similar to: DTMF doesn't work between Asterisk and Cisco SIP Proxy

Does anyone on-list know of a serice provider that can provide DIDs in 713-861-xxxx? I'd like to port my AT&T POTS lines to an IP based service into my Asterisk box. Michael

We're having an issue with connecting a Cisco ITS installation to * such that DTMF tones are passed to *. DTMF tones aren't passed to voicemail or to any of the interfaces behind *. On the Cisco Side: dial-peer voice 8 voip destination-pattern 9999$ session protocol sipv2 session target ipv4:172.16.1.249 session transport udp dtmf-relay rtp-nte codec g711ulaw no vad We have also

Small sidenote in regards to the wiki, there is also an V6 since windows 10 aniversary. https://technet.microsoft.com/en-us/library/jj649079%28v=ws.11%29.aspx Am 03.05.2017 um 09:52 schrieb Rowland Penny via samba: > On Wed, 3 May 2017 09:15:30 +0200 > Jakub Kulesza via samba <samba at lists.samba.org> wrote: > >> [profiles] >> path =

Dear all, I have an error message, when I try to convert a zoo object (called test) to ts (on R 2.4.0, Package zoo version 1.2-1, Windows XP) > test 1994-05-10 1994-06-09 1994-07-09 0.0024943889 0.0024881824 0.0006955831 > str(test) atomic [1:3] 0.002494 0.002488 0.000696 - attr(*, "index")=Class 'Date' num [1:3] 8895 8925 8955 > is.regular(test) [1] TRUE

Please apply directly using the forms or URL below, or contact Luiz if you have questions. --Vikram Begin forwarded message: > From: "Luiz DeRose" <ldr at cray.com> > Date: December 5, 2006 5:26:01 PM CST > To: <vadve at cs.uiuc.edu> > Subject: Compiler opportunities at Cray > > Hi Vikram, > > I am attaching PDF descriptions of the compiler

All - I have an uneven set of replicates and would like to sample from this set X number of times to generate a mean for each grouping variable. I was thinking the boot package would be the thing to use, but now I'm not so sure ... given the discussion here: http://finzi.psych.upenn.edu/Rhelp10/2010-June/243828.html Given these data (a small subset): > dput(x) structure(list(numSpp

Hi I install the CPOS-7-Installation-20141120-1.ISO on the AIX power 720 (CentOS power PC version for the AIX power serices) Then I saw a message below Please wait, loading kernel. :0,/vmlinux: Unable to open file, Invalid device Boot: How can I boot it up? Please give me an advice . Thanks Sean

Hi, I''m using drbfire to send a "DRbUndumped" ActiveRecord object from the server to the client, through an ADSL line of 4Mbits/s download, and 600Kbits/s upload. I was afraid there would be too many round-trips between the client and the server, and unfortunately this is true: Server code: ------------ #!/usr/local/bin/ruby require ''rubygems'' require

I run a bunch of the Linksys ATA's.. I always use rfc2833 for DTMF. works very well and have never had a problem with it. ..o-------------------------------------------------------o. Brian Fertig NOC/Network Engineer Planet Telecom, Inc. Tampa, FL Office -----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of

I am having trouble with a DTMF-based application on Asterisk 1.0.3. Specifically, when two IAX2-based devices are talking, when they send DTMF to eachother, the other side only hears clicks, and maybe a millisecond of DTMF tone, but not any real duration. Furthermore, when one IAXy device calls the Echo test program, we can hear our echo, but when we punch DTMF in, we get the same effect

How can I debug SIP DTMF? If I do 'sip set debug peer xyz' I see no messages related to DTMF... or if I just do a global SIP debug for that matter.... I am using RFC DTMF but it's not being passed to the PSTN and I need to debug this further. I've tried to increase the verbosity and the debug ('set debug n') and that didn't help either. I assume this is because even

Hi, We are running 1.4.22 and have been experiencing problems with certain IVRs and DTMF Tone duration. We would like to be able to increase DTMF Tone duration by 50 to 100ms over what the user is pressing on his phone. We have a PRI test circuit and an analyer in between to measure tone duration. We have tried setting chan_dahdi.conf parameter 'toneduration', but that does not do

Is it possible to make Asterisk do inband DTMF over IAX?

Hi all, I use a GSM device to send dtmf on my asterisk system (via SIP). the codec I use is ulaw (or a-law). dtmf mode is INBAND. relaxmode is on. but most of the case, I 'missed' some DTMF or I 'double' one. as anybody as seen this before? is there any way to prevent this thanks

Hi all, I am trying to detect DTMF keys from a mobile when asterisk make an outgoing call to the mobile. The DTMF detection on incoming calls (also FROM mobiles) is working very well. The only problem is if asterisk called the phone... Nothing is detected. I am using a digium te205p with PMX/PSTN connection. Everything that I can find in forums are problems with dtmf detection on SIP. Any

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over SIP trunk from which calls get routed to third server (C) (1.6.0.9) via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is where SIP

We have an application which is primarily DTMF driven (automated on both sides), which we are trying to deploy over VOIP and Asterisk (using some Sipuras and some IAXY's). We are finding that in around half the cases, the Asterisk server can't decode the DTMF digits from the field office (or at least some of them). Though, when we place voice calls for testing, we can hear eachother quite

Hello, I am setting up 2 ISDN 4 linux cards and have had great success now that I have got over the initial problems with : and / characters. The only problem I am experiencing now is the sending of DTMF tones over the line to a remote IVR system. If I dial SIP (Cisco 7905 and 7940) to a number over the line, no DTMF tones are heard. I dialed my own home phone and tried it, no matter which

folk, I see from the archives that the issue of nortel handsets not sending dtmf tones to asterisk has been discussed a couple of times, but there is something I quite havent seen answered yet. Is this dtmf issue a problem with the nortel handsets or the PBX itself? If the handset were changed to another one, would this issue be solved? Or must the PBX be changed completely? rgds eb

HI, quick and simple question: is it possible to use inband dtmf with g729? What I would like to do is to have sip clients connected to asterisk and a zaptel card to make pstn phone calls. My concern is to allow sip users to use digits for call destinations that do require menu actions while retaining low bandwith occupation. Tnx ! -- Best regards, Alessio