[Asterisk-Users] problem with g729 and CME-Asterisk

Andrea Riela ml at nesys.it
Wed Nov 9 10:01:01 MST 2005


-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1


On Nov 9, 2005, at 5:18 PM, Greg Oliver wrote:

> Post up your dial-peer 500 config as well.  It is doing codec 0x2
> (g.711Alaw) from the get go.
>
> Also post relevant config for the phone from asterisk and dialplan 
> entry
> used.
>

the call flows are:

[ISDN in only] --> ntte [CME]

[VOIP in] --> 5600 [asterisk] --> 601 [CME] (codec g711a)*

[VOIP out] <-- [asterisk] <-- CME (codec g729 if possible)

* multiple sip UA are registered with forwarding to 5600 --> 601 on CME
   maybe that's not a "pass-thru" solution, that is maybe I could'n use 
g729 without license, isn't it?

Cisco (only voip out):

!
voice class codec 1
  codec preference 1 g729r8
  codec preference 2 g711alaw
  codec preference 3 g711ulaw
!
dial-peer voice 500 voip
  description ITA through Messagenet
  destination-pattern .T
  voice-class codec 1
  session protocol sipv2
  session target ipv4:192.168.17.10
  dtmf-relay rtp-nte
  no vad
!
ephone-dn  3
  number 603 secondary xxxxxx no-reg
  label Home
  call-forward noan xxxx timeout 30
!

Asterisk:

sip.conf
- --------

[general]
context=cme-pbx
language=it
realm=sip.nesys.it
port=5060
bindaddr=192.168.17.10
srvlookup=yes
useragent=Nesys Asterisk PBX
disallow=all
allow=g729
allow=alaw
allow=ulaw
tos=0xb8
nat=yes
register => xxxxxxx:yyyyyyy at sip.messagenet.it:5061/5600
...

[5600]
type=friend
host=192.168.17.10
dtmfmode=rfc2833
canreinvite=yes
context=myphones
qualify=yes

[cme-pbx]
type=peer
canreinvite=no
host=192.168.17.1
qualify=yes

[60x]
type=friend
language=it
host=dynamic
dtmfmode=rfc2833
canreinvite=yes
mailbox=60x at default
context=myphones
qualify=yes

[messagenet-MI-out]
context=cme-pbx
type=friend
language=it
username=xxxxxxx
fromuser=xxxxxxx
fromdomain=sip.messagenet.it
secret=yyyyyyy
host=sip.messagenet.it
port=5061
nat=yes
canreinvite=no
insecure=very
qualify=yes

extensions.conf
- ---------------

[myphones]
include => cme-pbx
include => messagenet-ITA-out

[messagenet-ITA-out]
exten => _X.,1,Dial(SIP/${EXTEN}@messagenet-MI-out,30,r)
exten => _X.,2,Playback(invalid)
exten => _X.,3,Hangup

[cme-pbx]
exten => _6XX,1,Dial(SIP/${EXTEN}@cme-pbx)
exten => _6XX,2,Playback(invalid)
exten => _6XX,3,Hangup
exten => 5600,1,Dial(SIP/601,45)
include => messagenet-ITA-out

I know, that's a complicated implementation, the confs will be better ;)

Thanks for your support
Regards
Andrea
-----BEGIN PGP SIGNATURE-----
Version: GnuPG v1.4.1 (Darwin)

iD8DBQFDcitSMakHrsrHP9wRAuf3AJ9q1TAQcngi5h+rlBzviWs5/GsjugCfd+fG
J87utc0S2yvKZT27w/cn4Dc=
=RJnY
-----END PGP SIGNATURE-----




More information about the asterisk-users mailing list