[Asterisk-Dev] Problem with IAX

Sergio Serrano Revuelto sergio.serrano at avanzada7.com
Tue Apr 1 11:40:47 MST 2003


Problem solved. It was a problem with codec_gsm.

Thanks

srsergio



-----Mensaje original-----
De: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com] En nombre de Sergio Serrano
Revuelto
Enviado el: martes, 01 de abril de 2003 19:33
Para: asterisk-dev at lists.digium.com
Asunto: [Asterisk-Dev] Problem with IAX


Could someone help me?
I have a problem with IAX.
I'm usinge the next extension to call:

exten => 731,1,Dial,IAX/nbx1/712 at voip-h323

In destinatination I obtain the next error:

  -- Accepting unauthenticated call from 192.168.0.202, requested format
= 2, actual format = 2
  == Accepting call on 'IAX[nbx at nbx]/1' (11)
    -- Executing Wait("IAX[nbx at nbx]/1", "1") in new stack
    -- Executing Dial("IAX[nbx at nbx]/1", "oh323/12|20|m") in new stack
WARNING[18451]: File app_dial.c, Line 419 (dial_exec): Asterisk music on
hold ok
WARNING[18451]: File channel.c, Line 1373 (ast_request): No translator
path exists for channel type oh323 (native 64) to 2
NOTICE[18451]: File app_dial.c, Line 440 (dial_exec): Unable to create
channel of type 'oh323'
  == Everyone is busy at this time
WARNING[18451]: File cdr_mysql.c, Line 64 (mysql_log): cdr_mysql: SQL
command as follows:  INSERT INTO cdr
(calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,dura
tion,billsec,disposition,amaflags,accountcode) VALUES ('2003-04-01
20:37:18','11','11','712','voip-h323',
'IAX[nbx at nbx]/1','','Dial','oh323/12|20|m',8,0,2,3,'')
    -- Hungup 'IAX[nbx at nbx]/1'


Configuration Files are the next:


[general]
port=5036
bindaddr=192.168.0.204
bandwidth=low

disallow=g723.1
disallow=lpc10                  ; Icky sound quality...  Mr. Roboto.
disallow=g711a
disallow=g711u
allow=gsm                       ; Always allow GSM, it's cool :)

jitterbuffer=yes
dropcount=3
maxjitterbuffer=500
maxexccessbuffer=100

tos=lowdelay

;register => nbx1 at 192.168.0.202

[nbx]
type=friend
context=voip-h323
;username=nbx
;auth=plaintext
;secret=supersecret
;host=dynamic
;sendani=no
;port=5036
host=192.168.0.202




AND THE OTHER


[general]
port=5036
bindaddr=192.168.0.202
bandwidth=low

disallow=g723.1         ; Hm...  Proprietary, don't use it...
disallow=lpc10                  ; Icky sound quality...  Mr. Roboto.
disallow=g711a
disallow=g711u
allow=gsm                       ; Always allow GSM, it's cool :)

jitterbuffer=yes
dropcount=3
maxjitterbuffer=500
maxexccessbuffer=100

tos=lowdelay

;register=>nbx at 192.168.0.204

[nbx1]
type=friend
context=voip-h323
;username=nbx1
;auth=plaintext
;secret=supersecret
;host=dynamic
;sendani=no
;port=5036
;mask=255.255.255.0
host=192.168.0.204



Thanks in advance
srsergio

_______________________________________________
Asterisk-Dev mailing list
Asterisk-Dev at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-dev




More information about the asterisk-dev mailing list