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: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Michael Manousos Sent: Wednesday, 14 April 2004 7:24 PM To: [EMAIL PROTECTED] 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 [EMAIL PROTECTED] 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 _______________________________________________ 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
