3300/6510 Call Forwarding Issue over SIP

[SeanKirby]

Hello, this is my first post in this forum so please excuse and lapses in etiquette.

I am running a Mitel 3300 MXc v. 8.0.9.20 which passes calls to an Asterisk box which provides NATing, and from there to my SIP provider (also using asterisk)

I am experiencing a problem with inbound calls, when the call is transferred back out to the PSTN via either embedded voicemail or the 6510 voice messaging server.

The two conditions I have seen so far are as follows:

1. Call comes into 3300 via SIP trunk. Caller reaches auto-attendant mailbox and is prompted to dial 0 to reach a cell phone. 0 (operator extension) goes to ext. 895 which is a setup in speed call assignment to dial the cell phone.

Once caller hits 0, they hear on-hold music from the Asterisk SIP peer. Call rings on cell phone, but cell phone user hears same hold music. The two calls never connect.

2. Similar setup as above, but auto-attendant is configured on the 6510 messaging server, as prompted caller dials 1 to be forwarded to cell phone. Cell phone rings but both parties are left on hold, cannot hear each other.

Any thoughts on this issue would be deeply appreciated.

 

[kwbMitel]

I'm not very familiar with SIP but something in your scenario is strange.

The MOH source is from the SIP Peer?

This would appear to indicate that the call is being transfered/Conferenced via the Asterisk Box and not being conferenced thru the 3300.

To test for this I would try calling internally set to set and see if the symptoms are the same or if the call succeeds. (eliminate the inbound leg from SIP)

I might also just try a call directly to the speedcall and verify that it works as advertised.

BTW- your details are great.



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Occam's Razor - All things being equal, the simplest solution is the right one.

 

[SeanKirby]

That is correct, the MOH is coming from the SIP peer. Strange indeed, you are correct that it indicates the transfer/conference is happening on the asterisk box.

I have confirmed the same, by testing via internal extensions. The calls connect perfectly. The speed call is also setup properly. The issue only occurs when the call is coming in from SIP.

Perhaps someone around here who knows SIP/Asterisk might have some tips for how I can prevent the SIP peer from doing the transferring, and keep this function on the PBX? I think if I could do that it would eliminate the problem. It seems like right now the SIP peer is creating two different calls in and out, but has no idea how to connect them.

Thank you for your reply, kwb.

 

[fissure]

in Asterisk, I think you control whether it creates separate legs for the calls, with 'canreinvite', see:

http://www.voip-info.org/wiki/view/Asterisk+Letting+SIP+clients+connect+directly

we'd be interested, in making the Mitel work in 'full media proxy' mode.
At the moment, we think, if a Mitel user calls out through the Mitel, on a SIP trunk, the user's Mitel phone ends up directly speaking SIP to the external SIP trunk provider (which means, the IP subnet the phones are in, need to be able to get out to the Internet, not just the Mitel 3300)

you might also want to try something with:
Renegotiate SDP To Enforce Symmetric Codec
under: SIP Peer Profile
default is no

 

[kwbMitel]

I still can't help with the specifics of the SIP/Asterisk issue bit I can offer a couple of suggestions to those that can.

The Trunk appears to be operating in one of 2 ways.

1) Similar to B-Channel Transfer on ISDN lines where a transfer command is sent to the CO (asterisk in this case) initiating the transfer at the CO level. This frees up the channels from the 3300 to the CO.

2) Similar to Route Optimization on the 3300 where a call that originates in 3300 system A, goes to 3300 System B, and then returns to (or thru) system A creating a loop. The 3300 system A can be configured to detect the loop and remove it and then routing the call directly internally. The asterisk box would be system A in this example.

*******************************************************
Occam's Razor - All things being equal, the simplest solution is the right one.

 

[Canuckvoip]

Maybe I didn't read this right but let's roll back a bit.
Forget the Asterisk for now...
Check the cos of the sip trunks/esna.
Can they or can they not connect two sip trunks together?
Can a set user do that?

Dave

You can't believe anything you read... unless of course it's this.