Topic: First Time Setup Gremlins - EXT 2 EXT calls

[the_owl]

Hi All,

I am currently undertaking my first SMESERVER + SAIL/Asterisk Installation.  I have been working in the home/SOHO computer field for a while now but have not used this system before - So far I can only praise the contributors of this project for an excellent job.

I have been asked to install a server at a charity's Head office for the usual file sharing etc - while there i saw a need for an internal phone system and experimented with Sail/Asterisk and SJ Softphones - all worked excellently!!

They then asked about linking in the local offices ( at most 1 pc and 2 staff per office.) As I had already set up a VPN for me to link in and work remotely the idea occured i could do this to all offices and they would just login and use the file sharing etc - but more to the point as part of the network they could use the softphone system - that's where I hit Problems!!!

On the mail LAN perfect, at the remote site outgoing sound was acceptable but no incoming sound during voice call and breaking up during system messages - tries pointing softphon directly at H/O IP without the VPN and got perfect system voice messages, but no incoming voice (HO -> remote client)  i could be heard the other way though.. also I can't login to the voicemail system.

In "extentions" the branch phone is set to "remote" and they can ring any of the main LAN Extentions with the same result.

As a test I tried Removing any Hardware/Software firewalls to try and isolate the problem - no effect - tried putting PC's in the DMZ - no effect.

If possible I am trying to avoid having to learn about external trunks at this stage, all other vpn functions work fine (email, server-manager, group office, etc) and the remote extention is recieving a valid server IP address.

I have enabled "Multilink"  and disabled use remote router in the vpn settings.

Could one of you fine people please point me in the right direction?

Many thanks in advance

[Franco]

If you have these systems in server-gateway mode, all you have to do is create a new SailtoSail trunk in each server.
If in server-only mode, then read here http://www.selintra.com/docs/cgi-bin/view/Main/DocChapter051#Server_only_Mode
This is very easy, create a route or use the 2-digit assignment to reach the extension on the other server.

[the_owl]

Thank you for your amazingly quick response to this post!!

I realize now I was not clear in my initial post (or I misunderstand your reply - more likely!)

At the remote sites there will be no server, just the existing PC connected to the internet running windows (mix of XP and Vista.)  I would like to connect them using the SJ Softphone alongside or through the VPN I have established for access to the groupware, internal email etc.

I can connect the softphone and register it with the server at the Head office, but through the VPN outbount calls from the branch result in clear outbound voice but no inbound voice and choppy system generated voice messages (*55* etc) as i cannot make out what is being said, I am unable to confirm if i can access the voicemail system.

If i configure SJ phone to connect directly to the HO server's ip address i have clear system voice messages, and clear voice transmition from the head office to the client, but nothing the other way around.
Also if i enter the voicemail system (which I can now hear clearly) and enter *50* and enter the password it gets refused. The same with *51* for other extention numbers.

Would you use the sail2sail in this instance? I admit to being on a steep learning curve here - so please forgive any confusion on my part!!

PS the server at HO is in server/gateway mode if that helps...

[Franco]

OK, got it!
-Did you set these phones as 'remote' on the extensions panel?
-What version of Sail are you using? Newer version allows to modify the password as you create the extension, older versions uses the same as the extension for the passwords (voicemail).
-Do you have to use SJ as the client? If not, then you can go with IAX as a client (iaxlite). It'll use a single port 4569 UDP and make everything better if you're using NAT.

Thanks,

[SARK devs]

Hello owl,

Usually, if you are getting one way sound (in your case outbound from the server but not inbound) it is because you are unintentionally spoofing the far end.  Check that you have the correct external IP address set in GLOBALs panel.

Also, in our experience, VoIP over VPN doesn't work too well or, it needs a fair bit of setup.  You are usually better to let the VoIP packets find their way over the public network.

Kind Regards

S

[the_owl]

Thank you once again for responding so quickly!

in answer to your questions..

Yes, I had set the extention as remote when trying it without the VPN and local when trying it with.

I wondered when that "external ip" in globals came into play, but how would i configure it? apart from entering the panel and filling the box!!

At the HO the path looks a bit like this..

WWW -> wired router -> SME/SAIL -> switch and wifi AP -> LAN

We have had to use what we could find as much as poss due to next to no budget - the server was the only pre-approved new purchase.

We have the ADSL coming in with a dynamic IP, calling around their current and other ISP's for a fixed IP account turned out to be beyond their budget (although I think they are tied into a contract with the current one but that's by the by)

I have got around this using the DYNDNS which is handled by the router and passes a fixed local IP tot the WAN interface of the SME.

I would Guess I put the static lan IP in the globals IP - am I correct in that assumption?


Next - I used SJ Softphone because it was the first supported one I came across i could find a download for - no other reason.  I can move over to IAXlite with no problems

I realy appreciate your support in this setup, as a comparison another client of mine has been waiting nearly 2 weeks for an answer from the "Support Department" of a "major" software house that actualy contains usefull info, instead of "we aknowledge your reply........" so thank you again.

I will try these corrections to this setup and report back

[SARK devs]

Somewhat difficult if your server is behind a dynamic ip.  The external-ip should be set to the true external ip of your system (i.e. it's internet ip as opposed to its localnet ip).  It is only necessary to set the external-ip if your SME box is running server-only (behind a NAT).  Otherwise, asterisk will figure out for itself what the IP address is.  The external IP value is placed into each SIP packet so it's important you get it right.  Full description can be found here...

http://www.voip-info.org/wiki/index.php?page=Asterisk+SIP+externip

Extensions outside of the local LAN MUST be defined as remote unless they are coming over the VPN.  Also you must ensure that you have all SIP and RTP ports forwarded to the asterisk box.   The easiest way to guarantee that this is happening is to put the asterisk box into the DMZ and forward all ports to it. 

Best

S
 

[the_owl]

Thanks All!!

As promised, I am writing with an update on how this advice, so promptly given, has panned out.

Fingers crossed - things seem to have worked!  - I tried the IAX solution as advised and that seems to have done the trick.. I have replaced the internet facing router they were using with the original one provided by their ISP, but put in a drawer and forgotten about! - this allowed me to present the external interface with an Internet IP address with DYNDNS being taken care of by SME.

I settled on the solution of having the data features over the vpn and letting the softphones find their own way over the net as this has provided better results in this case.

It's been a very interesting project and we are in the process of replacing our aging server at the workshop with a SME/SAIL/Asterisk box as a direct result of being so impressed with the quality of both the software and support.

Keep up the great work all, have one more question but think that's the subject of another post!