similar to: DTMF problems with IVR - What DMTF Tx method

I've been working in the VoIP industry for just a bit over a year now... Mostly taking care of the underlying systems. I've now reached the point where I'm being drawn more and more into the call processing side of things. My background is in computer and "classic" telephony systems (DMS250/MTX, DSC 400, T1, channel banks. telabs analog echo supressor modules and

Hmmm.... Can't place calls... Can't access website... Neither of the 3 nameservers answer anything... Anyone heard/know something to explain all this?

Hi, I've gone through the source tree and I don't see a MIB description file for the SNMP agent in asterisk. can someone point me to it. Thanks, Bruce ferrell

I'm trying to make an AMI call. I want to call a number, play an announcement when the call is answered, then call a second number and connect the two when the second call is answered. I an able to make a simple call to two numbers and connect them using the manager API but playing the announcement has me beat. Suggestions anyone? Bruce Ferrell

Hi Bruce Thank you very much for reaching out. This is likely to be a short discussion due to me potentially learning that my infrastructure possibly has shortfalls that I did not appreciate until just recently. What I was hoping to do is deploy Windows from a Thecus branded NAS. Natively the NAS does not support PXE, but with community created mods it is achievable. So with this mod

A while back (last year maybe?), there was a Digium blog post on setting up WebRTC. I was never able to get that working. I was working with Asterisk 15 on a RHEL derived distro and had no idea of where to go to shoot the failure. Has anyone got a tutorial with trouble shooting?

Thank you for the prompt reply. What you said is promising so if I may, with your help can we decipher what SYSLinux files goes where from the ground up? If this is too broad a question or may take up too much of your time then I understand and appreciate your help thus far. Richard -----Original Message----- From: Bruce Ferrell [mailto:bferrell at baywinds.org] Sent: Saturday, 17 November 2018

Hi All, Does anyone know if there has been any developments in asterisk to use LineJacks to dial out (connect to the PSTN)? The card works perfectly with virtually anything else but asterisk. Maybe the CVS versions have some work on it? Cheers, -Z -- __________________________________________________________ Sign-up for your own FREE Personalized E-mail at Mail.com

On Thu, 8 Jun 2017, Bruce Ferrell wrote: > Yes, 7 does track upstream. upstream 6 uses systemd also and Scientific > Linux 6 does not. I would say that indicates a solution. Upstream 6 uses systemd? jh

Hi all, I'm looking for pointer for setting up VNC so that access to the system is via gdm/kdm. Yes, I know about vino, and /etc/sysconfig/vncservers but what I'm looking for is a sertup that allows me to see the *dm login screen instead of being dropped direct into a desktop. Thanks in advance

so far all the mirrors I've checked have 3.9 in the directory for 3.x Can anyone tell me how to get back versions? I'm looking for 3.4 or 3.5 Thanks in advance Bruce Ferrell

I think the title more or less says it all. Is there any such animal? TIA

Hi Jean-Marc, thank you for that suggestion! It seems that the file "fixed_c6x.h" is not part of the Opus sources, so the compiler cannot find it after enabling the TI_C6X_ASM config option. Maybe it was only part of an early version of the Opus sources? I looked for the file in versions V1.1, V1.1.1, V1.2alpha and V1.3 but did not found it. Do you have an idea, where I can get the

Dear Opus family, we have implemented the Opus codec at a Texas Instruments DSP C6418. It is working fine! Does anyone has experience with the configuration of the codec for a speed optimized implementation on that DSP? At the moment, we use the following settings: #define NONTHREADSAFE_PSEUDOSTACK 1 #define FIXED_POINT

Dear List; I am trying to create a link between Asterisk and My softswitch, the link to be SIP Trunk. I did the below configuration and I do not know if any one can help me and advise me to have better configuration to be sure that link is fine. But I do not know how to determine the SIP username to be sent for my softswitch as sometimes the softswitch need to check it. Also, does asterisk

On Mar 11, 2019, at 6:16 PM, Bruce Ferrell <bferrell at baywinds.org> wrote: > > What I've learned to do when I have this sort of issue is to pop out of CPAN and into ~/.cpan/build. If you mean that you do that manually, you don?t have to. The ?look? command in the cpan shell or the --look option to cpanm does that automatically. That is, it unpacks the module and drops you

much static testing of my realtime configuration and applications I'm almost ready to pull the trigger. The one thing I've been able to determine is what I need to do to migrate my g729 licenses. Has anyone got any advice for me on this? The Digium site is... difficult to navigate TIA Bruce Ferrell

Hi Jean-Marc, Alfred and Greg, Are you receiving the mails from IETF about draft-ietf-avt-rtp-speex The mails are not coming from AVT mailing list, but I think we are all 3 part of a minimal list (draft-ietf-avt-rtp-speex at tools.ietf.org) dedicated to latest discussion about the draft. I have answered some questions, but there are small changes and adaptation still required to the ietf

Gents, I've built an Asterisk system to replace our PBX at work and have Cisco 7960 phones (SIP 7.4) running with Asterisk 1.0.7. How to I get Asterisk to recognise the '#' being pressed during a call? In sip.conf I have entries likle this: [2001] type=friend context=local-phone auth=md5 username=2001 secret=xyzzy callerid=Jack Tubby <2001>

Dear Alex; Thanks alot for your nice help. This is if I need to let Asterisk register with another softswitch (so I used register =>), what if I need asterisk to send call for the softswitch without register to it (directly)? If I removed the register => then how it will distiguish the IP address in the "host" at the [sip_trunk] is the IP address of the softswitch that need to