[Asterisk-Users] SIP->h323 problem DTMF

[rr80]

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