search for: inbanddtmf

see below I tried to call 98013356 from the following phone (from mgcp.conf) [iptlf03] host = 192.168.33.3 context = default inbanddtmf = 1 callerid = 22545062 line => aaln/1 Console output: == Spawn extension (capiring, 9988001133335566, 1) exited non-zero on 'MGCP/aaln/1@iptlf03-1' -- MGCP mgcp_hangup(MGCP/aaln/1@iptlf03-1) on aaln/1@iptlf03 -- Delete connection 4 aaln/1@iptlf03-1 with new mode: recvonly o...

What models? Jeff Noxon (jeff-asterisk at planetfall.com) wrote*: > >I just purchased a bunch of Nortel Meridian POTS phones that support >intercom on the 3rd pair. I intend to get it working with Asterisk. >The phones support MWI, have a 3-line display, callerID, call waiting >callerID, 2 lines...very nice. > >On Thu, Feb 27, 2003 at 01:07:19AM -1000, James H. Thompson

I am getting (with older image): RFC3389 support incomplete. Turn off on client if possible How do I turn that off for the DG104s? Or if I can't how do I tweak asterisk? I see posts about ATA-186's having an audiomode, but the closet I came to was inbanddtmf. I tried =0 and =1, no effect. Thanks! -- Zot O'Connor <zot@zotconsulting.com> White Knight Hackers, Inc.

...00.10szxn-xxxx e.164 - 22xx2912 ID - HMA0200.10szxn-xxxx e.164 - 22xx2913 Here is my oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 gatekeeper=AVS@210.21.118.XXX gatekeeperTTL=600 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout userInputMode=TONE amaFlags=default accountCode=H323 language=en context=voip-h323 [register] alias=ASTERISK [codecs] codec=G711A frames=20 [22xx2912]...

hello i am using asterisk-oh323-0.7.1. i want to convert sip call to h323 (h323 sjphone or h323 proxy). what could be the best way for this. i am successfull in converting h323->sip by using asterisk as gateway. help required on sip->h323. kamran __________________________________ Celebrate Yahoo!'s 10th Birthday! Yahoo! Netrospective: 100 Moments of the Web

...-Server. All works fine, i just wanna know if there's a better way to do it? Is there anything wrong with my Config? OH323.conf [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=8000 udpEnd=8005 fastStart=no h245Tunnelling=no h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=500 ipTos=reliability outboundMax=10 inboundMax=10 simultaneousMax=10 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 amaFlags=default accountCode=H323 [223];OpenPhone type=friend de...

Hi group! Does DTMF inband work with oh323 channel driver ver. 0.6.7? If yes, how to enable it, make it work? I have tried with "inBandDTMF=yes" in general context of oh323.conf, but I get this message when I * is starting. [chan_oh323.so] => (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found Apr 12 13:38:17 NOTICE[3622]: cha...

...Creating new gatekeeper. Ouch ... error while writing audio data: : Broken pipe Segmentation fault ------------ oh323.conf ---------------------------- [general] listenAddress=myip listenPort=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=5 libTraceFile=/var/log/asterisk/oh323.log gatekeeper=mygnugk ;gatekeeperPassword=secret gatekeeperTTL=600 userInputMode=TONE amaFlags=default accountCode...

...39;re expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=30000 udpStart=20001 udpEnd=30000 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk 1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 libopenh323-1.15.3c2 1.15.3-4 --...

....192.252". Sep 29 00:07:32 ERROR[-151090528]: chan_h323.c:1987 load_module: Gatekeeper registration failed. my h323.conf is: [general] listenAddress=0.0.0.0 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=200.69.192.252 gatekeeperPassword = secret name=xxxxxxxxxx secret=xxxxxxxxxx AllowGKRouted = yes gatekeeperTTL=100 user...

...; 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=...

