No need for thirdlane or any proprietary extensions .
All you need is jssip open source that works with webrtc.
Just point it to wss of asterisk.
No restrictions nor min or max number and best of all it is free open source
and web based (no software installation).
Also here is another web phone th
Agree and that should be avoided.
Sent from my iPhone
> On Apr 30, 2017, at 5:54 PM, Tech Support wrote:
>
> I thought this was a non-commercial list.
>
>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf
iPhone and android : growndwire
Also you have media5 works well for iPhone
Linphone for iOS,android and Windows
Jitsi for windows works very well.
Sent from my iPhone
> On Apr 29, 2017, at 5:35 PM, Thomas wrote:
>
> Hello,
> Iam lookong for an Softphone for iPhor oder Android smartphone us
Sorry forgot to attach the CLI trace:
=
CLI> pjsip show aors
Aor:
Contact:
=
Aor: 210220
Aor: 2103
Hi,
after a long pause (Asterisk 1.8 times), I have started again playing with
VOIP. A lot has changed since last time I did setup an Asterisk system!
So I am asking for some help.
PJSIP seems tougher..
So my problem is that I do have a test system up in the cloud, behind a
Hello,
I am trying to get CDR works for my asterisk installation.
My OS is Ubuntu 15 with asterisk 13.8 compiled locally on the machine.
MYSQL Server version is 5.6.28-0ubuntu0.15.04.1 (Ubuntu)
I also have another machine Ubuntu 15.04 same os but with asterisk 13.8.1
having the same issu
Hello,
I am trying to get CDR work for my asterisk installation.
My OS is Ubuntu 15 with asterisk 13.8 compiled locally on the machine.
MYSQL Server version is 5.6.28-0ubuntu0.15.04.1 (Ubuntu)
Would appreciate if someone can help solving this issue
The error that I am getting:
[2
I did using acrobits groundwire on asterisk 13.7.2
Had to add a statement in pjsip.endpointxxx
I do not have it in mind but can look it up for you tomorrow.
Sent from my iPhone
> On Jul 8, 2015, at 9:05 PM, ricky gutierrez wrote:
>
> Hi list , I'm doing some tests with asterisk 13.4 and tls, a
Hello,
You need to determine the correct MTU value by doing the following:
ping www.google.com -f -l 1400 and you go up or down
An example:
1440 Max packet size from Ping Test
+ 28 IP and ICMP headers
1468 is your optimum MTU Setting
Reference : http://www.tp-link.fr/FAQ-190.htm
Please set correct MTU at server side, it is definitely an MTU issue.
Sent from my iPhone
> On Mar 3, 2016, at 5:31 PM, Olivier wrote:
>
> Hello,
>
> I'm remotely managing an asterisk setup using an OpenVPN client on this
> Asterisk box, connecting to an OpenVPN server of mine).
>
> This b
Hello,
I am interested.
Regards
Toufic
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Helvio Junior
Sent: Monday, June 22, 2015 5:36 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Product CDR/Queue/Meetme
G
Beronet Gateway BFSB2HY , it works well for me two.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Markos Vakondios
Sent: Tuesday, May 26, 2015 4:00 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asteris
is no longer working with Firefox after
upgrade to version 37
Toufic Khreish (Gmail) wrote:
> Hello,
>
> Webrtc stopped after upgrading firefox from version 36 to version37.
> I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3
> and firefox version 36 without a
Hello,
Webrtc stopped after upgrading firefox from version 36 to version37.
I have been running webrtc with freepbx 12 and asterisk 13.2 or 13.3 and
firefox version 36 without any issues until firefox was upgraded to version
37.
Unfortunately Chrome works well in one direction (from chrome to any
Hi,
I have tried Groundwire on IOS , and Android Alcatel (voice and video calls
with asterisk 13.3)
Also tried Bria on both OS in video and voice.
Regards
Toufic
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sevana Oy
Hello,
I have a weird problem between Asterisk 13.3 and a Yeastar U200 pbx over IAX
trunk.
Should I call from Yeastar to my asterisk 13.3 the call goes through without
issues.
