G729 codec eyebeam + goautodial

[trandos]

hi guys,

i have install from iso:

GoAutoDial CE 2.0
Vicidial 2.2.1
Asterisk 1.4.27.1-vici

VERSION: 2.2.1-260 BUILD: 100527-2211


i have include the g729 codec:

# codec_g729-ast14-gcc4-glibc-pentium4.so

show translation:

g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726 g722
g723 - 3 2 2 2 2 1 3 5 - - 2 -
gsm 9 - 2 2 2 2 1 3 5 - - 2 -
ulaw 9 3 - 1 2 2 1 3 5 - - 2 -
alaw 9 3 1 - 2 2 1 3 5 - - 2 -
g726aal2 9 3 2 2 - 2 1 3 5 - - 2 -
adpcm 9 3 2 2 2 - 1 3 5 - - 2 -
slin 8 2 1 1 1 1 - 2 4 - - 1 -
lpc10 10 4 3 3 3 3 2 - 6 - - 3 -
g729 9 3 2 2 2 2 1 3 - - - 2 -
speex - - - - - - - - - - - - -
ilbc - - - - - - - - - - - - -
g726 9 3 2 2 2 2 1 3 5 - - - -
g722 - - - - - - - - - - - - -

i use voip for my calls. my provider tells me that the incoming calls on their server are g729.

i use eyebeam as softphone. if there is only g729 enabled i got the message: 499 not acceptable here

and asterisk show;

chan_sip.c:5653 process_sdp: No compatible codecs, not accepting this offer!

where iam wrong?? its the eyebeam or my asterisk codec???

thanks for any help

[williamconley]

1)

You posted both your installer AND your vicidial version with build. Way cool.

2)

use sip debug to find the exact issue. It may be that you have not enabled g729 for the agents or there may be a g729 version issue (although i've never bumped into this before, it is technically possible as there IS more than one flavor of g729). It is also possible you used up all your g729 licenses before the agent tried to log in.

3)

using g729 for the agents is ONLY helpful if the agents are NOT on the same subnet as the server. if they are on the same subnet, there is no reason to use anything other than ulaw for the connection to the agent. everything will be converted to slin for the conference call anyway, so you'll not be using "pass-through" without transcoding ... you'll just be adding another layer of transcoding.

[trandos]

after many many times i post it

what did you meen with enabled for the agents ?? where i can do that ??


i think the version i used are wrong hmmm...

i have download the free g729 lizence on (http://asterisk.hosting.lv/) sorry for posting its not advertising.


my sip debug:




<------------->
[Dec 20 01:56:30] --- (7 headers 0 lines) ---
[Dec 20 01:56:30]
<--- SIP read from 109.242.183.159:2528 --->
INVITE sip:812105549957@XXXXXXXXXX SIP/2.0
Via: SIP/2.0/UDP 192.168.1.59:2528;branch=z9hG4bK-d8754z-a80ee44228214461-1---d8754z-;rport
Max-Forwards: 70
Contact: <sip:cc100@109.242.183.159:25XXXXXXXXXXX28>
To: "812105549957"<sip:812105549957@>
From: "cc100"<sip:cc100@XXXXXXXXX>;tag=0d742113
Call-ID: ZGJjODk3NmNiOGI4N2ZiMjgxMzM1OWI4MzNjOTRjNzY.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
Proxy-Authorization: Digest username="cc100",realm="asterisk",nonce="10118e93",uri="sip:812105549957@XXXXXXXXXXX",response="7c308ebfc5b617384cb4e54d055cd7dd",algorithm=MD5
User-Agent: eyeBeam release 1100z stamp 47739
Content-Length: 486

v=0
o=- 6 2 IN IP4 192.168.1.59
s=CounterPath eyeBeam 1.5
c=IN IP4 192.168.1.59
t=0 0
m=audio 25374 RTP/AVP 18 101
a=alt:1 4 : zX9Lb8bu 7QSVTosV 169.254.16.109 25374
a=alt:2 3 : KHDwWzE4 79N79bgC 192.168.138.1 25374
a=alt:3 2 : qdZOTDQE aFoixzcN 192.168.70.1 25374
a=alt:4 1 : tQTsIkRE gbfp128r 192.168.1.59 25374
a=fmtp:18 annexb=yes
a=fmtp:101 0-15
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
a=x-rtp-session-id:164BBFEF96F84BEB8203B7BB26AEED02

<------------->
[Dec 20 01:56:30] --- (13 headers 16 lines) ---
[Dec 20 01:56:30] Sending to 109.242.183.159 : 2528 (NAT)
[Dec 20 01:56:30] Using INVITE request as basis request - ZGJjODk3NmNiOGI4N2ZiMjgxMzM1OWI4MzNjOTRjNzY.
[Dec 20 01:56:30] Found user 'cc100'
[Dec 20 01:56:30] Found RTP audio format 18
[Dec 20 01:56:30] Found RTP audio format 101
[Dec 20 01:56:30] Found audio description format G729 for ID 18
[Dec 20 01:56:30] Found audio description format telephone-event for ID 101
[Dec 20 01:56:30] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing)
[Dec 20 01:56:30] Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
[Dec 20 01:56:30] NOTICE[2622]: chan_sip.c:5653 process_sdp: No compatible codecs, not accepting this offer!
[Dec 20 01:56:30]
<--- Reliably Transmitting (NAT) to 109.242.183.159:2528 --->
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/UDP 192.168.1.59:2528;branch=z9hG4bK-d8754z-a80ee44228214461-1---d8754z-;received=109.242.183.159;rport=2528
From: "cc100"<sip:cc100@XXXXXXXXXXX>;tag=0d742113
To: "812105549957"<sip:812105549957@XXXXXXXXXXX>;tag=as7065450f
Call-ID: ZGJjODk3NmNiOGI4N2ZiMjgxMzM1OWI4MzNjOTRjNzY.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0

105 s)


my server is online and in my office 15 agents want to connect to the server and my internet connection are full. other 35 agents are on homeoffice. but for the agents in the same office the connection isnt enough. so i want to reduce the voip packeges.

have edit my ip adress

[williamconley]

[Dec 20 01:56:30] Capabilities: us - 0x6 (gsm|ulaw), peer - audio=0x100 (g729)/video=0x0 (nothing), combined - 0x0 (nothing)

Combined = Nothing (resulting from: server=gsm or ulaw; peer(agent)=g729)

So you will need to put allow=g729 in the agents sip entry. Look up how to modify the sip template (or how to "use" the sip templates). Not hard

[trandos]

where i can do this ???

have i put the allow=g729 in the SIP_generic ??

or to modify in the sip.conf ?

in my carrier i have:
disallow=all
allow=g729

right now.

[trandos]

ok i have it ! thanks very much !!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!

[williamconley]

no related to the carrier.

do you have (have you read) the manual?

____-

ah!

[trandos]

not need the manual if william is THERE !

[neeraj_tiwary]

were to find how to "use" the sip templates

[williamconley]

were to find how to "use" the sip templates

Do not hijack threads for your own nefarious needs. Create your own post and ask your own question.