similar to: PTT push to talk solution

Hi I've seen some posts on the list regarding integrating Nokia's PTT (nokia 6020 and nokia 6230i) with asterisk And use * as a PTT server.. So far I was able to have mobile register itself , send an invite to it, and get SIP error 603 (DECLINED) back from it. And ofcourse the PTT sign on the mobile is off. App_rpt , it mentioned that is can do PTT , but it is not clear..

I get an echo when going from a SIP phone to a PRI trunk. I hear the echo on the SIP phone. From reading some other post I think that I need to tell me phone company to turn on echo canceling. If the echo was on the other end than it would be my problem? Is this right? What exactly should I say to my phone company so they know exactly what I'm talking about? -- Michael Sampson

Hello, I just installed a Debian Jessie box from scratch which sports a Digium TE435 digital card. I installed the software, built and loaded the kernel modules: # dpkg -l|grep dahdi ii asterisk-dahdi 1:11.13.1~dfsg-2+b1 amd64 DAHDI devices support for the Asterisk PBX ii dahdi 1:2.10.0.1-1 amd64

Thanks a lot Tony and Dhaval for your much appreciable suggestions. Regards, Rajib Rajib Deka SIEMENS Ltd. Robert V Chandran Tower, First Floor, West Wing, #149, Velechery Tambaram Main Road, Pallikaranai, Chennai-100, INDIA. www.siemens.com Mob: +91-9176780669 | E-Mail: rajib.deka at siemens.com Date: Wed, 20 Apr 2011 13:55:25 +0530 From: DHAVAL INDRODIYA <dhaval.it01034 at gmail.com>

On Thu, Jan 15, 2015 at 12:58:26PM -0600, Russ Meyerriecks wrote: > On Thu, Jan 15, 2015 at 2:05 AM, Bertrand LUPART - Linkeo.com > <bertrand.lupart at linkeo.com> wrote: > > However, dahdi_genconf keeps finding no span: > > What am i missing? > > It looks like your driver is loaded correctly. My guess would be maybe > the dahdi-tools is packaged as an older

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On 1/8/18 9:38 AM, Bertrand LUPART - Linkeo.com wrote: > Hello Carlos, > > >> We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it was never

Hi all [Exclamation] I decided to register because i cant solve the problem with a game and speaking program from the topic. When i run those 2 things separately they work very fine. But when i try to run them together World of Warcraft just freezes. Ventrilo still works fine, people can hear me and i can hear them. When i run WoW without Vent, it works really nice. Even voice chat in game

Hello, I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : $ sudo dpkg -l|grep -Ei 'dahdi|asterisk|libpri' ii asterisk 1:11.13.1~dfsg-2+b1 amd64 Open Source Private Branch Exchange (PBX) ii asterisk-config 1:11.13.1~dfsg-2 all Configuration

I am starting wine with the command: wine explorer /desktop=Name,640x480 PPTVIEW.EXE myppt.ptt and I get the following error: [silentm@geisjdell PowerPoint Viewer]$ fixme:actctx:QueryActCtxW 80000010 0x3018b4d0 (nil) 1 0x34fb60 8 (nil) X Error of failed request: BadWindow (invalid Window parameter) Major opcode of failed request: 1 (X_CreateWindow) Resource id in failed request:

Hi Eric, > On 12 avr. 2016, at 17:48, Eric Cooper <ecc at cmu.edu> wrote: > > On Tue, Apr 12, 2016 at 04:36:58PM +0200, Bertrand LUPART - Linkeo.com wrote: >> I just made a asterisk / dahdi fresh install on Debian 8.4, and ended up with the following packages : >> [...] >> However, i can't find any dahdi startup script, neither init.d neither systemd

> We once developed a reminder system for a customer. He's a cleaning > company, cleaning homes and offices. He was spending two hours a day calling > his customers to remind them of their appointment the next day. Two hours a > day equates to 40 hours a month that he saved with that system. He's been > using it for maybe 6-7 years now and not once was a customer upset

Thanks for reply, but this is irrelevant, I'm looking for an *Iridium* gateway. Regards, -- Jean-Denis Girard SysNux Syst?mes Linux en Polyn?sie fran?aise https://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.797.527 Le 03/04/2018 ? 16:05, albert zhang a ?crit?: > http://www.dinstar.cn/en/index.php/GSM/ > > 2018-04-04 10:01 GMT+08:00 Jean-Denis

Hi, I have a mailbox where indexing fails with the following error: # /opt/dovecot2/bin/doveadm -c /tmp/dovecot.conf -o mail_location=/tmp/skesselring index '*' doveadm(root): Panic: file charset-iconv.c: line 132 (charset_to_utf8): assertion failed: (*src_size - pos <= CHARSET_MAX_PENDING_BUF_SIZE) doveadm(root): Error: Raw backtrace:

We have a server that records all calls so we set Mixmonitor with the b option to only record calls that are actually bridged. I notice that we have lost of 44 byte files in /var/spool/asterisk/monitor which correspond to calls that were not answered. If a call is not answered I assume it was never bridged so why would Asterisk create a file? Is there a way to avoid getting those empty

I'm not getting any ringing when I use option r with Dial: Dial("DAHDI/1-1", "motif/8447/+1<called-num>@voice.google.com,,rTt") in new stack Otherwise all works. The call goes through, good audio. sean

AFAIK g729 patent is expiring sometime in 2019-2020. Mitul Limbani On Feb 8, 2017 5:02 AM, "Victor Villarreal" <mefhigoseth at gmail.com> wrote: > Hi Steve, > > I understand your question and your point, but I use the g729 codec from > the link that Carlos share, for almost 6 years from Asterisk 1.4 to v13 > without a single problem. > > So, sory but I

Hi All. I am beginning a project of Call center and predictive diales, my call center have 50 operators, I have 50 analog phone line with the company PTT in my country. I have the following questions: 1- Can I to work this project with Asterisk? 2- What caracteristic of hardware need for my servers? 3- For 50 analog phone line what tipe of card digium I need? Thanks in advanced, Regards.

Hi, I'm trying to get our TE110P card up and running on our * test server. Initially we were getting a flashing red light on the card, however after changing to a crossover cable, we now get a green light for a moment, followed by a solid red light. I am also getting the following message at the command prompt; PRI: !! Got S-frame while link down == Primary D-Channel on span 1

Help. Why is it that I can call out from my GSBudgetone SIP phone but the audio is "one-way'? Why is it that when I call my asterisk phone number, I get a fast busy?