DomainAdapter or how to manipulate sip domain. 1

[B0nes]

Hello everyone!

A bit about the configuration:
CM 8.x
SM 8.x
SMGR 8.x
AADS
J1xx series phones and IX Workplace
Users with SM, CM and PS profile set on SMGR. Login to IX workplace through login@domain.

The company is planning a domain change project for users. During testing, I moved one user to the new domain and started testing calls.
User 1 - 1234@abc.com
User 2 - 4321@xyz.com

Both users are registered on the same SM, with different registration ports. When User 2 calls User 1, the call goes through successfully (the adaptation odstd=abc.com is applied). However, when calling in the opposite direction, the call results in a 404 Not Found, which is logical since the user 4321 with the domain abc.com does not exist.

In the Session Manager documentation, there is a mention of the DomainAdapter, which, according to the description, seems to perform the required function, i.e., it allows changing the domain based on the dialed number. However, the documentation does not provide any details on how to use this adapter or what parameters it supports.

I also tried using the Conditions - Regular Expression Adaptations method, but when applying it to the SIP Entity, the call gets rejected with a "Request Timeout" regardless of what action I choose, even something as simple as changing the digits in the number causes the same result...

If anyone has documentation or knowledge on how use DomainAdapter or to accomplish this task, I would be extremely grateful.

PS:
Session Manager - Global Settings - Set Precedence for Routing = RegularExpressions.

 

[JefferP]

Just spit balling here. I would think that as long as both domains are listed in Domains under routing would be all you'd need. Calls to/from/between SIP endpoints use SM database routing and not dial patterns.

 

[B0nes]

Just spit balling here. I would think that as long as both domains are listed in Domains under routing would be all you'd need. Calls to/from/between SIP endpoints use SM database routing and not dial patterns.

Both of them already in Domain tab... When user 4321@xyz.com calling 1234@abc.com INVITE sendings with "To:sips:1234@xyz.com" then adaptation "odstd=abc.com" change it to abc.com.
When 1234@abc.com calling 4321 "INVITE To:sips:4321@abc.com." But there are no user with this domain in SM database...

 

[B0nes]

It sounds like you're facing a challenge manipulating SIP domains during your domain change project. There are a couple of approaches you might consider.

The documentation seems to mention a "DomainAdapter" feature, but lacking details on its usage can be frustrating. However, there's also the possibility of using Session Manager's "Conditions - Regular Expression Adaptations" method.

While you mentioned encountering a "Request Timeout" with this approach, it might be worth revisiting the configuration. Double-checking the regular expressions and chosen actions could be key.

If you're still stuck, consider reaching out to a communications consulting firm like Branex. They may have experience with manipulating SIP domains in similar scenarios and offer valuable guidance.

You are absolutely right!
"Request timeout" encounter when I add "@" in "Replace\Add Expression" or in "Match Expression" fields...
All documentation does not mention the need to escape the @ symbol or smth, but there is also no example of an expression replacing the domain.
And now I cant correctly write regular expression for domain changing.

 

[kyle555]

I think your problem is that CM can only be authoritative to one domain. If one SIP user is calling another and they have different domains, the originating or terminating application sequence is going to fail for one of them. I don't have a good answer for you beyond changing all your domains all at the same time.

Maybe you can add a second handle of type Avaya SIP with the old domain and see if that works, but I think you're in some trouble if you need to change the domain across all users and CM/SM globally.

 

[dialtonebone]

I think your problem is that CM can only be authoritative to one domain. If one SIP user is calling another and they have different domains, the originating or terminating application sequence is going to fail for one of them. I don't have a good answer for you beyond changing all your domains all at the same time.

Maybe you can add a second handle of type Avaya SIP with the old domain and see if that works, but I think you're in some trouble if you need to change the domain across all users and CM/SM globally.

Kyle - I am in agreement with you on this. I don't see how you can have the CM taking care of two SIP domains -- only the IPNR SIP Domain assigned to PROCR will work.
A flash cut would be most appropriate if your interpretation is accurate (CM sip domain change).

 

[G van Hamburg]

Agree with Kyle also. TIP: Choose a SIP domain and keep using that. The registrar (SM) can be a ‘real’ domain name but the SIP domain is just the SIP domain, like sip-abc.com.