similar to: iax2 registration being rejected

Hi All, I am using Asterisk 1.4.26.2 and I am getting the following problem making connections to this server. My other servers are Version 1.2.x which have no problems and this 1.4.26.2 server can call the other 1.2.x servers. The error is: chan_iax2.c:4251 handle_call_token: Call rejected, CallToken Support required. If unexpected, resolve by placing address 192.168.25.250 in the

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

I am using a server with asterisk and I am trying to prioritize voip traffick -- I am a newbie at this traffic shaping, so please bear with me. I used the script below and what happens is thatall traffic in the bulk class stops after a couple of minutes. Also, should I include the ports for rtp in the filter statements with the ports 5061 and 4569? Note I have a fairly big pipe -- supposed to

Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 15:44, covici at ccs.covici.com wrote: > > Rowland penny <rpenny at samba.org> wrote: > > > >> On 29/12/15 13:59, covici at ccs.covici.com wrote: > >>> Hi. I am having problems accessing subdirectories on a samba share. I > >>> am using windows 10 build 10586 and linux kernel

I am using viatalk as my voip provider and they use dtmf=rfc2833, but asterisk is not seeing any of the dtmf. I am using CVShead as of 8/26/05. Nothing in the logs indicates a dtmf is being seen. If I use my pots line it sees it fine. Any assistance would be appreciated. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici

I am using asterisk 1.6.0 and -- not all the time -- when a caller comes in and my ivrdials an extension, the ring he gets sounds like a modem handshake instead of the normal ring tone and it only sounds once even if the phone is not picked up. Anyone seeing this -- the logs look fine as far as I can tell. -- Your life is like a penny. You're going to lose it. The question is: How do you

Rowland penny <rpenny at samba.org> wrote: > On 29/12/15 13:59, covici at ccs.covici.com wrote: > > Hi. I am having problems accessing subdirectories on a samba share. I > > am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba > > 4.2.7. I have two shares, one called audio and the other called > > myshare. I cannot access the subdirectories

Oh, I forgot to mention that Asterisk 15 went End-Of-Life last Thursday. :) You should use Asterisk 16. On Mon, Oct 7, 2019 at 5:58 AM George Joseph <gjoseph at digium.com> wrote: > > > On Fri, Oct 4, 2019 at 1:19 PM John Covici <covici at ccs.covici.com> wrote: > >> Hi. I am trying to install asterisk 15.7.4 from git onto a Debian 10 >> system and I am

How did you install these packages -- make sure you do ./configure and if needed make menuselect in each one of these before the make and make install. This is the only thing I can think of -- check whether there are any built-in modules as well. on Monday 03/12/2007 Asterisk Asterisk(asteriskbunnies@yahoo.com) wrote > Hey! > > Thanks for your interest, i checked the modules and i

Hi. I am having problems accessing subdirectories on a samba share. I am using windows 10 build 10586 and linux kernel 4.1.15-gentoo and samba 4.2.7. I have two shares, one called audio and the other called myshare. I cannot access the subdirectories in the myshare share. Here are the definitions. [myshare] comment = root directory path = / #fake oplocks = yes writable = yes printable =

Hi. I am using *1 in features to initiate a mix monitor recording. However, when I hit *1, the callee hears the courtesy tone which I have, so I know when the recording is started or stopped. This is a problem, particularly in automated system where the beep is mistaken for a tone or other problems. Should I file a bug, or is this going to be fixed? -- Your life is like a penny. You're

Hi. I am using asterisk 1.8 and everything was working fine when I was at svn 342661. I then upgraded to vrsion 349339 and discovered the following problem -- one of the end points is a freeswitch box which offers a number of codecs, including PCMU. However, when I tried to make a call I got a 488 response and a message "multiple audio streams not supported" in the log. Is this by

Hi. I am using asterisk 11 svn r401076M and I am getting this warning at times. I can't find much doing a google search, so anyone with any ideas? I have looked at the logs, but can find no particular pattern to indicate where this is happening and the system appears to be otherwise working, but I am still wondering if something is wrong. I am also using freepbx in case there are known

Hi. I have been using asterisk-1.6.2 and if I update the version -- using svn -- to around May 19 or after, when I dial a digit on my fxs port which is on an X400p card, asterisk seg faults. If I go back before about this date, this problem does not occur. The dahdi version is svn 7445. Any ideas would be appreciated. -- Your life is like a penny. You're going to lose it. The question

Hi. I have a SIP provider which wants RFC2833 for the dtmfmode, however I would like to increase the duration of the tone, its pretty short and some IVR's are unhappy or don't detect it. I did poke around, but it looks like when RFC2833 is used, it actually generates rtp packets of some sort, so I have no idea how to increase that duration. Any assistance would be appreciated. -- Your

Hi. I am having a very strange problem --aren't they all -- with the release candidate. I have softphone which talks to asterisk from behind nat -- the asterisk is on a public ip -- and when I hit mute on the softphone, all rtp traffic ceases! Now, a version which does work is r281875, this does not happen in that vrsion, but right after that this strange thing starts and is not fixed in

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

Hi. I hava a variable and in 1.6 I set the string variable to "" and it got the null string. In 1.8, it gets the quotes, I have to set it to nothing at all to make it get the null value. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com

Under linux-2.6.38 I was able to compile and install dahdi, however when I ran dahdi_cfg -vv, I got an invalid argument on my fxs port. I have an old 400P card with one FXS and one FXO module. I have dahdi-trunk r9868 and dahdi-tools-trunk 8670. How can I get this to work correctly? Thanks in advance for any ideas. -- Your life is like a penny. You're going to lose it. The question

Hi. For several months now asterisk will mysteriously stop inserting records into cdr database. I am using mysql and the asterisk addons 1.6.2 to accomplish this. Sometimes there is a strange error about column names, but often there is no error, it just stops. I just have to restart asterisk to get things going again, so I am stumped as to what is happening, or even how to troubleshoot. I