UKVoIPTalk.com

Has anyone succeeded in setting up Asterisk to send caller ID with
Betamax/Finarea providers?

With SMSListo.com I've registered my number using the Windows application,
received the confirmation call, and the app says my number is OK.

On an ATA the settings to provide CLID are:
username=00441234567890 [my registered CLID number]
auth user=myusername
password=mypassword
This worked fine for a VOIPCheap.com account I have (I don't have the ATA
here to play with)

So I tried that in Asterisk 1.2's sip.conf:
[smslisto]
type=peer
username=00441234567890
auth=myusername
secret=mypassword
fromuser=myusername
host=sip.smslisto.com
fromdomain=sip.smslisto.com
realm=sip.smslisto.com
context=smslisto.com
insecure=very
caninvite=no
canreinvite=no
qualify=no
disallow=all
allow=ulaw

But Asterisk can't make calls with these settings. It calls fine when
username=myusername, secret=mypassword and auth isn't defined.

I've also tried
callerid="Sample" <00441234567890>
in that section of sip.conf,
exten => _X.,1,Set(CALLERID(num)=+441234567890)
in my extension context, and
CallerID: Sample <+441234567890>
in a .call file

and similiar attempts with 0044 instead of +44.

Any other suggestions? When using the 'auth' field with an ATA, what does
that actually do in the SIP protocol? Does it differ from Asterisk's auth?

Thanks
Theo

Theo Markettos brought next idea :[color=blue]
> Has anyone succeeded in setting up Asterisk to send caller ID with
> Betamax/Finarea providers?
>
> With SMSListo.com I've registered my number using the Windows application,
> received the confirmation call, and the app says my number is OK.
>
> On an ATA the settings to provide CLID are:
> username=00441234567890 [my registered CLID number]
> auth user=myusername
> password=mypassword
> This worked fine for a VOIPCheap.com account I have (I don't have the ATA
> here to play with)
>
> So I tried that in Asterisk 1.2's sip.conf:
> [smslisto]
> type=peer
> username=00441234567890
> auth=myusername
> secret=mypassword
> fromuser=myusername
> host=sip.smslisto.com
> fromdomain=sip.smslisto.com
> realm=sip.smslisto.com
> context=smslisto.com
> insecure=very
> caninvite=no
> canreinvite=no
> qualify=no
> disallow=all
> allow=ulaw
>
> But Asterisk can't make calls with these settings. It calls fine when
> username=myusername, secret=mypassword and auth isn't defined.
>
> I've also tried
> callerid="Sample" <00441234567890>
> in that section of sip.conf,
> exten => _X.,1,Set(CALLERID(num)=+441234567890)
> in my extension context, and
> CallerID: Sample <+441234567890>
> in a .call file
>
> and similiar attempts with 0044 instead of +44.
>
> Any other suggestions? When using the 'auth' field with an ATA, what does
> that actually do in the SIP protocol? Does it differ from Asterisk's auth?
>
> Thanks
> Theo[/color]

Here's my working setup:

username=myusername
type=friend
secret=mypassword
qualify=yes
nat=yes
insecure=very
host=sip.smslisto.com
fromuser=00448700xxxxxx
fromdomain=smslisto.com
dtmfmode=rfc2833
disallow=all
authuser=myusername
allow=ulaw&alaw

Jono <nothanks@blueyonder.invalid> wrote:[color=blue]
> Here's my working setup:[/color]
[snip]

That indeed works. Thanks!

Theo

Theo Markettos <theom+news@chiark.greenend.org.uk> wrote:[color=blue]
> That indeed works. Thanks![/color]

I've now added this to the Asterisk wiki:
[url]http://www.voip-info.org/wiki/view/Finarea+SA[/url]

Theo

After serious thinking Theo Markettos wrote :[color=blue]
> Theo Markettos <theom+news@chiark.greenend.org.uk> wrote:[color=green]
>> That indeed works. Thanks![/color]
>
> I've now added this to the Asterisk wiki:
> [url]http://www.voip-info.org/wiki/view/Finarea+SA[/url]
>
> Theo[/color]

Good stuff.

There is another way to achieve it, though I've not had time to
experiment.....I'm sure Sparks was the individual ... or was it
Herman.. who had it set up the ohther way.? Basically, "the other way"
allows you to leave your trunk with Generic settings and put your
outbound CLI in the appropriate field for a specific extension.

