Asterisk Setup for Community VoIP

ok, that's better! - nearly there

The error you're getting now is because no valid codecs are available.

See my example in the top post of this thread.

--ian

aha, you beauty, its sorted...thank you, i had only allow=ilbc (which incidently works fine for incoming?) is it possible that you can let us use ilbc please?

Many thanks

Rick

riksta :

aha, you beauty, its sorted...thank you

Great!

Quote:

i had only allow=ilbc (which incidently works fine for incoming?)

Ah, yeah - I tried ilbc when I was beta testing the outoging service and as you say it doesn't work. I asked asked about it but Dean/TJ have been busy and no one got round to replying.

I believe ilbc is better quality than gsm - about on par with normal pstn but not quite as good as g729. I'm currently using g729 which works fine with the outbound service (but not incoming) but obviously I am limited to the number of licences I bought (2).

Quote:

is it possible that you can let us use ilbc please?

Hopefully Dean/Tj will see this thread and comment.

Is there anyone else who would lile to see ilbc?

--ian

Quote:

is it possible that you can let us use ilbc please?

Sorry must have missed the original request for that Ian......

We can't do I'm afraid - there is no ilbc for Cisco, and we're terminating via Cisco hardware.

Dean

ichilton :

Is there anyone else who would lile to see ilbc? --ian

Yeah, I would. It's the least bandwidth intensive non-patented codec out there (and CPU time really isn't an issue for this sort of application, at least on the client side).

dean :

We can't do I'm afraid - there is no ilbc for Cisco, and we're terminating via Cisco hardware.

Drat, that's a shame.

Unless you considered putting something else acting as a gateway to the SIP proxy, like an Asterisk box to provide IAX2 outbound with ilbc

Is anyone else interested in IAX outbound or ilbc? (it might not be possible, but if there is an interest, Dean/Tj might be able to think up a solution...)

--ian

forkqueue :

It's the least bandwidth intensive non-patented codec out there (and CPU time really isn't an issue for this sort of application, at least on the client side).

Yep, that's what I thought.

Have you used it on any other providers then? - how does it compare in real usage to gsm, g729, g711 (ulaw) or even normal pstn?

--ian

Having trouble dialling out, with asterisk, im seeing whats pasted below, my dial line is

exten => _99.,1,Dial(SIP/${EXTEN:2}@voipuser)

and the following is the result


-- Executing Dial("Zap/2-1", "SIP/0035314XXXXXX@voipuser") in new stack
-- Called 0035314XXXXXX@voipuser
Jan 12 12:45:20 NOTICE[2399]: chan_sip.c:6750 handle_response: Failed to authenticate on INVITE to '"14XXXXXX" <sip:MYUSERNAME@voipuser.org>;tag=as4e8cfc32'


Anyone have any idea?

dean :

Quote: is it possible that you can let us use ilbc please? We can't do I'm afraid - there is no ilbc for Cisco, and we're terminating via Cisco hardware.

I deduce this is also the answer to my question, isn't it?

Quote:

Are you planning to support the free (as in freedom) Speex codec?

voditrord :

Are you planning to support the free (as in freedom) Speex codec?

I'll let Dean give the definative answer but I expect that if it doesn't work, it's not supported. I assume they'll have all supported codec's turned on.

[edit] -> To keep things in sync, dean replied to the other thread saying:

Quote:

Speex : probably not at the moment, unless it becomes supported on Cisco hardware.

Out of interest, why do you want to use Speex? - isn't it much lower quality?

--ian

Hi,

stephen :

exten => _99.,1,Dial(SIP/${EXTEN:2}@voipuser)

That looks fine.

Quote:

-- Executing Dial("Zap/2-1", "SIP/0035314XXXXXX@voipuser") in new stack -- Called 0035314XXXXXX@voipuser Jan 12 12:45:20 NOTICE[2399]: chan_sip.c:6750 handle_response: Failed to authenticate on INVITE to '"14XXXXXX" <sip:MYUSERNAME@voipuser.org>;tag=as4e8cfc32'

Have you set it up the same as I did and posted above?

Try typing:
sip show registry
..does this show you as registered?

--ian

Yep setup as above, when i hangup i see the following
-- Channel 0/2, span 3 got hangup
== Spawn extension (default, 990035314XXXXXX, 1) exited non-zero on 'Zap/8-1'
-- Hungup 'Zap/8-1'
-- Got SIP response 483 "Too Many Hops" back from 216.127.66.119

Hi,

What does: sip show registry say though?

stephen :

== Spawn extension (default, 990035314XXXXXX, 1) exited non-zero on 'Zap/8-1' -- Hungup 'Zap/8-1' -- Got SIP response 483 "Too Many Hops" back from 216.127.66.119

Zap? - Are you using a Zaptel FXS with an analogue phone connected?

Any chance of trying to dial a uk phone number?


Thanks

--ian

ichilton :

Out of interest, why do you want to use Speex? - isn't it much lower quality?

I didn't say I wanted to use it, I was just curious . I can't tell you about the quality (haven't compared the codecs myself nor seen others' comparisons), but at least there are a few things that make Speex interesting:

*You can choose the bitrate, so you can use it with any type of connection.
*It's free software.
*It's actively developed, so it will probably be better quality in the future.

voditrord :

but at least there are a few things that make Speex interesting:

Ah, interesting.

--ian