asterisk users - Jul 2004 - G.729 codec doesn't seem to work *even* after installing the license

Hi,

I am trying to post this again as I am getting no answers and the
support@digium.com bounces...

(I have searched the whole list and can't find the answer either) 

I have installed a 5 user license for G.729 and want to route calls through
Asterisk from my G.729 phone to Cisco AS5300 also using G729. 

Both Cisco and the phone connect using this codec if I do not force the call
to go through *

However if I say canreinvite=no in the sip.conf for either of these gadgets,
the call will fail with No compatible codecs!

I have bought a 5 user license just to try and fix this, apparently it
doesn't work. I want to protect the Cisco gateway from unauthorized use, but
still using a
cost-effective codec such as g.723 or g.729 ? 

[codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec
Translator)

  == G.729 Host-ID:
5f:a1:18:82:47:6f:a8:f7:33:4e:7d:77:e8:1d:60:15:53:ec:49:aa

  == Found license 'G729-700241AB' providing 5 channels

  == Found total of 5 G.729 licenses

  == Registered translator 'g729tolin' from format G729A to SLINR, cost
2

  == Registered translator 'lintog729' from format SLINR to G729A, cost
12


I was hoping by letting it ring out, I would get a voice-mail message, but
that doesn't work either...
 

*CLI> Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No
compatible codecs!

    -- Executing Dial("SIP/67.23.212.25-0814f830",
"SIP/334|20") in new
stack

    -- Called 334

    -- SIP/334-26f8 is ringing

    -- Nobody picked up in 20000 ms

    -- Executing VoiceMail("SIP/67.23.212.25-0814f830",
"u334") in new stack

    -- Playing 'vm-theperson' (language 'en')

  == Spawn extension (default, 4084, 2) exited non-zero on
'SIP/67.23.212.25-0814f830'

 
I have dropped this question at the asterisk user list some days ago, but
it's being ignored... (or nobody has the answer)

Can anybody shed some light on this ?

Warmest Regards,

Walter Klomp

try sip debug and see what each side is offering in codecs (make sure yo 
u have allow=g729

Walter Klomp wrote:
> Hi,
> 
> I am trying to post this again as I am getting no answers and the
> support@digium.com bounces...
> 
> (I have searched the whole list and can't find the answer either) 
> 
> I have installed a 5 user license for G.729 and want to route calls through
> Asterisk from my G.729 phone to Cisco AS5300 also using G729. 
> 
> Both Cisco and the phone connect using this codec if I do not force the
call
> to go through *
> 
> However if I say canreinvite=no in the sip.conf for either of these
gadgets,
> the call will fail with No compatible codecs!
> 
> I have bought a 5 user license just to try and fix this, apparently it
> doesn't work. I want to protect the Cisco gateway from unauthorized
use, but
> still using a
> cost-effective codec such as g.723 or g.729 ? 
> 
> [codec_g729a.so] => (Annex A/B (floating point) G.729/PCM16 Codec
> Translator)
> 
>   == G.729 Host-ID:
> 5f:a1:18:82:47:6f:a8:f7:33:4e:7d:77:e8:1d:60:15:53:ec:49:aa
> 
>   == Found license 'G729-700241AB' providing 5 channels
> 
>   == Found total of 5 G.729 licenses
> 
>   == Registered translator 'g729tolin' from format G729A to SLINR,
cost 2
> 
>   == Registered translator 'lintog729' from format SLINR to G729A,
cost 12
> 
> 
> I was hoping by letting it ring out, I would get a voice-mail message, but
> that doesn't work either...
>  
> 
> *CLI> Jul 13 11:29:08 WARNING[98310]: chan_sip.c:2696 process_sdp: No
> compatible codecs!
> 
>     -- Executing Dial("SIP/67.23.212.25-0814f830",
"SIP/334|20") in new
> stack
> 
>     -- Called 334
> 
>     -- SIP/334-26f8 is ringing
> 
>     -- Nobody picked up in 20000 ms
> 
>     -- Executing VoiceMail("SIP/67.23.212.25-0814f830",
"u334") in new stack
> 
>     -- Playing 'vm-theperson' (language 'en')
> 
>   == Spawn extension (default, 4084, 2) exited non-zero on
> 'SIP/67.23.212.25-0814f830'
> 
>  
> I have dropped this question at the asterisk user list some days ago, but
> it's being ignored... (or nobody has the answer)
> 
> Can anybody shed some light on this ?
> 
> Warmest Regards,
> 
> Walter Klomp
> 
> 
> 
> _______________________________________________
> 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
>

>Can anybody shed some light on this ?
>
>Warmest Regards,
>
>  
>
Can you get an Ethereal dump of both legs of the call (phone to * , and 
* to Cisco).  That way you can know for sure what codecs are being 
negotiated.

-- 
Andres
Network Admin
http://www.telesip.net

Perfect, that seemed to do the trick.

I didn't have the codecs specified, so assumed all were enabled, but
didn't
know you have to *explicitly* allow all the codecs individually... So now I
added in sip.conf:

disallow=all                    ; Disallow all codecs
allow=g723.1
allow=g729
allow=gsm
allow=ilbc
allow=alaw                      ; Allow codecs in order of preference
allow=ulaW


And it works now...

Now I only have to figure out why asterisk passes the IP address of the
client as caller-id to my cisco box, instead of the phone number...

Walter.

Date: Fri, 16 Jul 2004 12:38:50 +1000
From: Adam Hart <adam@teragen.com.au>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] G.729 codec doesn't seem to work *even* after
installing the license
Reply-To: asterisk-users@lists.digium.com

try sip debug and see what each side is offering in codecs (make sure yo u
have allow=g729

Walter Klomp wrote:
> Hi,
> 
> I am trying to post this again as I am getting no answers and the 
> support@digium.com bounces...
> 
<snip>