UKVoIPTalk.com

OK guys - here's another one...

I've set up a Linksys/Sipura 3102 with Asterisk. It works OK but not
reliably - ie: I can use it to ring other IP phones on the network or
can call it from other phones for a few times and then it needs a
reboot. I'm currently playing with the SIP settings but any ideas would
be appreciated. Haven't even hooked it up to the PSTN yet.

I've also found that although I have enabled the Web interface from the
'Internet' side, the interface dies after a minute or so, but it works
fine from the 'local' side.

Anyway - can anyone advise on a dial plan that will do the following:

All outbound calls to go out via VoIP EXCEPT 999 and 112 which should
use the PSTN line. Calls starting 9 also to go out via PSTN (eg: '9' for
a PSTN line - like 901234 567 8900).

Thanks

On Wed, 04 Oct 2006 20:52:15 +0100, linker3000
<linker3000@google-minushyphenmail.com> wrote:
[color=blue]
>OK guys - here's another one...
>
>I've set up a Linksys/Sipura 3102 with Asterisk. It works OK but not
>reliably - ie: I can use it to ring other IP phones on the network or
>can call it from other phones for a few times and then it needs a
>reboot. I'm currently playing with the SIP settings but any ideas would
>be appreciated. Haven't even hooked it up to the PSTN yet.
>
>I've also found that although I have enabled the Web interface from the
>'Internet' side, the interface dies after a minute or so, but it works
>fine from the 'local' side.
>
>Anyway - can anyone advise on a dial plan that will do the following:
>
>All outbound calls to go out via VoIP EXCEPT 999 and 112 which should
>use the PSTN line. Calls starting 9 also to go out via PSTN (eg: '9' for
>a PSTN line - like 901234 567 8900).
>
>Thanks[/color]
Are you wanting dial plan help for the SPA-3000 or do you want help
connected with Asterisk?
If it is SPA-3000 can I suggest that you learn the basics of how to
construct a dial plan. Dialing 9 to route all the numbers that need to
go via PSTN is OK for the exception, where your VSP is down, but not
for day to day operations. If you construct your dial plan properly
you should have to think whether a number is going via VOIP or PSTN -
the dial plan should take care of that. Further, as you are developing
you system you may want to make changes, coming back here every time
to get a new dial plan isn't going to be very good regime.
Can I suggest that you take a look at some notes I have made and, if
after that, you are stuck then please come back here for help.
Check out the notes on
[url]http://www.leafcom.co.uk[/url]
Remove 'no_spam_' from email address.
Skype Free Zone!!

On Wed, 2006-10-04 at 20:52 +0100, linker3000 wrote:[color=blue]
> OK guys - here's another one...
>
> I've set up a Linksys/Sipura 3102 with Asterisk. It works OK but not
> reliably - ie: I can use it to ring other IP phones on the network or
> can call it from other phones for a few times and then it needs a
> reboot. I'm currently playing with the SIP settings but any ideas would
> be appreciated. Haven't even hooked it up to the PSTN yet.
>
> I've also found that although I have enabled the Web interface from the
> 'Internet' side, the interface dies after a minute or so, but it works
> fine from the 'local' side.
>
> Anyway - can anyone advise on a dial plan that will do the following:
>
> All outbound calls to go out via VoIP EXCEPT 999 and 112 which should
> use the PSTN line. Calls starting 9 also to go out via PSTN (eg: '9' for
> a PSTN line - like 901234 567 8900).
>
> Thanks[/color]

Hi,

Will your PSTN be attached to the * server or connected to the Sipura
device?

Open up a shell to your * box. If your using Trixbox i believe it
includes a Java SSH client. If not download putty* and login to your
asterisk server (using asterisk's IP). once in type 'asterisk
-rvvvvvv'. This will get you into the asterisk console. When you
make/recieve calls you can see what asterisk is doing. Watch for when
the line ceases and report back what appears on the screen.

Also commands such as 'sip show registry' and 'sip show peers' will give
you some idea of the connection status of the extenions(peers) and voip
provider(registry).

Let us know how you get on!


