search for: dtmfmode

Hi folks, I?m using chan_h323 to dial out to a gateway which connects me to the PSTN. In order to use a menu system such my bank menu system, I have to set dtmfmode=info in my sip.conf for my Cisco 7960 phone. However, dtmfmode=info won?t work with Asterisk?s voicemail system. I?m using the g.729 codec for h323 and Asterisk. I?m told dtmfmode=inband won?t work with g.729. Is it possible to use dtmfmode=info with h323 and access my Asterisk voicemail? Summa...

Incoming FWD calls from other FWD users, iaxtel, or via ipkall, need to have dtmfmode=rfc2833. However, incoming FWD calls from the dialup access numbers (such as libretel) need to have dtmfmode=inband. To solve this problem, I created a second FWD account and configured sip.conf as follows, in order to match the incoming number to the proper dtmfmode: [fwd-rfc] type=friend s...

...one given by them. Here it Is (sip.conf). For others who have a working config could u please share so that I could compare. Thank You [broadvoice] type=friend username=[number] fromuser=[number] secret=[password] host=sip.broadvoice.com fromdomain=sip.broadvoice.com context=from-broadvoice dtmfmode=inband disallow=all allow=ulaw canreinvite=no nat=yes [bv-in-1] type=friend host=147.135.8.128 context=from-broadvoice dtmfmode=inband canreinvite=no nat=yes allow=ulaw [bv-in-2] type=friend host=147.135.0.128 context=from-broadvoice dtmfmode=inband canreinvite=no nat=yes allow=ulaw [bv-in-3]...

Hi.. I just wondering why DTMF are not recognized by aterisk on incoming calls from my SIP provider ... ANy suggesteions ?` /Mike

I have two SIP extensions defined in sip.conf register => 4450 at 10.20.1.1/4450 [4450] type=friend username=4450 host=10.20.1.1 allow=all dtmfmode=inband context=incoming register => 4451 at 10.20.1.1/4451 [4451] type=friend username=4451 host=10.20.1.1 allow=all dtmfmode=inband context=incoming Pretty straight forward... The first one works the second one does not. Then if I switch them again the first one works the second one does not....

...istering with asterisk. The phone boots DHCP gets an address, loads the SIP software and sets there for me to dial. However, I get the INV when I dial. Any ideas on why the phone is displaying invalid and what to do about it??? Thanks, jerry sip.conf ------------------------ [201] type=friend dtmfmode=rfc2833 username=201 secret=201 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Media Assistant" <201> [202] type=friend dtmfmode=rfc2833 username=202 secret=202 disallow=all allow=ulaw allow=alaw host=dynamic context=smvoice-sip callerid="Media As...

...X localnet=10.0.0.0/255.0.0.0 disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw register => ##########@sip.broadvoice.com:XXXXXXXXX:##########@sip.broadvoice.com/1234 [sip.broadvoice.com] type=peer host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=########## context=default dtmfmode=inband canreinvite=no disallow=all allow=ulaw allow=g729 allow=g726 allow=alaw [kevin] type=friend regexten=1001 username=kevin fromuser=Kevin Marvin ; Specify user to put in "from" instead of callerid secret=XXXXXX host=dynamic canreinvite=no defaultip=10.1.1.16 amaflags=defau...

...Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP host=192.168.11.200 wcardep=aaln/* callerid = "test" <8000100> callwaiting=no transfer=no cancallforward=no dtmfmode=rfc2833 canreinvite=no singlepath=no slowsequence=yes line => aaln/1 callerid= "test" <8000101> callwaiting=no transfer=no cancallforward=no canreinvite=yes dtmfmode=rfc2833 line => aaln/2 callerid= "test" <8000102> callwaiting=no transfer=no cancallforward=no...

...e is an IVR to select the extension. The DTMF tones are not being sensed so the IVR does not work and incoming calls are not being answered. I have listed my sip.conf entries. Is there any solution to this? ;Sipura units [101] type=friend host=dynamic context=default secret=mysecret mailbox=101 dtmfmode=inband disallow=all allow=ulaw [3200] type=friend host=dynamic context=pstn-in secret=mysecret qualify=yes dtmfmode=inband disallow=all allow=ulaw insecure=very [pstn-spa3k1] type=peer auth=md5 host=192.168.101.11 port=5061 secret=mysecret username=asterisk fromuser=asterisk dtmfmode=inb...

...h Vitelity was not working for me until 1-2 days ago when I changed it from rfc2833 to inband! Maybe I just missed the change date and I should change it back? ---- Date: Tue, 22 Jul 2008 12:23:39 -0400 From: "Mark G. Thomas" <Mark at Misty.com> Subject: [asterisk-users] Vitelity dtmfmode=rfc2833 started working! To: asterisk-users at lists.digium.com Message-ID: <20080722162339.GA12794 at lucky.misty.com> Content-Type: text/plain; charset=us-ascii Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, inco...

Hello folks. I'm not sure if this is the right list for this question, but I'll start here. If I'm using a SIP provider and I have an entry in sip.conf that looks like: [8315551212] type => friend ... dtmfmode => inband ... When I pick up the phone, call someone through this provider, and press numeric digits to generate dtmf tones, who is actually generating the tones at the other end? What I'm noticing is that if I call a pstn line using an entry like this through asterisk, and then press digit...

