I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833)
and OpenPhone (H.323, G.711).
But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I
press any key on analogue phone connected to ATA, Asterisk shows following message:
-- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack
-- Called 62.213.36.100
-- OH323/L4366 answered SIP/519-3781
1:36.475 LogChanTx:8130bc0 PWLib Assertion fail: Invalid parameter,
file rtp.cxx, line 385, Error=22
<A>bort, <C>ore dump, <I>gnore?
┘and connection becomes one-way style - voice transmits from OpenPhone only.
This problem doesn't appear while calling from OpenPhone to ATA186.
extensions.conf
---------
[general]
static=yes
writeprotect=no
[globals]
[demo]
exten => s,1,Wait,1
exten => s,2,Answer
exten => s,3,Dial(SIP/519,20,Tt)
exten => s,4,Hangup
exten => s,104,Hangup
[default]
include => demo
[extensions]
exten => 100,1,Dial(OH323/xx.xx.xx.xx,xx,Tt)
exten => 100,2,Hangup
exten => 100,102,Hangup
exten => 102,1,Dial(SIP/519,20,Tt)
exten => 102,2,Hangup
exten => 102,102,Hangup
[local-access]
include => extensions
-------------
h323.conf
-----------
[general]
listenAddress=xx.xx.xx.xx,xx
listenPort=1720
connectPort=1720
tcpStart=10000
tcpEnd=20000
udpStart=10000
udpEnd=20000
fastStart=no
h245Tunnelling=no
h245inSetup=no
inBandDTMF=no
silenceSuppression=no
jitterMin=20
jitterMax=100
ipTos=none
outboundMax=10
inboundMax=10
simultaneousMax=10
wrapLibTraceLevel=1
libTraceLevel=0
libTraceFile=stdout
gatekeeper=DISCOVER
gatekeeperTTL=600
userInputMode=RFC2833
amaFlags=default
accountCode=H323
context=voip-h323
[register]
;
alias=asterisk
alias=123
;
; Aliases/prefixes routed in "all-aliases" context.
;
context=all-aliases
alias=ASTERISK
alias=666
;
; Aliases/prefixes routed in "more-aliases" context.
;
context=more-aliases
alias=665
;
; Aliases/prefixes routed in "all-prefixes" context.
;
context=all-prefixes
gwprefix=00
gwprefix=01
;
; Aliases/prefixes routed in "more-stuff" context.
;
context=more-stuff
alias=664
gwprefix=02
;-----------------------------------------
; Specify and configure CODEC related
; options
;-----------------------------------------
[codecs]
codec=G711A
frames=20
;codec=G711U
;frames=20
;codec=GSM0610
;frames=4
codec=G7231
;frames=2
;codec=G729
;frames=2
;codec=G7231
;frames=6
-----------------------
sip.conf
-------------------
[general]
port = 5060 ; Port to bind to
bindaddr = xx.xx.xx.xx,xx ; Address to bind to
context=INVALID
tos=lowdelay
;disallow=all ; Disallow all codecs
;allow=ulaw ; Allow codecs in order of preference
trancfer=yes
threewaycalling=yes
[519]
type=friend
host=xx.xx.xx.xx,xx
context=local-access
reinvite=no
canreinvite=no
dtmfmode=RFC2833
qualify=300
callerid="ATA186" <519>
;mailbox=21
nat=no
[520]
type=friend
host=xx.xx.xx.xx,xx
context=local-access
reinvite=no
canreinvite=no
;dtmfmode=inband
qualify=300
callerid="x-lite" <520>
;mailbox=21
nat=yes
-----------
Pavel Riko
_______________________________________________
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
