After upgrade my trixbox up to 2.4 my g729 stop to work
But on *CLI> show translation system show:
________g729
g723 -
gsm 4
ulaw 4
alaw 4
g726aal2 4
adpcm 4
slin 3
lpc10 4
g729 -
speex 5
ilbc 5
g726 4
g722 -
This shows g729 is loaded into the system
or *CLI>show modules:
codec_g729.so g729 Coder/Decoder
Why g729 doesn't work?
please help
Can you be a little more detailed than "stopped to work"?
What exactly happens? Can we get a cli output of a call to a g729 enabled device?
WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.
192.168.0.1 device
192.168.0.10 trixbox
INVITE sip:3001@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5062;branch=z9hG4bK32e648e246640235
From:
To:
Contact:
Supported: replaces, timer, path
Call-ID: 23d74620389546a0@192.168.0.1
CSeq: 20976 INVITE
User-Agent: Grandstream GXV3000 1.0.1.20
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 395
v=0
o=3001 8000 8000 IN IP4 192.168.0.1
s=SIP Call
c=IN IP4 192.168.0.1
t=0 0
m=audio 5004 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 5006 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801E; packetization-mode=1; sprop-parameter-sets=J0KAFJWgUH5A,KM4BryA=
a=framerate:15
--- (13 headers 16 lines) ---
Sending to 192.168.0.1 : 5062 (no NAT)
Using INVITE request as basis request - 23d74620389546a0@192.168.0.1
Arax*CLI>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.1:5062;branch=z9hG4bK32e648e246640235;received=192.168.0.1
From:
To:
Call-ID: 23d74620389546a0@192.168.0.1
CSeq: 20976 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="5ee31cd3"
Content-Length: 0
Scheduling destruction of SIP dialog '23d74620389546a0@192.168.0.1' in 32000 ms (Method: INVITE)
Found user '3001'
Arax*CLI>
ACK sip:3001@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5062;branch=z9hG4bK32e648e246640235
From:
To:
Contact:
Call-ID: 23d74620389546a0@192.168.0.1
CSeq: 20976 ACK
User-Agent: Grandstream GXV3000 1.0.1.20
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
--- (11 headers 0 lines) ---
Arax*CLI>
INVITE sip:3001@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5062;branch=z9hG4bK192636d576920e56
From:
To:
Contact:
Supported: replaces, timer, path
Proxy-Authorization: Digest username="3001", realm="asterisk", algorithm=MD5, uri="sip:3001@192.168.0.10", nonce="5ee31cd3", response="0db97a52c011960236d5773fc9f6e223"
Call-ID: 23d74620389546a0@192.168.0.1
CSeq: 20977 INVITE
User-Agent: Grandstream GXV3000 1.0.1.20
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 395
v=0
o=3001 8000 8001 IN IP4 192.168.0.1
s=SIP Call
c=IN IP4 192.168.0.1
t=0 0
m=audio 5004 RTP/AVP 18 101
a=sendrecv
a=rtpmap:18 G729/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-11
m=video 5006 RTP/AVP 99
a=sendrecv
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42801E; packetization-mode=1; sprop-parameter-sets=J0KAFJWgUH5A,KM4BryA=
a=framerate:15
--- (14 headers 16 lines) ---
Sending to 192.168.0.1 : 5062 (no NAT)
Using INVITE request as basis request - 23d74620389546a0@192.168.0.1
Found user '3001'
Found RTP audio format 18
Found RTP audio format 101
Found RTP video format 99
Peer audio RTP is at port 192.168.0.1:5004
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Found unknown media description format H264 for ID 99
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Arax*CLI>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.0.1:5062;branch=z9hG4bK192636d576920e56;received=192.168.0.1
From:
To:
Call-ID: 23d74620389546a0@192.168.0.1
CSeq: 20977 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
Scheduling destruction of SIP dialog '23d74620389546a0@192.168.0.1' in 32000 ms (Method: INVITE)
Arax*CLI>
ACK sip:3001@192.168.0.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.1:5062;branch=z9hG4bK192636d576920e56
From:
To:
Contact:
Proxy-Authorization: Digest username="3001", realm="asterisk", algorithm=MD5, uri="sip:3001@192.168.0.10", nonce="5ee31cd3", response="0db97a52c011960236d5773fc9f6e223"
Call-ID: 23d74620389546a0@192.168.0.1
CSeq: 20977 ACK
User-Agent: Grandstream GXV3000 1.0.1.20
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
What does
show g729
say?
WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.
show g729
No such command 'show g729'
Ok, most likely you need to re-install your g729 modules. It's just like you did the first time, but you'll probably have to go re-download them and re-license them. The licenses are keyed to the NIC, so as long as that hasn't changed you should be OK. If it has, you'll have to call Digium to get it straightened out.
WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.
Trixbox 2.4 contains a newer version of asterisk -- version 1.4. So if you upgraded from a previous version of trixbox (2.0 or 2.2) to 2.4 you will need to download new g729 codec modules for asterisk 1.4.
--
Tom -- sirthomas@gmail.com -- +1.320.310.0778 (enum enabled)
I know
My previous asterisk version is 1.4
So , and codec modules for 1.4
Command show translation show g729 normaly loaded...
The important bit is that it looks like it's not licensed. You can try re-applying the licenses using the Digium tool.
WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.
I'm using free version g729 http://www.trixbox.org/forums/trixbox-forums/help/adding-g-729-fr...
I can't help you with that, sorry. There's really no "free" version, only one that can be used in an academic setting, or in an area where software patents aren't enforced.
All I can say is try re-doing the setup from scratch. Beyond that -- the licenses from Digium only cost $10 each.
Good luck.
WARNING: I no longer actively participate in these forums. My thoughts on trixbox in a nutshell: http://www.youtube.com/watch?v=q4xBMkWu1pE Use AsteriskNOW instead.
Copyright © 2011 Fonality
Fonality and trixbox are trademarks of Fonality. Trademark Policy. Privacy Policy.
Asterisk is a trademark of Digium, Inc. Fonality and trixbox are not affiliated with, nor endorsed by Digium, Inc.
Member Since:
2008-01-24