*Putty client: [url]http://the.earth.li/~sgtatham/putty/latest/x86/putty.exe[/url]

Brian A wrote:[color=blue]
> On Wed, 04 Oct 2006 20:52:15 +0100, linker3000
> <linker3000@google-minushyphenmail.com> wrote:
>[color=green]
>> OK guys - here's another one...
>>
>> I've set up a Linksys/Sipura 3102 with Asterisk. It works OK but not
>> reliably - ie: I can use it to ring other IP phones on the network or
>> can call it from other phones for a few times and then it needs a
>> reboot. I'm currently playing with the SIP settings but any ideas would
>> be appreciated. Haven't even hooked it up to the PSTN yet.
>>
>> I've also found that although I have enabled the Web interface from the
>> 'Internet' side, the interface dies after a minute or so, but it works
>> fine from the 'local' side.
>>
>> Anyway - can anyone advise on a dial plan that will do the following:
>>
>> All outbound calls to go out via VoIP EXCEPT 999 and 112 which should
>> use the PSTN line. Calls starting 9 also to go out via PSTN (eg: '9' for
>> a PSTN line - like 901234 567 8900).
>>
>> Thanks[/color]
> Are you wanting dial plan help for the SPA-3000 or do you want help
> connected with Asterisk?
> If it is SPA-3000 can I suggest that you learn the basics of how to
> construct a dial plan. Dialing 9 to route all the numbers that need to
> go via PSTN is OK for the exception, where your VSP is down, but not
> for day to day operations. If you construct your dial plan properly
> you should have to think whether a number is going via VOIP or PSTN -
> the dial plan should take care of that. Further, as you are developing
> you system you may want to make changes, coming back here every time
> to get a new dial plan isn't going to be very good regime.
> Can I suggest that you take a look at some notes I have made and, if
> after that, you are stuck then please come back here for help.
> Check out the notes on
> [url]http://www.leafcom.co.uk[/url]
> Remove 'no_spam_' from email address.
> Skype Free Zone!![/color]

Hi Brian,

I will have a look at your notes - thanks.

This setup is for a small business with one incoming phone line hence
all outbound calls are planned to go via VoIP to keep the line free for
one inbound call - not ideal but best under the current circumstances.
The system is unlikely to develop any further than this.

Daviey wrote:[color=blue]
> On Wed, 2006-10-04 at 20:52 +0100, linker3000 wrote:[color=green]
>> OK guys - here's another one...
>>
>> I've set up a Linksys/Sipura 3102 with Asterisk. It works OK but not
>> reliably - ie: I can use it to ring other IP phones on the network or
>> can call it from other phones for a few times and then it needs a
>> reboot. I'm currently playing with the SIP settings but any ideas would
>> be appreciated. Haven't even hooked it up to the PSTN yet.
>>
>> I've also found that although I have enabled the Web interface from the
>> 'Internet' side, the interface dies after a minute or so, but it works
>> fine from the 'local' side.
>>
>> Anyway - can anyone advise on a dial plan that will do the following:
>>
>> All outbound calls to go out via VoIP EXCEPT 999 and 112 which should
>> use the PSTN line. Calls starting 9 also to go out via PSTN (eg: '9' for
>> a PSTN line - like 901234 567 8900).
>>
>> Thanks[/color]
>
> Hi,
>
> Will your PSTN be attached to the * server or connected to the Sipura
> device?
>
> Open up a shell to your * box. If your using Trixbox i believe it
> includes a Java SSH client. If not download putty* and login to your
> asterisk server (using asterisk's IP). once in type 'asterisk
> -rvvvvvv'. This will get you into the asterisk console. When you
> make/recieve calls you can see what asterisk is doing. Watch for when
> the line ceases and report back what appears on the screen.
>
> Also commands such as 'sip show registry' and 'sip show peers' will give
> you some idea of the connection status of the extenions(peers) and voip
> provider(registry).
>
> Let us know how you get on!
>
>
> *Putty client: [url]http://the.earth.li/~sgtatham/putty/latest/x86/putty.exe[/url]
>[/color]
Hi Daviey,

The PSTN will be integrated with the * (Trixbox) system via the SPA3102
- at the moment, the company takes a call on the PSTN line with a DECT
phone and they walk it to the recipient so I want instead for the
inbound PSTN call to be handled by Trixbox so they can transfer it to
the required extension. All other aspects of the system are working fine
- 7 extensions all with inbound 0845 numbers routing to the desks (apart
from inbound fax detection but I an working on it).