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

[SARK devs]

Hi Del

Here's another mask we dreamed up.  This one allows you to dial local without any prefix and national with the standard US NDD of  "1+".

:001407 0014071:001

Using an example;   691+8451 will be transformed to 001407+691+8451.  Likewise 1+513+662+2300 will be transformed in two stages - first to 001407+1+ 513+662+2300 and then to 001+513+662+2300.

The mask will fail if you have any local numbers which begin with 1, for example 192+5678, but you can use any prefix you like its just that the NDD "1" appealed.

You have to be very disturbed to play with masks... "Nurse!  Where's my medication?"

 

Best

Selintra

[del]

Hi Selintra,

Thanks for the new mask I will try it later tonight and let you know how I get on with it. Any chance you can help me with the incoming numbers? I have looked through the manual but I seem to be missing something somewhere Any help would be appreciated.

Regards,
Del

[del]

Hi All,

Could my incoming calls be a firewall problem? I have an SME server in server/gateway mode connected to a cable modem with my "test" sail/asterisk server behind it. Can anyone tell me if I need to forward any ports to allow incoming calls?

Selintra,
I tried both of those masks and I found that I couldn't call the UK (0044) with the second one, never tried the UK with the first one.  I will keep on trying/testing though.

Regards,
Del

[SARK devs]

Hello Del,

Yes you need to forward ports to your server.  Which ones depends upon whether your ITSP is running Session Border Control (SBC) or not.  Most of the better ones do but many don't.  As a first step, you must forward port 5060 (UDP) to your server.  If this results in one-way sound when you test inbound calls then you should also port-range forward 10000:20000 (UDP).  Furthermore, SME 7.0 runs with the firewall up even when it's in server-only mode.  You can turn the firewall off (which is usually what we do with our test servers) by doing the following at the (server-only) console

config setprop masq status disabled
reboot

To turn the firewall back on do

config setprop masq status enabled
reboot

----

I couldn't call the UK (0044)

Er.... No.  The mask is for US local and national calls.  Be fair mate, you never said you wanted to dial international on this carrier as well .  We'll give you another mask when one of the lads figures one out for you.

Best

Selintra


Selintra

[del]

Hi Selintra,

I have already turned off the firewall on the server only pc, I will forward the ports when I get home from work. Thanks for the help.

Er.... No. The mask is for US local and national calls. Be fair mate, you never said you wanted to dial international on this carrier as well Smile. We'll give you another mask when one of the lads figures one out for you.

You mean you couldn't read my mind! And I thought you were good!!

I could setup another carrier for the UK, I have signed up with so many in the last few days I could probably use one for each of my contacts!!
On a serious note I couldn't get the mask to work even for local and US calls. I must be doing something wrong. Probably something simple, normally is. Thanks again for perservering with me I appreciate your help.

Regards,
Del  

[del]

Hi Selintra,

I have forwarded the ports as you suggested but it still doesn't work. Do I need to open these ports on the server? I have run the Advanced Port Scanner from RadMin and it says that ports 3129-65535 are closed. Funny thing is it gives the same results for my asterisk box even though I have done

config setprop masq status disabled
reboot

Is there a better way to test them? If I do need to open ports is there still a contrib/addon to do this?
I have also downloaded and used a testports prog from voipuser.org, which basically pings their server and this also failed on port 5060. I have now run out of ideas. Funny thing is all my tried and tested sip accounts work both ways when setup directly in a softphone!

Regards,
Del

[SARK devs]

Hello Del,

Let's go back to first principles.  You have an SME Server-Gateway running the perimiter, right? - Let's call it server A.  Then you have a test server-only (running asterisk) somewhere on the internal net - let's call it server B.

What subnet id are you running (e.g. 192.168.1.x)?

You have shut masq down on B - yes?

Presumably you are forwarding ports from A to B - yes?

First point - we don't think you can port range forward from SME 7.0 (any takers on this?)  so how have you specified the port range for 10000 to 20000?

Presumbly you've also forwarded 5060 from A to B. - yes?

Get back to me on these please and we'll take it from there.

In the meantime, take a look at iptraf and experiment with using it to watch the 5060 packets arriving at and leaving your servers.  This may well give you a clue as to where the blockage is.

Kind Regards

Selintra




 










Which

[del]

Hi Selintra,

Let's go back to first principles. You have an SME Server-Gateway running the perimiter, right? - Let's call it server A. Then you have a test server-only (running asterisk) somewhere on the internal net - let's call it server B.

This correct.

What subnet id are you running (e.g. 192.168.1.x)?

10.0.0.x
255.255.255.0

You have shut masq down on B - yes?

Yes by doing: config setprop masq status disabled
reboot

First point - we don't think you can port range forward from SME 7.0 (any takers on this?) so how have you specified the port range for 10000 to 20000?

I haven't forwarded this range due to the fact that in your earlier post you said

If this results in one-way sound when you test inbound calls then you should also port-range forward 10000:20000 (UDP).

Because I haven't actually been able to receive any inbound calls to verify if I have two-way sound or not.

Presumbly you've also forwarded 5060 from A to B. - yes?

Yes
I am using X-Lite softphones, I tried to use SJPhone but it says in the "screen area"

SIP not registered
Host Remote 10.0.0.60
NAT/Firewall Port Restricted Cone NAT

I don't know why, I have followed the settings in your documentation (Chapter 23), X-Lite works Ok for internal ext to ext calls and I can make outbound calls to US & UK numbers.
I hope this is enough info for you. Please let me know if you need to know
anything else. Thanks again for the help and advice.

