search for: rfc2833compensate

...Help would be greatly appreciated. And by the way, my Asterisk box is talking to a Level 3 SIP gateway with the following configuration: [bandwidth] type=peer host=x.x.x.x dtmfmode=rfc2833 canreinvite=no disallow=all allow=ulaw context=incoming reinvite=no canreinvite=no nat=no directrtpsetup=yes rfc2833compensate=yes rtpkeepalive=60 Thanks in advance! - Justin Tunney

...rompt stops playing. SIP phones work correctly. I've trird everything I found searching the net. I've tried all the dtmfmode. I'm using 1.4.26 Currently my vitelity sip account is setup: insecure=very canreinvite=no host=xx.xx.xx.xx qualify=yes dtmfmode=rfc2833 disallow=all allow=ulaw rfc2833compensate=yes I need to trouble shoot this furthur. I read I can enable rtp debug <IP> but I can't find any output. Thanks, Bart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091005/735d97cd/attachme...

....net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gsm If you need to see more of the setup info I can provide. Thanks Brent

...ht away, then the '9' comes in and gets queued, but it doesn't start playing until five seconds later and it plays for six seconds. Then the last '5' is played. The DTMF is coming in as only 'end' packets and we can't change that. For this reason we have turned on rfc2833compensate. Using Asterisk 1.4.11. Any ideas? asteriskpri04*CLI> < Protocol Discriminator: Q.931 (8) len=5 < Call Ref: len= 2 (reference 499/0x1F3) (Terminator) < Message type: CONNECT (7) q931.c:3371 q931_receive: call 33267 on channel 3 enters state 10 (Active) > Protocol Discriminator:...

...es rtsavesysname=yes rtupdate=yes rtautoclear=yes ignoreregexpire=no [xxxxx] type=peer host=xxxxxx.xxxxxxx.com secret=XxXxXxXxXxX call-limit=128 defaultuser=00XXXXXXXXXXX fromuser=00XXXXXXXXXXX disallow=all allow=ulaw nat=yes context=trunk qualify=50 insecure=invite restrictcid=no dtmfmode=rfc2833 rfc2833compensate=yes canreinvite=no --------------- my extensions.conf ------------------ [general] static=yes writeprotect=no autofallthrough=yes extenpatternmatchnew=yes clearglobalvars=no [vm-xxx] exten => s,1,Answer exten => s,n,Set(TIMEOUT(response)=3) exten => s,n,Set(TIMEOUT(digit)=5) exten =>...

...llback: sip.conf: register => 1777286XXXX:XXXXXXX at callcentric.com/1862772XXXX ... [callcentric] type=peer context=from-callcentric host=callcentric.com username=1777286XXXX secret=XXXXXXX fromuser=1777286XXXX fromdomain=callcentric.com disallow=all allow=alaw dtmfmode=inband canreinvite=no ;rfc2833compensate=yes insecure=very extensions.conf: NOTE: 1862772XXXX is a real phone # I have in my callcentric account [from-callcentric] exten => 1862772XXXX,1,NoOp(callcentric callback to ${CALLERID(num)) exten => 1862772XXXX,2,Wait(1) exten => 1862772XXXX,3,system(cp /var/spool/asterisk/skelett.c...

...bility but gets past the issue. I am considering upgrading the box to 1.6 as the working one is 1.6 The other box is a digium AA50 appliance so I can't do much with it, other than find the right settings. I have on the first one relaxdtmf=yes - relates to old issues too as far as I can tell rfc2833compensate=yes - this only appears to matter for inbound I'm not sure these do anything useful From what I can tell it could be the toneduration, but don't know what it should be, and while technically its probably the IVR being fussy that doesn't help me and I want to see why one system works a...

...promiscredir: NULL useclientcode: NULL accountcode: NULL setvar: NULL callerid: NULL amaflags: NULL callcounter: NULL busylevel: NULL allowoverlap: NULL allowsubscribe: NULL videosupport: NULL maxcallbitrate: NULL rfc2833compensate: NULL mailbox: NULL session-timers: NULL session-expires: NULL session-minse: NULL session-refresher: NULL t38pt_usertpsource: NULL regexten: NULL fromdomain: testers.com fromuser: 660 qualify: NULL defaultip: NULL rtpt...

....net username = user name secret = password trunkname = via:talk - galvatron ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = user name authuser = user name insecure = port,invite dtmf = rfc2833 dtmfmode = rfc2833 relaxdtmf = yes rfc2833compensate = yes port = 5060 canreinvite = no fromdomain = galvatron.vtnoc.net disallow = all allow = ulaw,gsm If you need to see more of the setup info I can provide. Thanks Brent

Asterisk intermittently does not send audio back to the callers in the meetme conference bridge. If the caller hangs up and calls back sometimes the audio will work and sometimes it does not. We have taken packet captures and reviewed the SIP and SDP, both are correct and you can actually hear the audio being transmitted from the callers to the conference bridge but no audio is sent back to the

...ulair handset [235] type=friend username = 235 callerid="R Vermeeren mobiel" <235> host = dynamic secret = 235 context = default qualify = yes login = 235 callgroup = 3 pickupgroup = 3 disallow = all allow = alaw call-limit = 6 default section of sip.conf [general] dtmfmode=rfc2833 rfc2833compensate=yes notifyringing=yes context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) ; bindpor...

...t; > > I can see DTMF RTP events pass through to carrier, RTP stream looks the same > as the 1.8 server with reliable responses. > > > > On both the 1.4 and 1.8 ast servers, these sip.conf parameters are active on > peer and global settings: > > relaxdtmf=yes > > rfc2833compensate=yes > > dtmfmode=rfc2833 > > > > Now it quickly appears like a problem between the customer PBX and Customer > PRI with the SIP trunks to the ast 1.4 servers but it all worked fine before > with the 1.2 call servers.? After the upgrade of the call servers to 1.8 > DTMF is...

...phone number> secret = blablabla trunkname = via:talk ; GUI metadata hasiax = no registeriax = no hassip = yes registersip = yes trunkstyle = voip hasexten = no fromuser = <phone number> authuser = <phone number> insecure = port,invite dtmf = inband dtmfmode = inband relaxdtmf = yes ;rfc2833compensate = yes port = 5060 canreinvite = no disallow = all allow = ulaw,gsm I did set up a very basic Asterisk box yesterday that put all the conection settings in sip.conf and I even renamed users.conf so it could not load. I then put in about a 10 line hand coded dial plan in extensions.conf and got...

Friends, Thanks for all the feedback. If you have additional success stories or important issues, feel free to continue the discussion. I've learned a lot from your input. As a developer, I spend too much time in the bug tracker, working with particular bugs, so I often wonder how on earth anyone can use this buggy platform for anything business-like. It really feels good to get