Displaying 20 results from an estimated 2000 matches similar to: "using Wifi smartphones as SIP clients"
2012 Dec 11
1
DECT phone for home: siemens A510 v. Grandstream DP715
I have an asterisk server at home. I'm looking to replace my internal
phones with sip cordless (DECT) phones. I'm now looking at the Siemens
A510IP base ($90 ) and A510H handset ($40) and the Grandview DP715 base
($80) and DP710 handset ($45).
The Siemens has a feature were I can also use a PSTN landline, but I not
sure that's a great benefit.
Has anybody tried these phones? I
2012 Jun 03
2
Caller ID : FSK ETSI or FSK US
Hello, All :)
Regarding to incoming caller ID on PSTN line, which one is best supported
by asterisk: is it FSK ETSI or FSK US?
I bought some caller ID converter hardware (convert DTMF to FSK and vice
versa) but still asterisk can not detect it.
The converter has a switch FSK ETSI or FSK US
This is what I put in /etc/asterisk/chan_dahdi.conf
...
cidsignalling=bell
cidstart=ring
...
If after
2015 Jun 29
2
Product CDR/Queue/Meetme
Hi Helviom
I am interested to evaluate your product.
What asterisk version you build this product around?
--
regards,
abdul basit | p: +92 32 1416 4196 | o: +92 30 0841 1445
On Tue, Jun 23, 2015 at 7:34 PM, Tech Support <asterisk at voipbusiness.us>
wrote:
> Please keep the ?me to? emails off the list.
>
> Regards;
>
> JV
>
>
>
> *From:*
2016 Feb 17
2
1000 analogue lines with asterisk
On Wed, Feb 17, 2016 at 8:14 AM, Mitul Limbani <mitul at enterux.in> wrote:
> Sangoma 50 port FXS
Thanks.
Will I now stack 20 boxes in order to achieve the 1000 FXS lines?
Regards
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2012 Jun 02
1
Asterisk pickup call on first ring
Hello,
Currently my asterisk system pickup incoming call after 3 or 4 rings.
How can I ask it to answer the call on the first ring? I put
immediate=yes on /etc/asterisk/chan_dahdi.conf but result in no
different.
Thanks in advance :)
BR,
Anam
--
Sent from my mobile device
2010 Apr 13
2
SNOM M9 base station A to base station B
<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN">
<html>
<head>
</head>
<body bgcolor="#ffffff" text="#000000">
<small><font face="Helvetica, Arial, sans-serif">Hello,<br>
<br>
I have a question concerning SNOM M9 base station.<br>
<br>
If my customer places a SNOM M9 base
2013 May 14
4
dial and bridge
Hi all,
I need some advice - I have been working on originating multiple calls
using AMI and then joining them.
What I want to do is:
- dial call 1 (where the caller is in a "channel" format, like SIp/1234 or
Local/1234 at ext) and "park" it somehow
- dial call 2 (where again the caller is in channel format) and join it to
the previous call.
As a requirement, I cannot use the
2016 Feb 17
2
1000 analogue lines with asterisk
Thanks Mitul,
The server spec is okay but I need information on the fxs hardware to use.
Regards
On Wed, Feb 17, 2016 at 8:07 AM, Mitul Limbani <mitul at enterux.in> wrote:
> Quad core Xeon with 4GB ram
> On Feb 17, 2016 12:32 PM, "Goke Aruna" <goksie at gmail.com> wrote:
>
>> Hello all,
>> Can someone recommend what hardware to use for a 1000 analogue
2013 Jun 14
1
SIGTRAN Integration
Hello Everyone,
I was wondering how SIGTRAN/SS7oIP can fit into our currently 100% SIP model.
We are looking to interconnect with the PSTN world, and our supplier
has given us
a few options. We can either do this over traditional PRIs, A-Links or
the SS7IP new.
I am really interested in SIGTRAN, and was wondering how some of you
have integrated
it into your architecture. Can Asterisk handle
2013 Nov 08
1
Asterisk 1.8.22
Hello, I have a fully functional Asterisk Server, I want to configure this
server to be able to process call from Skype, can someone point me to a
howto? or if there are suggestions on best way to approach this problem.