Regards,
Del

[SARK devs]

OK - so far so good.

You need to set localnet (in headers, in sip.conf) to 10.0.0.0.  You also need to set your true external ip in "external ip" in globals.

ok now need to fetch up iptraf on both A and B and watch what happens when you bring asterisk up on B.  It should attempt to register with your carrier (using 5060).  You can watch the packets leaving and arriving at each machine and see if there are any obvious blockages.

Let us know how you get on.

Kind Regards

Selintra

[del]

Hi Selintra,

You need to set localnet (in headers, in sip.conf) to 10.0.0.0. You also need to set your true external ip in "external ip" in globals

Done

ok now need to fetch up iptraf on both A and B and watch what happens when you bring asterisk up on B. It should attempt to register with your carrier (using 5060). You can watch the packets leaving and arriving at each machine and see if there are any obvious blockages.

Done this, I don't really know what to look for here but there is plenty of activity on 5060. I am at work now, but I have an idea  what if I just shutdown server A and reconfigure server B (asterisk) as server/gateway? This will in effect eliminate any port forwarding issues, wouldn't it? Let me know if this is a good idea and I will do it as soon as I get home.

Regards,
Del

[SARK devs]

what if I just shutdown server A and reconfigure server B (asterisk) as server/gateway?

That's pretty much how we run most of our customer servers.  We have one or two running server-only behind Cisco firewalls and the like but none running behind SME server/gateway boxes.  

In server-gateway mode, SAIL automatically opens the necessary ports for SIP traffic so you shouldn't have anything else to do.  It should just work.

Let us know how you get on.

[del]

Hi Selintra,

First of all I apologise, this is going to be a long post.

When I got home server B HDD had died!  So I have now replaced the HDD and reinstalled SME 7 and installed the latest rpms as per your other post. So if I go back to basics this what I have done:
Installed SME 7 and configured as server/gateway connected to the outside world with Brighthouse Road Runner high speed cable modem. Internet works, all computers on LAN can also connect.
Ran yum update
Then installed the following rpms:
smeserver-asterisk-zappri-MPP-1.2.6-1.i686.rpm
smeserver-asterisk-1.2.10-1.i686.rpm
smeserver-asterisk-zappri-MPP-1.2.6-1.i686.rpm
I have no other contribs or addons installed
Set localnet=10.0.0.0/255.255.255.0 in sip.conf
Set my External IP to the REAL IP in Globals
Next, added 3 ext (5000,5001 and 5002) called each extension, so far so good
Next added sipdiscount and stanaphone as new carriers
Then added 2 trunks (1 is sipdiscount for outbound and 1 is stanaphone for inbound).
Next I created 2 dialing rules: _001xxxxxxxxxx for US calls and 0044xxxxxxxxxx for UK calls both set to use the sipdiscount trunk. I dialed out to a US number OK, sound was a bit patchy but it did work. Couldn't try the UK. all my mates are in bed!
Then I tried to call my stanaphone number and still it doesn't ring in
I have looked on their site and everything is set according to their guides. I really am stumped now
I thought maybe it is not possible to use stanaphone but there are people using it with asterisk and asterisk@home (a search on their forum showed this). Can you recommend any VoIP/SIP provider that I can try, that you know works with sail? Thanks for reading this and all the help you have offered so far.

Regards,
Del

[fpausp]

Hi All,
Hi Selintra,

I Still cant call out via ISDN, the ISDN-Card has the exact same settings as in your doc. I made a route _0XXX. ... is it maybe the Country Identifier ?

I live in Austria=at


regards
fpausp

[SARK devs]

Hi fpausp

We really need to see the asterisk log to understand what's going on.

Can you run a call (outbound and inbound) with the console in verbose mode?  (log on with asterisk -rvvvv).

Copy the messages and send them to us at admin@selintra.com

Also - which release are you running?

While you're at it, send us the output from lsmod, cat /proc/zaptel/1,  lspci -v and lspci -vn

Thanks

Selintra

[SARK devs]

HI Del,

This maybe doesn't look like a firewall problem.   Here's what to do...

modify /etc/asterisk/logger.conf;  look for the line which says...

;full => notice,warning,error,debug,verbose

Remove the semi-colon and save the file.

Stop and start asterisk.

Log in to asterisk with

asterisk -rvvvv

Type in the command

sip debug

Now try your inbound call...

You should see a flurry of activity at the asterisk console.  If you don't then it's a firewall problem and the messages are not reaching asterisk.  

Now type

sip no debug

recomment out the statement in etc/asterisk/logger.conf and re-save it.

Open server-manager and clisk on View Log Files.

In the drop down you will find a log called asterisk/full.  Open it.

Go to the end of the log and copy the messages from your dial in attempt (most of the messages are time stamped so it's isn't hard to figure out which ones you need.

Send them to us at admin@selintra.com

ALTERNATIVE CARRIERS

In UK we have our own Selintra carrier service  (we either use one of the Tier 1 VOIP carriers or terminate directly to a big Ericsson switch in Telecity).

Other good UK carriers are Telappliant (www.voiptalk.org) and Gradwell (www.gradwell.com).  They use Gamma and Global Crossing.

Best

Selintra