similar to: IAX2 and DTMF

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

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

Hello, I have some reports from users that occasionally DTMF will stop working in voicemail and they will have to exit the system to get it to work again. The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with Ulaw codec. This is all on an internal switched 100mb lan. Has anyone else seen anything like this? Thanks, - Brent

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both

I am not sure if this is the place for Digium user-to-user discussion, but... We have deployed many (20+) IAXy's in the field. At a couple of locations, the IAXy's have just stopped working after 1 or 2 days use. No lights go on, no DHCP lease is renewed as far as we can tell, and of course no dialtone and no registration with the server. Has anyone else experienced high failure rate

I think I have my echo problem solved, now i need to tackle the MWI. I can't seem to get it to light up. I'm using Asterisk 1.4.14. Here's a section from my sip.conf for my test phone: [general] context=internal allowguest=no allowoverlap=no allowtransfer=yes notifyhold=yes bindport=5060 bindaddr=0.0.0.0 srvlookup=yes pedantic=yes vmexten=9998 at internal ;vmexten=*97

Hello: I need to be able to reliably send out touchtone to any calling party who comes into my pbx. The standard things to help with this have been done as far as I know: 1. dtmfmode is rfc2833. 2. The phones themselves are set to rfc2833. 3. allow=ulaw 4. On internal calls between extensions, touchtone works fine. Also, I have reviewed sip.conf with my carriers. Now for the

We have an Asterisk server online, with many SIP clients (some Sipuras, some laptops), and we're also using some IAXy's. I've been trying to find a simple way to check "who's online", meaning who is reachable at the moment, without actually going through and dialing everybody. Is there a way to do this with Asterisk? I am sorry if I'm missing something obvious, but I

I'm somewhat new to Asterisk and am tasked with having it perform some automated functions. Is there a way with the current system (and/or extra modules out there to:) 1. Launch "something" from a command line (on the Asterisk server) to: 2. Dial an extension 3. Issue some DTMF sequences, 4. Record the output to a WAV (or GSM) file, and 5. exit Any quick pointers would be greatly

I've configured Asterisk 0.7.2 to work together with Cisco ATA186 (SIP,G.711. RFC2833) and OpenPhone (H.323, G.711). But there is an issue while calling from ATA186 to OpenPhone via Astrisk - when I press any key on analogue phone connected to ATA, Asterisk shows following message: -- Executing Dial("SIP/519-3781", "OH323/62.213.36.100|20|Tt") in new stack --

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

I have one use on our PBX who has been experiencing seemingly random disconnects. The user is on the same LAN as everyone else, using the same type of phone (79XX loaded with SIP firmware) as everyone else. He had some disconnects a few weeks ago, I suspected the phone, so I swapped his with mine. I have since not had issues with his old phone, however, he has had issues using mine. So, the

I've recently installed Asterisk-based servers at several of our branch offices. Each server has 2 X100P cards to handle 2 incoming voice lines. I was having a lot of trouble with Echo until I got OSLEC running on all of the servers, but now we have a new problem. Incoming callers are not always able to dial extensions. I would say probably 95% of the calls go through correctly, but

Still grasping at straws trying to solve DTMF detection issues with one of my asterisk servers. This particular server is now running Asterisk 1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console only) with 2 X100P cards. I have tried adjusting channel gains, turning call progress and relaxdtmf on and off, switching echo cancelers, just about everything that Google turns up and I can't

Good day Guys I forgot to add and mention a very important piece of the puzzle. We are making use of dovecots compression plugin. I.e. https://doc.dovecot.org/configuration_manual/zlib_plugin/#compression Regards Brent Clark On 2019/11/11 14:49, Brent Clark wrote: > Good day Aki > > Thanks ever so much for replying. > > Interesting that you ask the version of dovecot. Any

We are testing some DTMF-driven applications over VOIP (legacy systems which use fast pulses of standard DTMF tones). The applications work fine when Digium IAXy's are used - no loss or garbling of DTMF tones. However, when we use SIP modems (such as Sipura 1000's), the DTMF tones are frequently uninterpretable and our applications have to ask for retries. I am under the impression that

Hi Brent, You're correct. The segmented stack model is intended for performance, not security. Its goal is to allow lightweight threads to be very cheaply allocated and destroyed. For your model, you would most likely want to use different address spaces for the different memory types. Currently, clang does not allow you to define the address space of variables with automatic storage

I have several branch offices, each with their own Asterisk server (version 1.4.22.1) handling their PBX functions. All of these offices need to talk to each other. In sip.conf I created a peer entry for each office with a username of branch-user and a friend entry for every branch-user with the username being just the branch, for example: [Office2] username=Office1-user host=10.10.80.253

What version are you running? Aki On 11.11.2019 12.26, Brent Clark via dovecot wrote: > Good day Guys > > Just an update, my colleague and I came across this script. > > https://www.dovecot.org/tools/maildir-size-fix.pl > > We made a backup, ran it, but unfortunately the problem still persists. > > Regards > Brent > > On 2019/11/11 11:42, Brent Clark wrote:

Hi Duncan, It doesn't appear to be what I'm looking for as there is copying of the old stack to the new stack. Additionally, according to the LLVM docs on segmented stacks (http://llvm.org/releases/3.0/docs/SegmentedStacks.html) the stacklings are allocated memory from the heap. Since this places them all on the same memory segment, a read/write from one stackling could technically