Bingo!

Thanks for the help. It does work now.

But... There are is a little problem:

My main DID comes in via sip forward at sip:perfectum@pbxes.org, so there is no trunk assigned to it. How can I assign a "default" trunk which does not have to register with any provider? I want to set up inbound routes based on incoming CID and it seems that the trunk name is not optional...

Defining a Trunk is optional, as long as the inbound SIP URI is unique. Defining a Trunk name in the Inbound Route configuration is not optional, since the call might not be routed properly as you experienced.

In your case you should define unique URIs for your DIDs, as well as individual Inbound Routes based on the CID you require. This scenario works properly, albeit there is no trunk name displayed in the Call Monitor.

I'm wondering if there's anything special about accepting SIP URI calls from Voxalot, any incompatibility, etc?

I tried to call working and tested sip uri from ATA registered at Voxalot - it's always busy tone. Nothing appears in the logs. Voxalot guys say there's some known issue with PBXes - any idea?

Can Voxalot be a bit more specific regarding this incompatibility, maybe by providing the URL of a forum posting?

Well, I'm trying to get any more specific information, but so far I was referred to this very thread.

I'm not interested in blaming any party - connectivity problem happen, it's normal; I'm just wondering if you guys know anything that could prevent voxalot-to-pbxes sip uri calls...

RE: Reaching an extension from outside

Hi, sometimes (like now) account-ext@pbxes.org doesn't work. And it's critical for me to do not have downtime. So if i enter account-ext@www3.pbxes.com instead it works when @pbxes.org doesn't work but in case of fail of www3 it'll be another down.

My question is, to reach my external inbound routes what's better?

my account server's hostname (account-ext@ww3.pbxes.com)
my account server's ip (account-ext@88.198.69.237)
general pbxes.org hostname (account-ext@pbxes.org) (now doesn't work)
general pbxes.com hostname (account-ext@pbxes.com)
something else?

Please answer asap cause it's critical.

Thanks

RE: Reaching an extension from outside

Since you mention that call delivery is critical, you should consider setting up another PBXes account on a different data-center. This way, when calls to one server fail to get delivered, they will be delivered via the other.

In order to achieve this, you should inquire whether your ITSP is able to support concurrent registrations, or to send the inbound calls concurrently to 2 distinct URIs.

If they do, no calls should be missed, unless both PBXes servers go down at the same time.

RE: Reaching an extension from outside

Oh ok, thank you.
And in case the server is up to reach my external inbound routes what's better?

my account server's hostname (account-ext@ww3.pbxes.com)
my account server's ip (account-ext@88.198.69.237)
general pbxes.org hostname (account-ext@pbxes.org) (now doesn't work)
general pbxes.com hostname (account-ext@pbxes.com)
something else?

Why yesterday with everything up account-ext@pbxes.org doesn't work?

Thank you.

RE: Reaching an extension from outside

I use account-string@pbxes.org where "string" is not even a number, let alone an extension number. Using extension numbers seems to be the logical choice, but as you discovered doesn't work most of the time.

The string needs to be unique in your account, and it seems to route my calls all the time. The added value of using a string, is that is allows you to name your Inbound Routes in more meaningful ways than extension numbers allow you to.

RE: Reaching an extension from outside

Zitat:

Originally posted by i-p-tel No, your DID provider - gradwell - would have to do the change. It's not possible to deduct the destination account from the INVITE field.

I was trying to have my DIDs through FlowRoute route through a SIP URI using the uri sip:<account>-<identifier>@pbxes.org . That's worked well for me in the past with another DID provider (LocalPhone). But it did not work with FlowRoute. I contacted their support and they provided me with a trace of their "dialog" with PBXes.

The INVITE has the URI I gave them:

Zitat:

INVITE sip:AAAAAAAA-IIIIIIII@pbxes.org SIP/2.0.

(where AAAAAAAA is my account and IIIIIIII is the account identifier I chose)

The To: field has:

Zitat:

To: <sip:+DDDDDDDD@216.115.69.132:5060>.

(where DDDDDDDD is my DID)

PBXes replied with 404 Not Found.

So they too use INVITE rather than TO, and it would be possible to determine the account and account-identifier with the INVITE.

Would PBXes consider checking the INVITE as a back-up if the TO does not correspond to a valid account?

Thank you for your consideration.

P.S. I should note that I can get Flowroute to work using SIP registration. Using SIP URIs would be preferable as I could more easily have multiple DIDs use the same inbound route. With registration, the inbound route has to specify the DID as the Trunk name.

Updated

I asked Flowroute whether they'd be willing to provide an option to use the SIP URI provided for routing in the To: field rather than the INVITE header. This is their reply, for what it's worth.

Zitat:

I apologize, however we would not be able to manipulate the To header, as per SIP RFC, this is generated by the caller. To further clarify, the Request-URI in the INVITE header is what the SIP protocol is designed to use for referencing the remote resource/device. The To header should not be used for any routing decisions by any SIP proxy or User-Agent.

This post has been edited 1 time(s), it was last edited by iva on 03.02.2011 at 01:18.