Topic: [Announce] Selintra-sail-2.1.13-256

[del]

Hi Selintra,

Email sent, thanks a lot.

Del

[jonroberts]

Guys - I've been following this thread with interest & after reading up on the docs on the Selintra site, decided to get this going for myself.

I've hit a small problem - not sure if you can help.  Its a clean install on SME7 with the latest RPMS etc.  All seems OK.  For testing, I've downloaded SJPhone software & am trying to call between 2 PCs.

I can make the *56* call & get the 'You are extension 5000' response, but when I try to hang up the call it seems to take a long time & then I get the error "Critical Transaction Failed: Client Non-Invite transaction [Trying]: timed out" followed by a 'Network Error'.  

Same problem when calling between 2 PCs - I can't seem to hang up.  Any ideas where I should be looking to sort this.

Ta ... Jon

[jonroberts]

Ignore that last post - I found the problem   (... just me being dopey).

I had changed the IP address range by editing the sip.conf file manually & this was being reset by the other changes made via server-manager.

I changed it using the headers panel & all OK now.

[del]

Hi All,

Is there anyone in the US using sail for incoming calls on SIP (or IAX)? If so could you please post their website here. Thanks

Regards,
Del  

[jonroberts]

I've got my SAIL server up & working & all seems fine (.. pretty damned fine, actually   ).  Just one small question:

I've registered with Gradwell & have set up a trunk to recieve incoming.  I can also set up outgoing, but only by putting the following in a custom app:
exten => _0.,1,SetCallerID(My_Number)
exten => _0.,2,Dial(SIP/${EXTEN}@sip.gradwell.net)

Where My_Number is the CLI number I've registered with Gradwell.

I'm guessing its the SetCallerID that's failing when I try to do this through SAIL using a route & trunk.  Is that right?

Its not a big problem - it would just be nice to do it all through the SAIL options if poss.

Cheers
Jon

[SARK devs]

I'm guessing its the SetCallerID that's failing when I try to do this through SAIL using a route & trunk. Is that right?

Hi Jon,

Yes, you are right.  The Gradwell IAX stack has been set up ass backwards (at least, they don't argue when you challenge it!   ).   Essentially, what happens is this; they deliver calls against the account number (user name) rather than your geographic DiD phone number (which is just plain dumb, but quite a few carriers do it this way).  

This means you can't take inbound calls if you nominate your DiD number in the SIP/DiD field when you set up the trunk.  However, if you DO use the account number in the SIP/DiD field then you can't make outbound calls because the Gradwell stack objects if it doesn't get the true DiD number in the CLID (SAIL loads the SIP/DiD number into the CLID).  So you're stuffed both ways.  Fortunately there's a cute little trick you can do to get 'round this.  

Set up your trunk using the geographic DiD in the SIP/DiD field.  Now you can make outbound but you can't take inbound.  Next, at the end of the IAX Header in the SAIL headers panel, add the following tuples....

Code: [Select]

[gradwellusername]
type=user
context=mainmenu
secret=gradwellpassword
disallow=all
allow=codec
allow=codec
Save them away and restart asterisk (you shouldn't have to but belt and braces, as they say)

Cracked it!

Best

Selintra

[del]

No one in the USA is using this  

Del

[jonroberts]

Save them away and restart asterisk (you shouldn't have to but belt and braces, as they say)

Cracked it!

Cracked it, indeed !!!    - Thanks.  

Out of interest, I've got this test system up which is a couple of PCs (just using SJPhone software & headsets), dedicated SAIL server, Reasonable business Internet connection and Gradwell as a carrier.  All working OK, but voice quality not great (not bad, but can clearly tell its not a normal telephone connection).  Remote user on normal telephone gets a little echo back and calls distort a little.

Its this likely to improve with dedicated IP phones, or do I really need a serious infrastructure upgrade to make this comparible with a standard telephone system?

