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

[SARK devs]

Hi Del,

Firstly, certain elements in their instructions don't make lot of sense to us.  However, we'll check them out tomorrow (UK Time).  Should have something for you when you wake up.

As to which version of asterisk & zappri you should have, it depends upon which version of SAIL you are running.  Look at the 2.1.13 quickstart notes here...

http://81.149.154.14/docs/cgi-bin/view/Main/SysVersion2Release1Issue13QN

The folders contain the correct versions.

Best

Jeff

[Tib]

You may have to down load the files from a diff mirror ... ibiblio has been down for days now ... yesterday I managed to get on there site and all the files were missing.

Here is one mirror that still works.

ftp://ftp.oss.cc.gatech.edu/pub/linux/distributions/smeserver/contribs

Regards,

Tib

[SARK devs]

Thanks Tib

Del

Latest asterisk rpms (for SAIL 2.1.13) are here...

ftp://ftp.oss.cc.gatech.edu/pub/linux/distributions/smeserver/contribs/selintra/RPMS/AsteriskForSail-2.1.13/None-ISDN/

Best

Selintra

[del]

Thanks Selintra,

I have the correct ones installed
Telasip said that this part was the most important:
To receive calls you must have:
Context=pstn OR =trunk
If not all calls will go to their voicemail system

Regards,
Del

[del]

Hi Selintra,

I must be doing something fundamentally wrong (probably just mentally!), I have created a new carrier called telesip-voicexpress using gw3.telasip.com as the host URL and yourusername:yourpassword@gw3.telasip.com as the registration string. I then created a new trunk using telasip-gw as the DiD as per telasip instructions, entered my username and password and then edited it so I could update  the registration string with  my username and password. Called my local number and it went straight to telasip voicemail just like they said it would if I didn't have Context=pstn OR =trunk in extension_custom.conf file. It may not e related but if I call ext to ext and leave a voicemail I don't receive an email notification and if I check the ext voicemail it says that I don't have any messages. My SME is configured as Server/Gateway so there is no router involved in the setup to mess up the ports etc. Funny thing is if I shutdown the server and replace it with a SME test box in exactly the same setup minus Sail and Asterisk, install Trixbox on another box and follow the procedure to add trunks etc. it all works, I can receive calls, I get my voicemail in my inbox and I can dial out. I could just leave it at that, but I don't really want to have 2 boxes running and I don't like to give up. Can you suggest anything?
Do I have to enter anything in the peer stanza box for the trunks?

Once again I will ask is there anyone in the USA using Sail?

Regards,
Del

[SARK devs]

Hello Del,

We cannot comment further at this point because we've had a bit of a mare attempting to set-up an account with telasip.   Their system seems to be hand cranked; it took them two days to acknowledge our order and then they sent us an account number but no phone number.  Any profit they may have made on the deal must be long gone.

A soon as we get a proper ID from them we'll run a registration from here.

Kind Regards

Selintra

[del]

Hi Selintra,

Thanks for the reply, I signed up with them on a Friday night, got my account details on the Tuesday, they said that they had run out of numbers for my area code, but I did get the number on Wednesday. I agree they are a bit slow but they get good reviews on all the Trixbox and Asterisk sites mainly because they don't have any hidden costs or small print and they are among the very few that DON'T charge to receive calls! I have made a couple of appeals to other US users but no one has come forward yet. I know we already get a pretty good deal with calls but not so good it's not worth trying to save a bit especially on international calls  I wait for you in anticipation

Regards,
Del

[SARK devs]

Hello Del,

Our telasip number came through about ten minutes ago.  You can now call us at our "New York" office on 646 862 1594 (go ahead - it's patched straight through to my desk so you'll either get me or my voice mail).

The setup for the telasip SIP Stack is entirely conventional (except that it doesn't drive on the final registration couplet, which is no bad thing).

In order to avoid confusion - I will put up some screen shots of the actual screens used to construct the telasip carrier entry and corresponding trunk entry for you to reference.  It will appear in a small how-to here.

http://81.149.154.14/docs/cgi-bin/view/Main/DocChapter259

Kind Regards

Selintra

[del]

Hi Selintra,

Thank you, I followed the screen shots and it works  I can now make and receive calls using my telasip account Now let the games begin
I have setup a simple group called allphones that, you've guessed it, rings all my extensions on incoming calls, when I ring in from my cellphone, they all ring  if I let it continue it goes to the operator's ext (5000) and I hear the greeting, I then leave a message and low and behold I get an email with the message attached and I can dial *50* followed by  my password and I am told I have one new message, I then listen to it If I call ext 5000 from another ext I get the same greeting, I leave a message but I don't get an email and when I check with *50*  am told there are no new messages  Do I have to do something in  the setup for internal calls to work the same as external calls? Also can I assume from the docs that my PAP2 is the same as Sipura 2000?
Thanks again for all your help

Regards,
Del

[SARK devs]

hi Del,

Re Voicemail -  usually when we get reports of voicemail not working it is because the message is too short.  SAIL will delete any message which is shorter than 6 seconds and it will stop recording if it detects more than 6 seconds of silence.

The PAP2 browser interface is slightly different to the spa2002, however, the internal settings are identical and it will provision just the same as the 2002.

Kind Regards

Selintra

[del]

Hi Selintra,

Once again I must thank you, I tried the voice mail and made sure that the message was longer than six seconds and it works OK.   After reading the documentation am right in beleiving that I need tftp-server installed to configure the PAP2? If so how do I do that? I have done: yum install tftp-server and it comes up with the usual additional commands may be required etc. and at the end it says

No match for argument: tftp-server
Nothing to do
===========================================
No new rpms were installed. No additional commands are required
===========================================

But when I  go to globals in server-manager I still don't have the option to choose yes only No. Is there an rpm I need to download and install?
A search on Google found this rpm: tftp-server-0.39-1.i386.rpm could I maybe install this?    Also will I be able to use both ports on the PAP2? Thanks again.

Regards,
Del

[SARK devs]

try

Code: [Select]

yum --enablerepo=base install tftpserver
That should install tftp and xinet.d

The pap2 (spa200x) is configured as two separate extensions (they should both point to the same mac address).

Kind Regards

Selintra

[del]

Hi Selintra,

I have run

yum --enablerepo=base install tftpserver

and there was a lot of action, but no tftp server I then worked out that it may have been

yum --enablerepo=base install tftp-server

and it installed it OK. Only problem was when I rebooted my server asterisk didn't start  I had to start it manually, any ideas why this should happen?
Also my PAP2 will dial out to other extensions but won't receive calls, they go straight to voice mail. Thanks again.

Regards,
Del

[SARK devs]

re PAP2

You will need to send me the following...

a listing of sip.conf
a listing of the verbose asterisk log of a call being made to one of the PAP2 extensions (log in to the asterisk console with asterisk -rvvvv )

Send them to admin@selintra.com

It sounds like the PAP2 is not registering correctly.  Log in to asterisk (as above) then power cycle the PAP2 and watch the log to see if it the registration is accepted.  Send the output with the rest of the stuff.

Kind Regards

Selintra

[del]

Hi Selintra,

I will do that later, unfortunatley I have to work today One thing, my PAP2 has a static IP (did this to use with Trixbox) does it need to be DHCP? Also this is a ex-vonage PAP2 that as been unlocked.
Will email results when I get home.

Regards,
Del