[asterisk-dev] chan_sip SIP Authentication

Johansson Olle E oej at edvina.net
Thu Jan 29 02:54:11 CST 2009

29 jan 2009 kl. 08.42 skrev Klaus Darilion:

> Johansson Olle E schrieb:
>>> A Workaround is to put the originally called number in the To  
>>> header -
>>> but this is ugly as To based routing it is against all RFCs.
>> ´
>> ...and not using the contact URI bind to the AOR in the REGISTER as a
>> reqeust-URI
>> on the INVITE also breaks the RFCs...
> true
>> I've found devices who DO NOT accept calls to the URI they register
>> with Asterisk...
>> Which is why you can find both IP and port registred with us in the
>> SIP_PEER function...
> This one I do not understand. If the device does not accept the  
> contact
> the registered then the device is buggy. So what do these devices  
> accept?

Yes, it's a very buggy device. They only accept request-URI's with DIDs.
It was Cisco Call Manager Express...


More information about the asterisk-dev mailing list