[Asterisk-Users] SIP->h323 problem DTMF

[Michael Manousos]

Try with a different userInputMode in oh323.conf.

Michael.


rr80 wrote:
> 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