP :x.x.x.x:5060;branch=z9hG4bK69398d1a
>>From: <sip:asterisk at mydomain.com>;tag=as1d1a85bc
>>To: <sip:asterisk at mydomain.com>
>>Call-ID: 640c73336ff4ebf74020e4c42256bdea at 202.84.24.47
>>CSeq: 102 REGISTER
>>User-Agent: Asterisk PBX
>>Expires: 120
>>*Contact: <sip:s at X.X.X.X> <--registered on SER Contact column on 
>>location table
>>*Event: registration
>>Content-Length: 0
>>so it means that Asterisk is sending that information, how can I 
>>correct this? It should be <sip:asterisk at x.x.x.x> no <sip:s at x.x.x.x> .
>Ryan,
>Check the syntax of your register= statement. The last entry is the 
>extension. If you are not entering any extension, asterisk will send 
>"s" as in this case.
>
>You have plenty of examples in sip.conf.sample in the /configs directory
>of your source code.
>
>/O
>_______________________________________________
>--Bandwidth and Colocation provided by Easynews.com --
>
>Asterisk-Users mailing list
>To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users




More information about the asterisk-users mailing list