Displaying 20 results from an estimated 41 matches for "sipdtmfmod".
Did you mean:
sipdtmfmode
2003 Nov 07
0
sipdtmfmode problem
Greetings. I'm having a bit of a problem using the sipdtmfmode app. I have two
incoming paths to * from pstn via FWD that use differing dtmfmode. IPKall
wants rfc2833, libretel wants inband. If I set dtmfmode= in the fwd peer
config in sip.conf each works seperately, and I'm trying to use gotoif and
sipdtmfmode to switch based on the CID calling. Outp...
2003 Dec 03
0
BOOM! Crash when trying to use SIPDtmfMode on an outgoing call!
...ll the settings my Sipura SPA2000 offers, I found inband actually works..
unfortunately, I can't get anything else to pick up my inband DTMF
(including asterisk's builtin voicemail! It just times out and says I never
entered a login!). So, I did some digging around, and figured I might try
SIPDtmfMode to change my DTMF mode when I'm calling out.. that resulted in a
prompt crash, and the info included below out of gdb. Is it me? Am I
misunderstanding the appropriate use of SIPDtmfMode? If so, that's fine,
just bonk me on the head with a yellow pages book or something.. Also.. how
can I...
2017 Jun 29
2
PJSIP equivalent for SIPDtmfMode?
Can't find a way to control the dtmf mode on a per session basis with
pjsip, used to use SIPDtmfMode from the dialplan with chan_sip. Any
hints on how to do this?
2004 Jan 25
2
Incoming SIP matching
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
secret=*****
2004 Aug 10
1
DTMF issues
...ions, voicemail, etc. If making an
outgoing call like this spa --> * --> Cisco AS5350 --> PSTN, I get no dial
tone. I am working unsuccessfully with Cisco right now on this, but they
cant find anything wrong. I have tried all suggestions I can find from the
list and elsewhere. I have added SIPDtmfMode to my outgoing extensions, that
still doesnt help. Does anyone out there have experiance or ideas with this
setup?
2004 May 11
1
Use buttons (other than #) after call is bridged?
Hi,
can i somehow use the other buttons to execute some apps, *without* hanging
up the call?
Something like:
exten => s,1,Dial/SIP(1234)|4,5,7,9
exten => 4,1,Monitor(wav)
exten => 5,1,SIPDtmfMode(inband)
exten => 7,1,AGI(turnoncoffeemachine.agi)
exten => 9,1,System(smbnuke boss)
Regards,
AA
_________________________________________________________________
Watch movie trailers online with the Xtra Broadband Channel
http://xtra.co.nz/broadband
2004 Jun 02
1
DTMF and SIP
Hi
I have 2 x SIP hand phones. I have set the DTMF to rfc2833 on the
phones and tried both dtmfmode=rfc2833 and sipdtmfmode=rcf2833 (also
tried inband) and I get the following error:
june 2 17:21:10 WARNING[213006]: codec_ilbc.c:145 ilbctolin_framein:
Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)?
This means that I cannot get access to voicemail from the handsets !!!
Any clues???...
2004 Dec 14
3
Realtime problem
...arsing '/etc/asterisk/sip.conf': Not found (No such file or directory)
Dec 14 16:11:37 NOTICE[8868]: chan_sip.c:8462 reload_config: Unable to load
config sip.conf, SIP disabled
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered application 'SIPDtmfMode'
And my device(s) won't register. I don't even see them attempt the
registration...(from the CLI in vvvvery verbose.)
Maybe I'm not using the right version of asterisk??? Is that possible and how
would I know? My "show version" gives me this:
*CLI> show ver...
2004 Dec 31
2
Host IP address, Crash on startup, Console grabs soundcard - Newbie needs help
...chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
== SIP Listening on 0.0.0.0:5060
== Using TOS bits 0
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered application 'SIPDtmfMode'
== Registered application 'SIPAddHeader'
== Registered application 'SIPGetHeader'
Problem 3)
Sometimes the program crashes during (at end of) the startup sequence. Warning about "flexibel rate not heavily
tested". Is this just a codec I can con...
2007 Mar 15
1
asterisk n-way call problem
...ec settings but no success.
the extensions and sip configuration is below if you want to have a look. I
dont have any clue why its not working.
#######################extensions.conf#######################################
[local]
exten => _XX,1,Set(DYNAMIC_FEATURES=nway-start)
exten => _XX,2,SIPDtmfMode(inband)
exten=> 10,3,Dial(SIP/saad,,tT)
exten=> 10,n,Hangup
exten=> 11,3,Dial(SIP/riz,,tT)
exten=> 11,n,Hangup
exten=> 12,3,Dial(SIP/rehmat,,tT)
exten=> 12,n,Hangup
[dynamic-nway]
exten => _XXX,1,Answer
exten => _XXX,n,Set(CONFNO=${EXTEN})
exten => _XXX,n,Set(MEETME_...
2007 Oct 29
0
IAX2 weirdness and rejected calls: Invalid BYTE
...all: 10100 [193.82.116.194:4569]
USERNAME : vikki
DATE TIME : 2007-10-29 19:46:36
Rx-Frame Retry[ No] -- OSeqno: 001 ISeqno: 001 Type: IAX Subclass: ACK
Timestamp: 00004ms SCall: 10100 DCall: 00002 [193.82.116.194:4569]
-- Executing [01905888007 at sip-default:1]
SIPDtmfMode("SIP/213.166.5.134-086112f8", "inband") in new stack
-- Executing [01905888007 at sip-default:2]
NoOp("SIP/213.166.5.134-086112f8", "01905888007 ") in new stack
-- Executing [01905888007 at sip-default:3]
Dial("SIP/213.166.5.134-086112f8"...