Thanks,
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2014 Sep 18
1
Voice-Recognition / ASR / with barge in
Hi there,
I am using Asterisk 11.9 (with Sangoma-E1-Card/DAHDI) and it works fine
:-) But I am wondering if there is a solution/application which will
enable me to implement voice recognition while playing a voice file
(barge in). So that the caller hears a voice file and can interrupt it
with his voice.
Currently (on our platform) the caller has to wait for the end of the
voicefie. Then we play
2014 Mar 05
3
Enterprise VoIP Trunk
Am looking for a service provider who can provide enterprise SIP trunk with
100 channels concurrent sessions.
I see some like Inphonex, Broadvoice... and etc....
Is there any suggestions for the service providers.
Regards
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2013 Mar 15
2
app_rtsp.c ported to Asterisk 11.x
Hi,
If anyone is interested, I have ported app_rtsp.c to Asterisk 11.x. I have
tested it with GStreamer RTSP server and a C920 webcam streaming H264 SVC
video from one machine to another machine running Linphone. Contact me at
this e-mail address robkrakora at messagenetsystems.com for source code.
Best Regards,
--
Rob Krakora
MessageNet Systems
101 East Carmel Dr. Suite 105
Carmel, IN 46032
2020 May 25
2
Asterisk : CDR Analyzer Updated
Everybody,
I've been using the old Asterisk CDR Areski GUI CDR-Stats for at least a
dozen years, it was easy to configure and didn't requite installing
'connectors' on anything or adding tables on the DB server.
It's based off of PHP5 and the only reason I still keep around a Debian
7 system, since it won't work with the newer PHP7.
A friend of mine is learning PHP7
2014 Feb 04
2
Connect to remote GW
If SIP channel driver needs to connect to a remote GW over a dedicated SIP trunk BUT the remote GW has a 'standby' in case of failure, how can the sip configuration file be configured for the remote GW when there are actually two IP addresses. If the main remote GW fails control automatically switches to the standby GW, so how could the SIP configuration file hande this switch and support
2012 Jul 24
5
DAHDI problems
Is a normal functionality?
when I do
#dahdi_cfg -vvvvvv
In my Asterisk console shows this....
[Jul 24 13:39:08] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
If I do this a lot of times...then
[Jul 24 13:39:20] NOTICE[30263]: chan_dahdi.c:9461 pri_dchannel: PRI got
event: HDLC Abort (6) on Primary D-channel of span 1
[Jul 24
2015 Apr 07
2
OpenVZ with asterisk 13
On 04/07/2015 10:48 AM, Johan Wilfer wrote:
> Den 2015-04-07 15:41, Ikka Tirtawidjaja skrev:
>> Dear all,
>>
>> Is anyone has experience making Asterisk server with virtual server
>> OPEN-VZ (in proxmox 3.4 box) ?
>>
>> My boss want to build a production server with it, and it will have +/-
>> 300 sip user (concurrent call maybe < 150 call)
>>
2015 Feb 22
2
dialplan contexts syntax and terminology
I'm looking into the dialplan specifics:
tleilax:~ #
tleilax:~ # cat /etc/asterisk/extensions.conf
[general]
static=yes
writeprotect=no
[globals]
CONSOLE=Console/dsp ; Console interface for
demo
TRUNK=DAHDI/r1 ; Trunk interface
TRUNKX=DAHDI/r2 ; 2nd trunk interface
TRUNKIAX=IAX2/ASTtest1:test at 10.10.10.16:4569 ; IAX trunk
2014 Apr 21
3
Open Source Asterisk Polling Solution
Hello Everyone,
We are looking for a simple open source auto dialer with "polling"
capabilities. What we would like is a program that we can upload
leads to, and have asterisk:
i) Dial numbers
ii) Play pre-recorded
iii) If user presses one, forward the call to an agent
There are so many solutions out there it's hard to make a decision on what
works, what has just a limited free
2014 Jun 24
1
Redfone FoneBridge2 Quad T1/E1 Alternative
We have been using Red-fone foneBridge2 Quad T1/E1 for last few years.
As these devices are not available anymore, we are looking for alternatives.
Are there any similar devices available ?
--
Regards,
Tirveni Yadav
www.udyansh.org
What is this Universe ? From what it arises ? Into what does it go?
In freedom it arises, In freedom it rests and into freedom it melts away.
Upanishads.