SIP Integration / 302 Moved Temporarily

[dstrait]

I am working on integrating some 3CX locations into our OmniPCX. At this point, integration (SIP trunk) is very close to completion. Intra-system calling works along with the Caller ID name. Works fine.

However,

When 3CX is calling into the Alcatel side there is an issue with AA and immediate forwards.

When 3CX is calling an Alcatel AA or an extension set to immediate forward, Alcatel kicks back a 302 redirect (To the voicemail system), which 3CX is not acting on. I am currently looking into this as, as best I can tell, Alcatel is being sane and sending the redirect. It's just not being honored by 3CX. Obviously, that is out of the scope of this forum. I am instead hoping there is something that can be changed on the Alcatel side that will prevent the 302 redirection behavior, similar to how when a DID calls in.

I appreciate anyone's thoughts on this.

[frank]

hi
This forum is the place to talk about everything, including 3CX.
We actually use to use 3CX and dumped it to use FreePBX.
The issue is that a redirect in SIP Language is actually a 302, so the OXE is doing good.
3CX is not honoring it. Try their forum. Their support sucks, but you might have someone who can help over there.
If not come back here and we'll try. If not, replace 3CX lol..

[dstrait]

Thank you very much for your friendly response. And I am glad to hear this is also a generic SIP forum, not just for Alcatel.

It's funny you mention FreePBX. I actually installed that to confirm who was at fault, and it indeed is 3CX. For whatever reason, 3CX is ignoring the redirect.

I posted on their forum and left with a very bad taste in my mouth. They essentially said 'not supported' and did a mic drop.

However, there is still an issue. hah. I am thinking it's a limitation of the Alcatel 'generic' SIP integration. That's probably not an entirely accurate statement, but it's how I'm kind of looking at it.

With FreePBX, I am able to honor the 302 redirects. However, Alcatel does not know what to do with it.

Let's say, for example, we have an AA at extension 5000 and our voicemail system lives at extension 4000.

If I call 5000 from an Alcatel extension the call silently redirects to 4000 so the voicemail system can take care of the AA. All is fine. The problem is Alcatel is seeing 3CX calls (after the redirect) as a brand new phone call like I'm just calling the voicemail extension. It doesn't know why I'm there and just asks me to key an extension number. It is as if I just called 4000 in the first place.

Within Alcatel, this is of course fine. The signaling takes place and the voicemail system knows what to do. However, when a 3CX extension calls in that signaling apparently does not take place or is lost.

Is there any way to change this behavior in Alcatel? The same issue arises when an Alcatel extension is on DND or immediate forward to voicemail. If the call instead times out and goes to voicemail, the issue is not present as a 302 is apparently not involved in that flow. Just when it's immediate.

The only thoughts I have are standing up a second trunk acting like a carrier which would make Alcatel see these calls as external/coming from a carrier then handle the routing internally, like a normal external phone call. This wouldn't help with the DND/immediate to voicemail, but I may just need to accept that limitation.

So, for example:

I would have a fake DID, in Alcatel, of 222-225-000, which redirects to the same AA as our actual phone number. Then, in 3CX, I could catch any calls to 5000 and have it instead call Alcatel as 222-225-000. Being this is an external phone number, Alcatel should not advertise 302 and instead just route it internally. The same way any other external inbound call would be handled, in that Alcatel does not tell the carrier to redirect as the carrier has no idea how to redirect to an internal extension.

I know I'm really thinking outside the box here. Maybe good, maybe bad.

[sadim]

Hi,
I don't know the version of OXE you are using, but check if you have the parameter SIP/SIP Ext Gateway/Redirection response support.
If it exists, then change it to False and make new tests

[dstrait]

Our version is: OmniPCX Enterprise R100 n1.291.68.a

I do see 'support Redirection reponse', but it is already set to no. I tried flipping it just for, but no change.

[frank]

You might have to run "killall sipmotor" at the linux command line to restart the sip process.

On the other hand, I'm working on a document to use Asterisk / FreePBX as a voicemail for OXE.
This is on GitHub, so everybody can participate.

[sadim]

and he can also share the oxe configuration used and a sipmotor trace with the test description

[dstrait]

I reached out to the company who helps support our Alcatel and got this back

“support redirection response” setting is only for SIP ISDN which is not how the SIP gateway is setup.

I also have this trace:
[*]3CX Extension 40010 >>> OXE 10000 (Main Auto Attendant) >>> Redirect >>> OXE 19999 (Voicemail System)

Code: Select all

SIP/2.0 302 Moved Temporarily
Contact: <sip:19999@[OXE FQDN]:5060;user=phone>
User-Agent: OmniPCX Enterprise R100.0 n1.291.68.a
To: <sip:10000@[3CX/FreePBX]>;tag=c07b56c13506f78afbf0c55dd533f07
From: "John Doe" <sip:40010@[3CX/FreePBX]>;tag=b65bb564
Call-ID: jZTHyuzlfw_3iMpBHQSm1g..
CSeq: 1 INVITE
Via: SIP/2.0/UDP [OXE IP]:5060;received=[OXE IP];branch=z9hG4bK-524287-1---3838a1329899652f;rport=5060
Content-Length: 0

What other information would be helpful? How would I best provide the configuration? I am very much not familiar with the OXE CLI, so please let me know of any specific commands I should run.

[sadim]

Hi
SIP-ABCF is ONLY for interconnect certified equipments.
3CX is not certified and since it does not support 302 SIP message, it seems far to be, you must interconnect it using SIP-ISDN

[dstrait]

You are definitely accurate in that 3CX does not honor the redirect. However, FreePBX does and there is still the issue of the AA not knowing what to do where it just asks for an extension number.

With that in mind, would you still agree the best route is to tie the systems together via ISDN? Is there any traction to my thought of setting up a SIP trunk that is meant for external lines and using that?

For example, I plan to stand up a Twilio trunk into the Alcatel. When call comes through this Twilio trunk, I can't imagine Alcatel will respond with any sort of redirect because the carrier will not have any knowledge of internal extensions. Can I use that to my benefit? Standing up a SIP trunk between 3CX and Alcatel where I have a fake DID on Alcatel for each AA. Then, on 3CX, I modify the dialing plan to capture and redirect as needed.

So, if 3CX dialed 10000, I would redirect it to say... +1.111.0000 which would reside on the Alcatel system, and be configure to ring the AA.

I have a habit of thinking wild thoughts, so this may be ridiculous.