[del]

Hi Jon

I had the same problem with sjphone, I tried X-Lite and the sound quality is near perfect, my friends in the UK don't believe that I am using VoIP and not even paying for the call

Regards,
Del

[jonroberts]

Del - Thanks for the tip.  

The x-lite phone has sorted the problem for me too.

Jon

[jonroberts]

I'm trying to get to grips with IVR menus, but think I may be missing something.  How do you activate an IVR menu you've defined?

I have used the Trunks option & automation to change the IVR Greeting, but this doesn't seem to run the menu (just plays the greeting).  I've read that I can create a custom app (e.g. exten => s,1,agi(selintra,IVR,TestIVR) ) & then use the options in trunks menu to call the custom app - but is that the best way?

Thanks
Jon

[ntblade]

Hi all this is my first go at this so please be gentle,

Clean install of SME 7.0 Server Only mode on Dell PE SC420 / 512M RAM with:
selintra-sail-2.1.13-278
smeserver-asterisk-1.2.10-1
smeserver-asterisk-sounds-1.2.2-2
smeserver-asterisk-zappri-MPP-1.2.6-1

smeserver-phpmyadmin-multiuser-2.1-1

My problem is that I'm tring to test two extensions using X-Lite.  I can only register one softphone with the etension 5000.  No other will work and I'm getting this message from the Asterisk CLI:

Connected to Asterisk 1.2.10 currently running on test (pid = 3265)
Aug 13 14:24:06 NOTICE[3372]: chan_sip.c:11045 handle_request_register: Registration from '5002 <sip:5002@192.168.1.2>' failed for '192.168.1.3' - Username/auth name mismatch

For each extension I'm adding, I'm using the extension number as the username / pass Here's the text from the server-manager for  two extensions:

type=friend
username=5000
secret=5000
host=dynamic
qualify=3000
context=internal
callerid="5000" <5000>
canreinvite=no
mailbox=5000
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw

type=friend
username=5001
secret=5001
host=dynamic
qualify=3000
context=internal
callerid="5001" <5001>
canreinvite=no
mailbox=5001
pickupgroup=1
callgroup=1
disallow=all
allow=alaw
allow=ulaw

Can anyone help please?
Does each extension have to listen on a different port?
Also, how can I check that my X100P has been found and configured correctly?

Many thanks
N

[jonroberts]

I'm getting this message from the Asterisk CLI:

Connected to Asterisk 1.2.10 currently running on test (pid = 3265)
Aug 13 14:24:06 NOTICE[3372]: chan_sip.c:11045 handle_request_register: Registration from '5002 <sip:5002@192.168.1.2>' failed for '192.168.1.3' - Username/auth name mismatch

I think you may need to check your settings on SJPhone.  It looks like you've initialised the profile with a username of 5002, but your post only shows two extensions created (5000 & 5001).  Try to initialise your SJPhone profile with Account & Password of 5001.

By the way, I found that the sound quality with SJPhone wasn't great.  If you get the same problem, you may want to try the xlite phone.  It sorted the problem for me.

[ntblade]

Thanks for the quick reply.

I have set up three extension, 5000, 5001, 5002.  Only 5000 works and I am using xlite.

Even worse I've just done a server update from the server-manager and now asterisk won't even start throwing the error in /var/log/messages:

test asterisk: Waited too long for udev, - aborting Asterisk  startup.

and:

asterisk: FATAL: Module zaptel not found.

Not happy

[jonroberts]

I have set up three extension, 5000, 5001, 5002.  Only 5000 works and I am using xlite.(

Sorry - not sure why I thought you were using SJPhone.  Also not sure what's causing your current problem.  If your running a demo server, I'd suggest a re-install & start again.

I was pretty much able to get mine working 'out of the box'.  Just created a couple of extensions in server-manager & I also had to change the network setting in sip.conf (using the Headers panel in server-manager) to match my local network, but I think it defaults to 192.168.1.x so I don't think that's your problem.

Sorry I can't be more help