So, the CLI can be set by extension, rather than be universal for the
trunk.

on 14/07/2008, Jono supposed :[color=blue]
> After serious thinking Theo Markettos wrote :[color=green]
>> Theo Markettos <theom+news@chiark.greenend.org.uk> wrote:[color=darkred]
>>> That indeed works. Thanks![/color]
>>
>> I've now added this to the Asterisk wiki:
>> [url]http://www.voip-info.org/wiki/view/Finarea+SA[/url]
>>
>> Theo[/color]
>
> Good stuff.
>
> There is another way to achieve it, though I've not had time to
> experiment.....I'm sure Sparks was the individual ... or was it Herman.. who
> had it set up the ohther way.? Basically, "the other way" allows you to leave
> your trunk with Generic settings and put your outbound CLI in the appropriate
> field for a specific extension.
>
> So, the CLI can be set by extension, rather than be universal for the trunk.[/color]

So, leaving the fromuser=0044xxxxxxxx line out of the config, I can (in
Free PBX) enter any verified number in the Outbound CLI box on EITHER
the trunk or the extension...and it works.

Jono <nothanks@blueyonder.invalid> wrote:[color=blue]
> So, leaving the fromuser=0044xxxxxxxx line out of the config, I can (in
> Free PBX) enter any verified number in the Outbound CLI box on EITHER
> the trunk or the extension...and it works.[/color]

Hmmm... if I remove the fromuser line, Asterisk's own CLID doesn't work for
me with:

exten => 1236,7,Set(${CALLERID(num)}=0044myoutboundCLID)
exten => 1236,8,Dial(SIP/0044mymobile@smslisto-com)
(but it rings, Asterisk 1.4 this time)

What do you mean by 'on EITHER the trunk or the extension'?

Theo

Theo Markettos submitted this idea :[color=blue]
> Jono <nothanks@blueyonder.invalid> wrote:[color=green]
>> So, leaving the fromuser=0044xxxxxxxx line out of the config, I can (in
>> Free PBX) enter any verified number in the Outbound CLI box on EITHER
>> the trunk or the extension...and it works.[/color]
>
> Hmmm... if I remove the fromuser line, Asterisk's own CLID doesn't work for
> me with:
>
> exten => 1236,7,Set(${CALLERID(num)}=0044myoutboundCLID)
> exten => 1236,8,Dial(SIP/0044mymobile@smslisto-com)
> (but it rings, Asterisk 1.4 this time)
>
> What do you mean by 'on EITHER the trunk or the extension'?
>[/color]

I take it you're not a FeePBX user...

I can enter the CLI for the trunk; any calls going out on that trunk
will default to show that CLI.

I can also enter the Outbound CLI for an extension. This has the effect
of over-riding the default CLI set in the trunk.

This means that I can have all my extensions sending my home number as
CLI yet my sister's extension (remote) correctly sends her home
number...using the same tunk as me.

Jono <nothanks@blueyonder.invalid> wrote:[color=blue]
> I take it you're not a FeePBX user...[/color]

No, just plain old Asterisk for me :)
[color=blue]
> I can enter the CLI for the trunk; any calls going out on that trunk
> will default to show that CLI.[/color]

Ah, right. In plain Asterisk-speak that sounds equivalent to this in
sip.conf:

[someprovider]
callerid=00441234567...
host=sip.someprovider.net
....
[color=blue]
> I can also enter the Outbound CLI for an extension. This has the effect
> of over-riding the default CLI set in the trunk.[/color]

While that's something like this in extensions.conf:
[someextensioncontext]
exten => 1236,1,Answer
exten => 1236,2,Play(tt-weasels)
exten => 1236,3,Set(${CALLERID(num)}=0044.....)
(and various other ways of saying the same thing)

But neither work for me :(

Theo

On 16 Jul, 22:01, Jono <notha...@blueyonder.invalid> wrote:
[snip][color=blue]
> I can enter the CLI for the trunk; any calls going out on that trunk
> will default to show that CLI.
>
> I can also enter the Outbound CLI for an extension. This has the effect
> of over-riding the default CLI set in the trunk.
>
> This means that I can have all my extensions sending my home number as
> CLI yet my sister's extension (remote) correctly sends her home
> number...using the same tunk as me.[/color]

Useful and pratical tip, thanks Jono.

But, do you know of any way to send CLI based on number called? For
example, I'd like London based contacts to see my London number when
they receive a call, Cambridge contacts to see a Cambridge number and
Paris contacts to see a Paris CLI - no matter which extension the call
is made from.

Of course, incoming calls would not be affected as they can still be
routed to specific extensions as they always have been.....