Dear All
I want user who register with my openser also register with my asterisk for later lookup. I tried t_replicate function from TM module but no luck. I end up with Unauthorized message. Could any help me how to make this work? What do I miss ?
Thank so much
Regards,
Halomoan Zhou
Hi Halomoan,
If openser receives the REGISTER and fwd it to Asterisk, then do not try to do authentication on Asterisk...do it only on openser
regards, bogdan
Halomoan Chow wrote:
Dear All
I want user who register with my openser also register with my asterisk for later lookup. I tried t_replicate function from TM module but no luck. I end up with Unauthorized message. Could any help me how to make this work? What do I miss ?
Thank so much
Regards,
Halomoan Zhou
Users mailing list Users@openser.org http://openser.org/cgi-bin/mailman/listinfo/users
Bogdan-Andrei Iancu wrote:
Hi Halomoan,
If openser receives the REGISTER and fwd it to Asterisk, then do not try to do authentication on Asterisk...do it only on openser
That is correct....proxy authentication is a challenge and response operation. If you defined a secret in a type=peer based context, asterisk would challenge the request (forwarded from OpenSER), which would not succeed. I believe you can limit access using permit/deny to only accept call setups from your own proxies.
On the other hand, my own OpenSER+Asterisk based systems all run with a very very dumb Asterisk configuration - very very minimal configuration. Attempting to keep all of the SIP type=peer's straight and a complex dialplan, especially in terms of scale, is still a big issue for Asterisk. Thus I have almost gone out of my way to ensure I don't ever require very complex (or even regularly changing) Asterisk configurations.
Remember Asterisk is not a SIP Proxy and only generally complies with what is called a B2B-UA. Asterisk very much does its own thing. So, let asterisk do what its good at.... media and application logic and then build the rest of the system accordingly.
Jeremy McNamara
El Thursday 20 September 2007 18:06:50 Jeremy McNamara escribió:
Remember Asterisk is not a SIP Proxy and only generally complies with what is called a B2B-UA.
Specially, note that Asterisk sip_channel doesn't allow sip spiral, this is, call to a SIP proxy which returns the INVITE to Asterisk with different URI. In that case Asterisk rejects it with "Loop detected" (this is a pain if you want to implement parallel forwarding to PSNT and so). Imagine this case:
- Asterisk receives a call from PSTN. - Asterisk calls to a user in OpenSer. - OpenSer has a forwarding for this user that points to PSTN (through Asterisk then). - OpenSer creates a new branch with forwarding URI and sends it to Asterisk. - Asterisk detects it's the same call (maybe by call-id and from-tag) and rejects it because "Loop detected".
Afortunatelly there is a patch for that, but just for a past SVN version of chan_sip. I've tested it and works, but now they ask for more people testing it to confirm it works and upgrade it to the trunk.
This is the reported bug and patch: http://bugs.digium.com/view.php?id=7403
I explain how to get it working:
- ~# cd /usr/src - ~# svn checkout http://svn.digium.com/svn/asterisk/trunk asterisk-svn48358 -r 48358 - Download sip_spiral.patch from http://bugs.digium.com/view.php?id=7403 - Apply it: - ~# mv sip_spiral.patch /usr/src/asterisk-svn48358 - ~# cd /usr/src/asterisk-svn48358 - ~# patch -p0 < sip_spiral3.patch - configure, make, amke install and so.
With this patch Asterisk allows SIP spiral.
It could be great is people here could test and report it in the Asterisk bugtracker in order to developers of chan_sip include it in the trunk.
Regards.
Iñaki Baz Castillo wrote:
It could be great is people here could test and report it in the Asterisk bugtracker in order to developers of chan_sip include it in the trunk.
It is my intention to setup a testing environment in the CodeZode at Astricon 2007 in Phenoix, AZ next week. I will dedicate time to ensuring that patch gets time in the sandbox.
Jeremy McNamara