asterisk users - Aug 2004 - How to run different codecs between the same endpoints on an IAX trunk?

Or perhaps how to configure and refer to two parallel IAX trunks with
different codecs?

I have a situation where I'm using G.729A as my IAX trunking codec. Now I
need to push some short duration, low bitrate modem traffic over the link (a
credit card terminal). Obviously the modem audio isn't going to survive the
G.729 codec process intact, so for the times the device is used I'd like to
service calls from that device (and only that device) with a higher-data
rate codec.

Any suggestions?

Kris Boutilier
Information Systems Coordinator
Sunshine Coast Regional District

I believe you can do that kind of thing with SIP, using the
SetVar(${SIP_CODEC}="CODEC"}) in your extensions.conf... So, For
example, if
the extension of your card machine were say '100' then you would do
something like this

[outgoing]

exten => _1NXXNXXXXXX,1,Gotoif($[${CALLERID_NUM} = '100']?2:3)
exten => _1NXXNXXXXXX,2,SetVar(${SIP_CODEC}="ulaw")
exten =>
_1NXXNXXXXXX,3,Dial(IAX2/user:password@iax2.otherserver.com/$EXTEN,30)

I don't know if this works with IAX...

    -Chris

----- Original Message -----
From: "Kris Boutilier" <Kris.Boutilier@scrd.bc.ca>
To: <asterisk-users@lists.digium.com>
Sent: Thursday, August 19, 2004 3:11 PM
Subject: [Asterisk-Users] How to run different codecs between the same
endpoints on an IAX trunk?

> Or perhaps how to configure and refer to two parallel IAX trunks with
> different codecs?
>
> I have a situation where I'm using G.729A as my IAX trunking codec. Now
I
> need to push some short duration, low bitrate modem traffic over the link
(a
> credit card terminal). Obviously the modem audio isn't going to survive
the
> G.729 codec process intact, so for the times the device is used I'd
like
to
> service calls from that device (and only that device) with a higher-data
> rate codec.
>
> Any suggestions?
>
> Kris Boutilier
> Information Systems Coordinator
> Sunshine Coast Regional District
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users

> Or perhaps how to configure and refer to two parallel IAX trunks with
> different codecs?
> 
...
> 
> Any suggestions?
Well, push comes to shove, you can define two different entries in
iax.conf with different protocols. That is:

Box A:
[boxb-voice]
type=peer
host=boxb
disallow=all
allow=gsm
trunk=yes

[boxb-data]
type=peer
host=boxb
disallow=all
allow=ulaw

You'll lose some of the benefits of trunking for the data calls, but
if they're short-lived and not terribly common, this might be the
easiest solution.

--Will Glynn
Freedom Healthcare Group

On Thu, 19 Aug 2004, Kris Boutilier wrote:
> Or perhaps how to configure and refer to two parallel IAX trunks with
> different codecs?
> 
> I have a situation where I'm using G.729A as my IAX trunking codec. Now
I
> need to push some short duration, low bitrate modem traffic over the link
(a
> credit card terminal). Obviously the modem audio isn't going to survive
the
> G.729 codec process intact, so for the times the device is used I'd
like to
> service calls from that device (and only that device) with a higher-data
> rate codec.

How about two IP addresses at the remote end, configure two peers, one 
with trunking anf G729, one without and G711?

Steve

On Thursday 19 August 2004 18:11, Kris Boutilier wrote:
> I have a situation where I'm using G.729A as my IAX trunking codec. Now
I
> need to push some short duration, low bitrate modem traffic over the link
> (a credit card terminal). Obviously the modem audio isn't going to
survive
> the G.729 codec process intact, so for the times the device is used I'd
> like to service calls from that device (and only that device) with a
> higher-data rate codec.
Am I missing something?  I do this with my Office Asterisk and Colocation 
Asterisk boxes all the time.

On the colocation asterisk box (office asterisk box is pretty much identical 
to allow outgoing calls and outgoing faxes):

[officeasterisk]
type=peer
host=192.168.2.2
qualify=500
notransfer=yes

[officephone]
type=user
context=officephone
host=192.168.2.2
qualify=500
disallow=ulaw
notransfer=yes

[officefax]
type=user
context=faxout
host=192.168.2.2
qualify=500
disallow=all
allow=ulaw
notransfer=yes

When a fax call comes in from the PRI, it calls officefax.  When a phone call 
comes in, it calls officephone.  Similarly when someone calls out, it calls 
colophone (same as officephone but with different host), and likewise, 
outgoing faxes use colofax.

Seems to be working fine here, and has since June when we moved.  :-)

Now I have turned off trunking between officeasterisk and coloasterisk -- the 
20040806 CVS HEAD seemed to not like to trunk between two boxes running that 
code, but would trunk just fine to RC1 (with the trunking fixes that Steve 
did)...

Is this the kind of thing you want?  I was trunking between office and 
colocation up until the last source update I did and it seemed to work just 
fine.

-A.

> So it's not possible to trunk calls between two points if 
> they use different codecs?
In principle it is. As a protocol, IAX2 doesn't care. It's
just a matter of getting it configured. :-)
> Where does * tell you this when it runs... iax2 debug
> trunking output seems to indicate that it's trunking
> multiple codecs between two points (at least in my
> interpretation).
Well, the trunking code might be smarter than I anticipated :-)

--Will Glynn
Freedom Healthcare Group

Hi, All,

I run Asterisk with Digium hardware TDM 4 FXO & TE410P(E1).
When i dial PSTN to FXO or E1 PRI and softswich to SIP agent, the codec
always use
ulaw(TDM 4FXO) or alaw(TE410P).

Is it possible to change the default codec type to ILBC/GSM/G.729 when
PSTN -> SIP softswitch?

Thanks,