Hi, Last week my outbound (dtmfmode=inband) DTMF via Vitelity started acting more weird than usual, and for outbound calls, incoming DTMF tones would consistenly get stuck, breaking a call screen macro I had set up. I checked "sip show peer" and saw that Vitelity for inbound was now reporting "DTMFmode : rfc2833"...

I understand there has been some issues sending DTMF tone through Broadvoice. Can some provide me with symptoms? --- Outgoing mail is certified Virus Free. Checked by AVG anti-virus system (http://www.grisoft.com). Version: 6.0.700 / Virus Database: 457 - Release Date: 6/6/2004

...oing again. I can connect using a softphone (X-Lite) and make calls in and out so I know they are "up". What am I missing: [general] externip=216.73.179.xxx bindaddr = 0.0.0.0 port=5060 localnet=172.16.0.0/255.255.255.0 disallow=all allow=ulaw context=default ;from-broad dtmfmode=inband register => 212202xxxx:XXXXXXXX@sip.broadvoic tos=0x18 srvlookup=yes nat=yes insecure=yes [Broadvoice] type=peer username=212202xxxx fromuser=212202xxxx secret=XXXXXXXX host=sip.broadvoice.com context=sip fromdomain=sip.broadvoice.com canreinvite=no dtmfmode=inband nat=...

...#39;(Zap' (cause 66 - Channel not implemented) Configs: [root@asterisk asterisk]# cat sip.conf [general] bindport=5060 bindaddr=0.0.0.0 context=default disallow=all allow=gsm allow=ilbc allow=ulaw allow=alaw srvlookup=yes ; [400] type=friend username=400 host=dynamic secret=12345 regexten=400 dtmfmode=rfc2833 canreinvite=yes nat=no mailbox=400 ; [401] type=friend username=401 host=dynamic secret=12345 regexten=401 dtmfmode=rfc2833 canreinvite=yes nat=no mailbox=401 ; [402] type=friend username=402 host=dynamic secret=12345 regexten=402 dtmfmode=rfc2833 canreinvite=yes nat=no mailbox=402 ; [410]...

...key ; Password for device host=dynamic ; This host is not on the same IP addr every time ;context=from-sip ; Inbound calls from this host go here mailbox=2000 ; Activate the message waiting light if this ; voicemailbox has messages in it dtmfmode=rfc2833 [2001] ; Duplicate of 2000, except with different auth data type=friend username=2001 secret=homemonkey host=dynamic ;context=from-sip mailbox=2001 dtmfmode=rfc2833 [2002] ; Duplicate of 2000, except with different auth data type=friend username=2002 secr...

...register =>16194077214:<<password>@69.59.234.67:5060/202 [authentication] [3000] type = friend context = default username = 3000 host = dynamic mailbox = 3000 dtmfmode = rfc2833 [3001] type = friend context = default username = 3001 host = dynamic mailbox = 3001 dtmfmode = rfc2833 [3002] type = friend username = 3002 context = defau...

...11111:MYSECRET at pbxluca/00493511111111 register => 00493512222222:MYSECRET at pbxfax/00493512222222 register => 00493513333333:MYSECRET at pbxanika/00493513333333 register => 4444444444:MYVERYSECRET at messagenet/4444444444 [pbxluca] type=peer defaultuser=00493511111111 secret= MYSECRET dtmfmode=rfc2833 host=172.16.34.132 context=luca_incoming outboundproxy=172.16.34.132 port=5060 fromuser=00493511111111 fromdomain=172.16.34.132 usereqphone=yes canreinvite=no insecure=invite qualify=yes qualifyfreq=600 [pbxfax] type=peer defaultuser=00493512222222 secret= MYSECRET dtmfmode=rfc2833 host=1...

Hello, I am fairly new to asterisk. I am currently using asterisk as a more convenient sip side voicemail system. My problem: I have cisco 7960 phones whose out of band dtmf tones are recognized properly(when dtmfmode=rfc2833) by asterisk but whose in-band dtmf tones are recognised poorly(when dtmfmode=inband) . For example 7999 comes out as 799999, 4242 comes out as 442422 ... etc I also have a cisco 2600 acting as a router/bridge between a Nortel Option 11C pbx and the voip/sip side. It can only send in-ba...

...lso MSN to MSN works fine too. my SIP details [general] port = 5060 bindaddr = 0.0.0.0 context = bogon-calls ;context = default disallow=all allow=ulaw allow=alaw allow=ilbc allow=gsm ;My SIP phone - GS [2000] type=friend username=2000 secret=qweqwe host=dynamic context=from-sip mailbox=2000 dtmfmode=inband ;MSN Msgr [2002] type=friend host=dynamic insecure=yes dtmfmode=inband ;dtmfmode=rfc2833 context=from-sip mailbox=2002 ;auth=plaintext i did a SIP trace it says Format=UKN CSeq=BYE thanks for the help, -Balaji --------------------------------- Do you Yahoo!? Yahoo! Photos - Get your...