Hi everyone, Could somebody enlighten me on this one? I have configured my asterisk to run on oh323 using codec g729. Incoming calls are working okay. But the thing I want to work is say pressing some options, say dial 1 to go to voicemail or dial a certain number to dial a specific extension. I have a config for this and tried calling from a normal PSTN and is working. But i just can't seem

...tp.conf" ; udpStart=10000 udpEnd=20000 ; ; Enable fast start (yes,no). ; fastStart=yes ; ; Enable H.245 tunnelling (yes,no). ; h245Tunnelling=no ; ; Enable early H.245 messages in call SETUP message. ; h245inSetup=yes ; ; Enable in-band-DTMF detection. ; (Note: Netmeeting uses in-band DTMFs) ; inBandDTMF=no ; ; Enable silence suppression. ; silenceSuppression=no ; ; Set jitter buffer (in milliseconds, 20...10000). ; jitterMin=20 jitterMax=100 ; ; Set IP Type-of-Service byte for RTP channels. ; Valid values for this option are: ; lowdelay, throughput, reliability, mincost, none ; ipTos=lowdelay ;...

...e. Is there any way that i can send the called Number from CCM. I think it is sent but in a different way. My oh323.conf is: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=no h245Tunnelling=yes h245inSetup=yes inBandDTMF=no silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 wrapLibTraceLevel=1 libTraceLevel=1 libTraceFile=stdout gatekeeper=DISABLE userInputMode=TONE amaFlags=default accountCode=H323 context=voip-h323 [register] alias=3001 [codecs] codec=G711A frames=20 Hop...

...23 gatekeeper exten => _011.,1,SetVar(OH323_OUTCODEC=ulaw) exten => _011.,2,Dial(OH323/00${EXTEN}) Here is the oh323.conf: [general] listenAddress=0.0.0.0 listenPort=1720 connectPort=1720 tcpStart=10000 tcpEnd=20000 udpStart=10000 udpEnd=20000 fastStart=yes h245Tunnelling=yes h245inSetup=no inBandDTMF=yes silenceSuppression=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=100 inboundMax=100 simultaneousMax=100 ;bandwidthLimit=1024 wrapLibTraceLevel=1 libTraceLevel=0 libTraceFile=stdout gatekeeper=192.168.254.2 ;gatekeeperPassword= gatekeeperTTL=600 userInputMode=TONE amaFlags=default account...

Hello there. Is there somebody with this connection working? I can't seem to make this work at all. Could someone please share some .conf files? Cheers, Vedran. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050818/71480bd9/attachment.htm

I've this setup : CiscoAta186 -> asterisk with oh323 chan -> gsmgateway dtmf doesn't work, tryed inband, with g711a and g729 codecs CiscoAta186 -> gsmgateway works, even with g729, so it seems the problem is in * oh323.conf has inBandDTMF=yes, what else may I need to tweak ?

...39;re expecting us to generate the ringback tone. Does anybody know how to configure this in Asterisk? The relevant settings in oh323.conf are: [general] listenAddress=0.0.0.0 listenPort=1720 tcpStart=20001 tcpEnd=30000 udpStart=20001 udpEnd=30000 fastStart=yes h245Tunnelling=yes h245inSetup=yes inBandDTMF=no jitterMin=20 jitterMax=100 ipTos=none outboundMax=10 inboundMax=10 simultaneousMax=10 bandwidthLimit=2000 gatekeeper=DISABLE gatekeeperTTL=600 userInputMode=RFC2833 The package versions I'm using are: asterisk 1.0.9.dfsg-5 asterisk-oh323 0.6.6pre3-4 libopenh323-1.15.3c2 1.15.3-4 --...

...g the module [chan_oh323.so] => (InAccess Networks OpenH323 Channel Driver) == Parsing '/etc/asterisk/rtp.conf': Found == Parsing '/etc/asterisk/oh323.conf': Found Apr 5 14:01:04 NOTICE[4161]: chan_oh323.c:4821 reload_config: Ignoring unknown H.323 [general] keyword 'inBandDTMF', line 48. [1]WrapH323EndPoint::WrapH323EndPoint: Compile-time libraries OpenH323 v1.15.3, PWlib v1.8.7 Apr 5 14:01:04 ERROR[4161]: chan_oh323.c:5353 load_module: H.323 listener creation failed. Apr 5 14:01:04 WARNING[4161]: loader.c:414 __load_resource: chan_oh323.so: load_module failed, re...

Can any body give me an example how to configure h323 in Asterisk. Which files do I need to configure? just extensions.conf and h323.conf ? Thanks, Patricio _________________________________________________________________ Descubre la descarga digital con MSN Music. M?s de un mill?n de canciones. http://music.msn.es/