[Asterisk-Users] Oh323 and userInputMode in conf being ignored.

[David Hindmarsh]

Hi Guys,

I am using oh323 to connect to a provider.

No matter what I set in the oh323.conf for userinputmode it always uses
q931.

How do I set it to RFC2833.

Dave

-----Original Message-----
From: asterisk-users-admin at lists.digium.com
[mailto:asterisk-users-admin at lists.digium.com]On Behalf Of Michael Manousos
Sent: Wednesday, 14 April 2004 7:24 PM
To: asterisk-users at lists.digium.com
Subject: Re: [Asterisk-Users] SIP->h323 problem DTMF



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

_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users

---
Incoming mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.656 / Virus Database: 421 - Release Date: 9/04/2004

---
Outgoing mail is certified Virus Free.
Checked by AVG anti-virus system (http://www.grisoft.com).
Version: 6.0.714 / Virus Database: 470 - Release Date: 2/07/2004