Reaching an extension from outside

Hi,

I have setted up a sip extension and successfully registered a softphone with this extension. However, when I try to reach it using a sip url:

sip:<my account name>-<this extension>@pbx.i-p-tel.com

i get "user not found". I use the softphone Ekiga, which works fine with my FWD account (when I call to my FWD I get "user is busy"). I also tried to call this number from Gizmo Project without success.

Am I entering the right address?

RE: Reaching an extension from outside

Hi,
thanks for your request.

We have solved the problem. Eventually you have to edit something in your configuration and click on the red APPLY bar to reload the configuration.

However, you might need to specify the IP address instead of the name because we have not set up all name servers correctly yet:

sip:<account name>-<extension no>@217.195.32.11

Best regards,
Pascal

Still doesn't work.
My client is successfully registered with
the user: <my account>-<exension>
and registrar: pbx.i-p-tel.com
I can also see that it active (green) in the status window,
however, when I dial both:
sip:<my account>-<exension>@pbx.i-p-tel.com
sip:<my account>-<exension>@217.195.32.11
I got "User not found"

RE: Reaching an extension from outside

You're right. By default you can't call extensions; it's only allowed to call a whole PBX from outside:

Even if you specify an extension it will call the PBX as such like calling from a trunk. This is not a bug - it's a feature.
The intention is to keep the extensions private.

To work this way you have to choose which extension to send incoming calls to. This can be done under menu item "Incoming Calls".

Additionally it is possible to make particular extensions reachable from outside. You need to add inbound routes therefore. As trunk name enter the public address you want to be reachable at, e.g. <account name>-<extension no> or <account name>-<arbitrary name>.

help me please!

I want to achive a couple of things uing this service and would be willing to pay a reasonable fee to accomplish, however thus far I have had no luck.-


I have registered to the server and established a route to the PSTN , however the system askes me for a password even though I have left the password blank.

Secondly inbounds to not seem to be passing through either.

IOn addition and independent of the other itesms.

Can I register to an external server and do a simple SIP forward to a SIP address such as 613@fwd.pulver.com?

I believe in asterisk terms this is called an external bridge as opposed to a Native bridge. I do not see how I would differentiate between these two in your system.

RE: help me please!

Dear Mark,

are you sure that you have left the password field empty (in your outbound route)?

Inside of the PBX everything works by numbers. They are used to select trunks. After that the call is passed to the selected external server. So you need to enter an outbound route for 613.

Are the inbound calls still a problem?

Best regards,
Pascal

Password field IS empty, although I have tried it with a Password as well, and even with a Password, it did not work

Also,when sending traffic to a SIP address it will likely be a server that I do not have an account on to register, however WILL accept SIP inbounds. I use FWD as an exampe, however FWD is not my interntion. It is probably I would want a few of these.


Can you give me an example of how I use the route to signify a SIP address?? Continuing with the 613 example would mean if I dial 613, it would route to sip:613@fwd.pulver.com (or sip/21488@fwd.pulver.com), however from the examples given, I do not get it.

Also I should clarify again that I DO NOT want a native bridge I want an External bridge (SIP redirect).


EDIT......


After posting above, I have now successfully made an outbound call. It seems deleting the route code several times and/or changing it, it finally worked.

Also, Is there a limitation of defining a single trunk? I have tried but it never seems to add it.

PLEASE help with sending traffic to a SIP address!

This post has been edited 3 time(s), it was last edited by mar on 18.02.2006 at 08:06.

Reaching SIP addresses

There is no limitation to a single trunk. You can have as many as you want.

For reaching SIP addresses on external servers not registered to I have added an option. It's available under the Device Options of a SIP extension and called "dial". Try putting your SIP address there, e.g. sip/21488@fwd.pulver.com. Since this is an advanced option it only appears when editing an existing extension.

Best regards,
Pascal

Working

See thread in "Bugs" section. Fixing my problem with the SPA-3000 has also fixed this problem... you need to enter a SIP Server of 217.195.32.11 on the trunk. Trunk has to be named <userid>-something then you can SIP Dial into SIP/<userid>-something@pbx.i-p-tel.com

Thanks to Pascal for identifying the solution.

