who can tell me how to do TDM over enthernet ?
pc a connect pc b only use TDM card?
thank you
John.
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: 20041223 11:47
: asterisk-users@lists.digium.com
: Asterisk-Users Digest, Vol 5, Issue 336
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than Re: Contents of Asterisk-Users digest...
Today's Topics:
1. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave)
2. Re: Can't Receive/Send Calls (Norman Zhang)
3. RE: Zaptel/Zapata config from T410p to Brooktrout T1
(Jason Kawakami)
4. Re: Still unable to use g729 codec... please HELP
(Kristian Kielhofner)
5. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave)
6. Re: Still unable to use g729 codec... please HELP
(Kristian Kielhofner)
7. Re: hint extension and Snom phones - CVS or stable? (Karl Brose)
8. WARNING Maximum retries exceeded on call for seqno102 (Kevin)
9. Re: Re: 'I'nvalid extension handling problems,even with
workaround ([EMAIL PROTECTED])
10. Re: Still unable to use g729 codec... please HELP (Rodolfo Grave)
11. Re: Still unable to use g729 codec... please HELP
(Kristian Kielhofner)
12. RE: polycom and cdp (Richard)
13. Re: RE: Zaptel/Zapata config from T410p toBrooktroutT1 (jbebeau)
--
Message: 1
Date: Thu, 23 Dec 2004 03:32:22 +0100
From: Rodolfo Grave [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Still unable to use g729 codec... please
HELP
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Yeap... :(
Kristian Kielhofner wrote:
Rodolfo Grave wrote:
Hi.
Has anyone accomplished to use the g729 codec? I have the license
installed, and I have tried with X-Pro and a Grandstream Budgetone
configured to use g729 only. This is what I get from Asterisk:
Dec 23 02:38:07 WARNING[21176]: chan_sip.c:2764 process_sdp: No
compatible codecs!
Dec 23 02:38:09 NOTICE[21176]: chan_sip.c:7295 handle_request: Unable
to create/find channel
My sip.conf contains:
disallow=all
allow=g729
for both devices!!
I dont know what to do, I need to use the g729 codec. Please help.
If I enable GSM in the device and add allow=gsm everything works, so
it is a codec problem.
The license and the codec seems to be correctly installed:
*CLI show g729
0/0 encoders/decoders of 1 licensed channels are currently in use
*CLI
Please, help me.
RODOLFO
Rodolfo,
I assume that you tried the obvious stuff, re-reading the docs,
running the register program, getting the proper g729 binary from
Digium, etc?
--
Kristian Kielhofner
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Message: 2
Date: Wed, 22 Dec 2004 18:40:03 -0800
From: Norman Zhang [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Can't Receive/Send Calls
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
With this trimmed down versions of sip.conf and extensions.conf. I can
now receive calls from outside. But audio will not traverse out to the
internet. I can hear the caller no problem. Also I still cannot
dial-out. Any ideas?
Regards,
Norman Zhang
I removed all the PSTN stuffs. As I'm only trying to make SIP work.
Would someone kindly give me a few pointers?
[general]
disallow=all
allow=ulaw
port=5060
bindaddr=0.0.0.0
externip=207.34.136.26
localnet=192.168.22.0/255.255.255.0
context=inbound-sip
maxexpirey=180
defaultexpirey=160
tos=reliability
srvlookup=yes
register = 533990:[EMAIL PROTECTED]/533990
[fwd]
type=friend
secret=normanzhang
username=533990
fromuser=533990
fromdomain=fwd.pulver.com
host=fwd.pulver.com
dtmfmode=inband
nat=yes
canreinvite=no
[101]
disallow=all
allow=ulaw
type=friend
host=dynamic
dtmfmode=inband
username=101
secret=testing123
context=home
nat=no
; extensions.conf
[general]
static=yes
writeprotect=no
[globals]
MAINPHONE=SIP/101
FWDUSERID=533990
FWDUSERNAME=Norman Zhang
FWDPREFIX=*
; Macros
