similar to: no ringing tone with Dial option r

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

On 12/19/2014 09:42 AM, Rusty Newton wrote: > On Wed, Dec 17, 2014 at 1:09 PM, sean darcy <seandarcy2 at gmail.com> wrote: >> I've got a confbridge set up which works if dialed locally: >> >> -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack >> -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1",

Have you enabled DTMF logging and seen the DTMF codes being recognised by Asterisk? I had a bunch of soft phones that I had to change to using ?sip info? for the DTMF signalling as the RFC signalling was not always being recognised. This would cause transfers to appear as if the user had not dialled any digits. On 20/12/2014 20:52, "sean darcy" <seandarcy2 at gmail.com> wrote:

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

I've got a confbridge set up which works if dialed locally: -- Executing [266 at internal:1] Answer("DAHDI/1-1", "") in new stack -- Executing [266 at internal:2] SendDTMF("DAHDI/1-1", "1") in new stack -- Executing [266 at internal:3] ConfBridge("DAHDI/1-1", "1") in new stack -- <DAHDI/1-1> Playing

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

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

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

[This email is either empty or too large to be displayed at this time]

> 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

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 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

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

I have a call on gv over motif. I try to bridge it to another call over motif, but a different gv account, and I get congestion. motif only handles one 1 channel at a time?? sean

Using 11.4.0, trying to use SILK on the cell phone to ulaw over gv, but no success: [Jun 10 16:18:22] WARNING[4090][C-0000000a]: channel.c:6164 ast_channel_make_compatible_helper: No path to translate from SIP/ng-00000000 to Motif/+12025551212 at voice.google.com-da3c [Jun 10 16:18:22] WARNING[4090][C-0000000a]: app_dial.c:3032 dial_exec_full: Had to drop call because I couldn't make

On 12/21/2014 11:09 AM, sean darcy wrote: > On 12/21/2014 04:42 AM, Patrick Beaumont wrote: >> Have you enabled DTMF logging and seen the DTMF codes being recognised by >> Asterisk? I had a bunch of soft phones that I had to change to using ?sip >> info? for the DTMF signalling as the RFC signalling was not always being >> recognised. This would cause transfers to appear

Some calls I get from google voice, I just send myself an email about the call and want to hangup. But I can't seem to make gv know I've hung up. extensions.conf: same => n,GoToIf($["${CALLERID(num)}"="office"]?email) ................. same => n(email),System(/usr/local/bin/emailme........) same => n,Answer() ; also tried without this same =>

My internal lan is small, 100mb, all wired. aastra phones. sip show peers ....... 142/... 10.10.10.42 D A 5060 OK (136 ms) 144/... 10.10.10.44 D A 5060 OK (138 ms) 145/... 10.10.10.45 D A 5060 OK (133 ms) But pings are < 1ms: ping 10.10.10.42 ........ rtt min/avg/max/mdev = 0.479/0.483/0.497/0.021 ms Why are the sip latencies so

On Sat, Jan 17, 2015 at 09:31:33AM +0100, Bertrand LUPART - 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 version that doesn't know > >> about the newer te435

>Hi Jerry > >As others have eluded to, the 'PTT' feature can mean different things to different >people depending on their background.> > >Is it fair to say that you're looking for a one-touch button which initiates a call to >the other end and causes the other end to automatically answer in speakerphone >mode? >If that would foot the bill then have a