2004 Apr 02
1
error with asterisk -vvvvc
...on 0.0.0.0 port 5036
[chan_sip.so] => (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
== SIP Listening on 0.0.0.0:5060
== Using TOS bits 0
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered application 'SIPDtmfMode'
[chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem
Driver)
[chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver)
[chan_agent.so] => (Agent Proxy Channel)
== Registered channel type 'Agent' (Call Agent Proxy Channel)
== Registered application &...
2004 Apr 16
0
SIP IAX2 MySQL Config
...erisk/sip.conf': Found
== SIP Listening on 192.168.0.10:5060
== Using TOS bits 0
Connected to database 'asterisk_config' on 'localhost' as 'asterisk_user'
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Registered application 'SIPDtmfMode'
[chan_iax2.so] => (Inter Asterisk eXchange (Ver 2))
Apr 16 14:10:34 WARNING[1074449120]: chan_iax2.c:6218 load_module: Unable
to open IAX timing interface: No such device
== Manager registered action IAXpeers
== Parsing '/etc/asterisk/iax.conf': Found
Connected to database ...
2004 Dec 19
2
VoicemailMain can't read from phone keyboard!
Hello
I try to set up voicemails for extension. When VoicemailMain gets called, it
prompts for mailbox and password. It seems not able to read from the phone.
So the authentication always fails.
I desparately need help to understand what is wrong. Here is a part of my
extensions.conf:
exten => _8500, 1, Wait(2)
exten => _8500, 2, VoicemailMain(${CALLERIDNUM})
exten => _8500, 3, Hangup
2005 Mar 29
0
rfc2833 cisco 7960 DTMF issue
I'm having an issue sending DTMF to cisco
dialing this extension I should hear the dtmf tone
RTP playload 101 has been sent to the cisco phone, but no audio.
in the dialplan
exten => 8603,1,Answer(1)
exten => 8603,n,sipdtmfmode(rfc2833)
exten => 8603,n,SendDTMF(1|100)
exten => 8603,n,hangup()
sip.conf
dtmfmode=rfc2833
SIPDefault.conf
I did play with all possible settings for dtmf_outofband: avt,
avt_always, none and 0,1 for dtmf_inband
nothing happens
cisco 7905g is working OK with this example
cisco 7960 firmw...
2005 May 23
0
How to detect DTMF and change if needed
...to
inband (is this even a proper way to handle 2 DTMF modes?).
[sipprovider]
type=friend
host=xxx.xxx.xxx.xxx
disallow=all
allow=ulaw
maxexpirey=15
dtmfmode=rfc2833
dtmfmode=inband
nat=no
insecure=very
canreinvite=no
I have searched and searched and the closest thing that I have found
is "SIPDtmfMode" but from what it looks like it needs to be initiated
before the call is placed.
By the way - the reason inband is not being used is that digit
accuracy is terrible with the inband setting.
Any thoughts are appreciated.
2005 Jul 21
0
re: DTMF woes, continued
...rt = 5070 ; Port to bind to
disallow=all ; Disallow all codecs
allow=ulaw
allow=alaw
allow=ilbc
allow=gsm
dtmfmode=rfc2833
register => username:secret@myprovider.com/myDID
just to make sure everything is set properly, i even threw in
exten => myDID,3,SIPDtmfMode(rfc2833)
in extensions.conf to make sure that dtmf is coming over in rfc2833
and not inband.
if it helps, the provider in question is nufone and jeremy from nufone
told me to use only rfc2833. anyone else have this problem? any
pointers or ways to solve this? it's making me insane...if anyo...
2005 Sep 15
0
Changing the sip port in sip.conf does not work
...o, I always get
== Registered channel type 'SIP' (Session Initiation Protocol (SIP))
== Parsing '/etc/asterisk/sip.conf': Found
== SIP Listening on 64.1.16.172:5060
== Using TOS bits 4
== Parsing '/etc/asterisk/sip_notify.conf': Found
== Registered application 'SIPDtmfMode'
Is there any other way to make the port change work?
Also I never got an answer about how to prevent unregistered sip phones from
sending inbound SIP calls. I can send calls regardless if my softphone is
registered or not, when autocreatepeer=no. This is flaw that makes Asterisk
very insecu...
2005 Sep 22
1
WaitExten
Hi,
In my dialplan I'm using a WaitExten() command. It works only with Zap
phones. When I dial this command with Sip phone asterisk do nothing.
Should I put extra definition in sip.conf to make this work with Sip
phones?
Thanks in advance
Cheers
2006 Jun 28
1
asterisk -> my cell phone's voicemail sound problems
When I fail to pick up a call from Asterisk to the PSTN to my cell
phone and let it go to voicemail, the sound quality is always really
bad. When I call my cell phone's voicemail a few minutes later, it's
really garbledy and sounds clipped or something.
I've tried using Monitor to record the sounds that are being played to
my cell's voicemail, and the monitored sound sounds fine