Bogdan-Andrei Iancu wrote:

Hi Barry,

I've seen some time ago a similar post - a suggested solution was to enable multidomain support into Asterisk in order to avoid user overlapping.

Apparently it is not possible yet in Asterisk. Answer from the Asterisk list:

For authentication, we only look at

1. Whether the domain is a domain we host
2. If the user (regardless of domain) is a user we know

The user namespace is common for all hosted domains, so you can't have barry@domain1 *and* barry@domain2.

So, the only way around it that I can see is to somehow have OpenSER change the username to username_domain so that each will be unique.

It looks like uac_from_replace should handle this. I have tried it, and I can see that Asterisk does in fact get user_domain@domain in the first invite, but thereafter for some reason OpenSER changes it to just _@domain for subsequent requests.

Regards,

-Barry

regards, bogdan

Barry Flanagan wrote:

...

Hi,

I am trying to get a multi-domain setup working between OpenSER and Asterisk (PSTN gateway and Voicemail).

It appears that I need to munge the username before it goes to Asterisk so that it includes the domain element, otherwise there is clashing in the username namespace on Asterisk.

It looks like I should be able to use uac_replace_from to change the From URI from user@domain.com to user_domain@domain.com so that then I can set up the users in Asterisk as [user_domain]

Does this sound sane, and could someone tell me how I can accomplish this in OpenSER? I have tried a few things but they all end up as bad headers.

Thanks in advance.

On 01/31/06 23:18, Bogdan-Andrei Iancu wrote:

Hi Medve,

I would say the BYE is badly formatted:

U 2006/01/30 23:28:05.692840 NATBOX:5555 -> OPENSER:6060 BYE sip:imedve3@XLITE SIP/2.0. Route: sip:7741@OPENSER:6060;nat=yes;ftag=1123414629;lr=on. Via: SIP/2.0/UDP 192.168.1.101:5555;branch=z9hG4bKce5c531019c8eec7. From: sip:7741@sip2;tag=1700830800. To: imedve3 sip:imedve3@sip2;tag=1123414629. Call-ID: A412855F-F4E3-470E-AC30-F047AE517CA1@XLITE. CSeq: 1 BYE. User-Agent: Cisco ATA 188 v3.2.0 atasip (041111A). Content-Length: 0.

the FROM hdr is broken - the uri needs to be enclosed between brackets since the tag is a hdr param and not an URI param. The proper format would be:

From: <sip:7741@sip2>;tag=1700830800.

According to RFC, the From header is correctly formated, if the uri in the header is not enclosed in brackets, then the parameters are considered to be header parameters.

In my opinion the problem is that the Xlite puts a different From header value in the 200OK. In the BYE, the from is:

From: sip:7741@sip2;tag=1700830800.

and in 200OK is:

From: sip:7741@sip2;tag=1700830800;tag=1700830800.

I guess that the cisco ata does string comparison of header bodies to match the dialog, instead of the tag value.

Cheers, Daniel

regards, bogdan

Medve Istvan wrote:

...

Hi List,

I have similar problem what you can read at http://mail.iptel.org/pipermail/serusers/2004-June/008798.html

In my test lab I use openser 1.0.0 with mediaproxy. UAC A is an X-Lite with public IP. UAC B is a Cisco ATA 188 behind a natbox. Usernames are not numericals but I use alias table to resolve username based on CLI. Everything works fine expect one situation.

When X-Lite calls Cisco ATA and call terminated by calee side with BYE, Cisco ATA does not recognize X-lite's 200 OK response and resends BYE message some times.

ATA 188 reports the folowing to syslog:

Jan 30 22:33:29 a 00:02:23 192.168.1.101 [08]:Tags do not match <134>00:02:23 192.168.1.101 [08]:Failed to extract UID from RxMsg

If I replace ATA188 with an ATCOM phone, this sutiation works well.

So the problem is probably with Cisco ATA 188 but I am not sure that is it a real problem or Cisco ATA more sensitive for protocol violation than others or there is a missconfiguratin in nat traversal.

You can find SIP message capture at http://fuhur.euroweb.hu/ata.txt

If you have any ideal please let me know.

Regards, imedve

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