DAK

RE: Reaching an extension from outside

OK, it's getting easier. You don't have to setup a trunk for incoming SIP calls anymore.

Only if you want to change the default settings for language, dtmfmode, audio bypass etc. you can optionally create one with SIP server 217.195.32.11.

Best regards,
Pascal

I set up an incoming route with trunk name "seattle-in" (where "seattle" is my userid). I tried to call the SIP URI seattle-in@www2.pbxes.com and it does not work.

This post has been edited 1 time(s), it was last edited by sea on 06.10.2006 at 07:04.

seattle,

Have you tried to create an inbound route with seattle-extension# and see if that works?

I am trying to assign an additional DID number to my pbxes account.

I have tried directing my DID number to:

sip:gadgets@www5.pbxes.com
sip:gadgets@88.191.27.122
sip:gadgets@pbxes.org

and direct extensions:

sip:gadgets-100@www5.pbxes.com
sip:gadgets-100@88.191.27.122
sip:gadgets-100@pbxes.org

and also created an inbound route called gadgets-new and tried assigning the DID number to:

sip:gadgets-new@www5.pbxes.com
sip:gadgets-new@88.191.27.122
sip:gadgets-new@pbxes.org

None of the calls get through, and my DID provider (Gradwell) says that they pbxes responds "not found"..

I asked this elsewhere and was promptly directed to this thread, but despite reading it and other threads through many times, I can't see any clarification of what will actually work !

Any suggestions would be very much appreciated,

Thanks in advance,

jules.

To update this, and hopefully help others, all the above seem perfectly valid destinations when calling from a fwd.pulver account..

gadgets@ and gadgets-100@ both go to the default route, and as gadgets-new had been setup as a seperate incoming route it was routed as defined there.

They still dont work when accessing from my gradwell account, I have asked gradwell to confirm but perhaps someone at i-p-tel could clarify some potential reasons it may not be working?

thanks and regards,

jules.

RE: Reaching an extension from outside

The problem here is that your provider - gradwell - sends the call to <sip:09904yourDID@gk.magrathea.net> instead of <sip:gadgets@pbxes.org>.

This post has been edited 1 time(s), it was last edited by i-p on 26.07.2007 at 22:49.

RE: Reaching an extension from outside

Zitat:

Originally posted by i-p-tel The problem here is that your provider - gradwell - sends the call to <sip:09904yourDID@gk.magrathea.net> instead of <sip:gadgets@pbxes.org>.

I have had some further clairification on this.

According to RFC3261

Zitat:

8.2.2.1 To and Request-URI The To header field identifies the original recipient of the request designated by the user identified in the From field. The original recipient may or may not be the UAS processing the request, due to call forwarding or other proxy operations.

So, for various reasons, the "To" field may well not be the ultimate destination (ie may not show the pbxes.com account)

Zitat:

it is RECOMMENDED that a UAS accept requests even if they do not recognize the URI scheme (for example, a tel: URI) in the To header field, or if the To header field does not address a known or current user of this UAS.

In other words, that you should determine the destination account from the INVITE field, and not the TO field.

This would then solve the problem for us (and I suspect for many other users who want to use third party DIDs!)

Thanks in advance,

jules.

RE: Reaching an extension from outside

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

RE: Reaching an extension from outside

Zitat:

Originally posted by i-p-tel It's not possible to deduct the destination account from the INVITE field.

Why not, when that appears to be the SIP standard?

jules.

I want to take direct sip-in calls to some of my extensions. I have set up incoming route to ext 201, but it still does not work.

sip:perfectum@pbxes.org is fine, sip:perfectum-201@pbxes.org is dead...

What you describe is possible, and your configuration seems to be correct, but there are 3 other Inbound Routes which are not properly configured and might be interfering with the perfectum-201 Inbound Route.

These Inbound Routes do not have any Trunk specified, just the CallerID number which you are expecting calls from. Define the Trunk in them and try calling again via the perfectum-201@pbxes.org SIP URI.