Hello! I have asterisk updated from CVS on 31/8/2004 with sample configuration. I have just changed the sip.conf to register asterisk with sip proxy in out intranet. Then I can successfully make call to asterisk and go to demo IVR, but no response to dtmfs. I try to make call from several sip phones: Cisco7960, Ata186, Snom200. All of them send telephone-event in INVITE, but asterisk answers with no telephone-event in OK. Only Sipura3000 "manages" to get answer with telephone-event in OK and that's why asterisk detects dtmfs. I try to experiment with dtmfmode in sip.conf with no results. So seems there are two issues 1) How asterisk makes decision on telephone-event? 2) Why inband dtmfs doesn't work (g711)? Thanks, Arsen. P.S. All this used to work with the CVS version updated on april. __________________________________ Do you Yahoo!? New and Improved Yahoo! Mail - Send 10MB messages! http://promotions.yahoo.com/new_mail
I'm really new to asterisk, but everything works fine beside DTMF
recognition. As you can see in output generated by asterisk, I constantly
receive following line "Private structure not found in send_digit."
for every DTMF digit I enter on h323 side. I have analog phone atached to the
AudioCodes MP-124 gateway. Both, gateway and asterisk are registered on openh323
gatekeeper. in h323.conf and on AC gateway side DTMF relaying is set to rfc2833.
If anyone has the same problem, I wolud appreciate any help.
Sending 8700120 to context [incoming]
== Starting H323/ip$192.168.3.12:55885/14678 at incoming,500,1 failed so
falling back to exten 's'
-- Executing Answer("H323/ip$192.168.3.12:55885/14678",
"") in new stack
-- Executing ResponseTimeout("H323/ip$192.168.3.12:55885/14678",
"5") in new stack
-- Set Response Timeout to 5
-- Executing BackGround("H323/ip$192.168.3.12:55885/14678",
"agent-user") in new stack
-- Playing 'agent-user' (language 'en')
Dropping duplicate answer!
Ooh, format changed from UNKN to ALAW
Recieved Digit: 1
Private structure not found in send_digit.
Recieved Digit: 2
Private structure not found in send_digit.
Recieved Digit: 3
Private structure not found in send_digit.
Recieved Digit: 4
Private structure not found in send_digit.
Recieved Digit: 2
Private structure not found in send_digit.
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Guys. Im testing the default asterisk demo setup that comes after installing, but I have a problem with dtmf tones... I dial 1000 and listen to the welcome voice but if I try to enter any keys like 2, 500 etc.. nothing happens... Why are my dtfm tones not been recognized? what would be the normal setup on sip.conf for this to work? __________________________________________________________________ Anton Krall
Hi everyone I'm using Asterisk as a transcoding gateway between G711 and ilbc, and when I send out of band DTMF (rfc2833), the DTMF packet sent has the same timestamp as the last RTP packet received by Asterisk. The problem is that when I stop sending RTP, and only want to send DTMF packets (using VAD), the timestamp is the same for all packets, making my media gateway to ignore them. Can this behave be changed by configuration or do I have to work the code (rtp.c???) Thanks Neutel Rodrigues -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051013/34d26399/attachment.htm
Hi, I got this message in the asterisk console while sending the dtmf from a phone. Dec 8 14:55:50 WARNING[29098]: codec_ilbc.c:163 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Please help me to solve this. Thanks Jibu --------------------------------- Yahoo! India Matrimony: Find your partner now. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20051208/c3f464d5/attachment.htm
hi everyone, I do have 2 lines with broadvoice.>From 2 days on one line my dtmf tones are not passed to asterisk server.It siply goes through the extensions routine acting link it did not receive any tone. Could it be problem with my config??? It looks like this:(it worked for last 1.5 year) num2 is ok, but num1 is not working. Any ideas before I call support which is always a problem. [general] externip=lexon.ws bindaddr = 192.168.1.251 port=5060 localnet=192.168.1.0/255.255.255.0 disallow=all allow=ulaw register => number1:pass1@sip.broadvoice.com register => number2:pass2@sip.broadvoice.com/2000 tos=0x18 srvlookup=yes nat=never insecure=yes [sip.broadvoice.com] type=peer username=num1 fromuser=num1 authuser=num1 secret=pass1 host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [sip.broadvoice.com.home] type=peer username=num2 fromuser=num2 authuser=num3 secret=pass2 host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no nat=never dtmfmode=inband [broadvoice-incoming] type=peer host=147.135.8.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming2] type=peer host=147.135.0.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never [broadvoice-incoming3] type=peer host=147.135.4.128 context=from-broadvoice qualify=yes canreinvite=no disallow=all allow=ulaw nat=never thx Bart Wegrzyn