Should I call from asterisk 13.3 to Yeastar I can hear a ring tone however
the yeastar does not show any activities.
On th
Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
wrote:
> I see that my asterisk is started with the -g option, the core file I
> cannot find on my system (find / -name core*
Behalf Of Matthew Jordan
Sent: Tuesday, March 17, 2015 11:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
wrote:
> I see that my asterisk is started with th
Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
On Tue, Mar 17, 2015 at 5:53 PM, Toufic Khreish (Gmail)
wrote:
> I see that my asterisk is started with the -g option, the core file I
> cannot find on my system (find / -name core*)
>
I woul
:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
On Mon, Mar 16, 2015 at 6:12 PM, Toufic Khreish (Gmail)
wrote:
> Hello Matthew,
>
> I have compiled Asterisk 13.2 with the following compiler Flag
] Asterisk 13.2.0 Video issues
On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail)
wrote:
> Thank you, I needed a starting point to start my post.
>
> 1. Asterisk 12.8.1 (IAX2 voice issues) no video issues.
> Voice issues on IAX2 Trunks, All extensions are SIP.
> The IAX2 trun
Matthew Jordan
Sent: Thursday, March 12, 2015 3:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
On Tue, Mar 10, 2015 at 5:22 PM, Toufic Khreish (Gmail)
wrote:
> Thank you, I needed a starting point to start my post.
&g
concerns GXV3175 for the moment (with the
res_format_attr_h264.so removed). (GXV3175 version Hardware : 1.4A ,
program version: 1.0.3.76 and CPE version 1.0.1.32)
Any idea why ? and how could this be fixed ?
-Original Message-
From: Toufic Khreish (Gmail) [mailto:toufic.khre...@gmail.com
ehalf Of Matthew Jordan
Sent: Tuesday, March 10, 2015 1:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Asterisk 13.2.0 Video issues
On Tue, Mar 10, 2015 at 4:15 AM, Toufic Khreish (Gmail)
wrote:
> I recently compiled asterisk 13.2.0 on an RK328
I recently compiled asterisk 13.2.0 on an RK3288 , I am suspecting problems
with the format H264, Asterisk 12.8.1 compiled on the same hardware is
behaving very well for the same format H264
Problem of asterisk 12.8.1 is IAX2 trunk bad voice quality.
Could someone investigate the problem of Aster
It can be done, contact me offlist to discuss further
Frank
Sent from my iPhone
> On Jan 14, 2015, at 7:32 PM,
> wrote:
>
> Hello All,
>
> Please advise kindly about the following arrangement:
>
> I need to have Asterisk working with company's mobiles via company's WiMax
> mobile network.
Or educate him !
Sent from my iPhone
On 2013-07-06, at 3:03 PM, Steve Edwards wrote:
> On Sat, 6 Jul 2013, William Muriithi wrote:
>
>> Better to look for alternative product if your employer can't stomach one
>> Linux box in your office.
>
> Better to look for an alternative employer :)
>
TA-v4,
> relay=localhost [127.0.0.1]
> Jun 20 16:17:17 SERVER-NAME sendmail[21276]: r5KLHGgk021276:
> to=earohua...@gmail.com, ctladdr=root (0/0), delay=00:00:01, xdelay=00:00:01,
> mailer=relay, pri=146501, relay=[127.0.0.1] [127.0.0.1], dsn=2.0.0, stat=Sent
> (r5KLHGNY02128
I believe there are options for rtp / audio..
Look at pcap play and rtp echo...
Transcoding would be another beast - if you are allowing it
Sent from my iPhone 5
On May 22, 2013, at 10:02 AM, Tommy Cooper wrote:
> From the little experience I have I do not think that that is a good way of
>
We found this URL: http://sourceforge.net/projects/asteriskvideo/
But these applications seem too old for Asterisk 11.
Are there any video applications for Asterisk 11?
We need these applications to implement IVVR.
Or any other solution is to be appreciated.
Thanks in advance.
--
_
Might also want to check the google hasnt detected an unusual login and is
asking for the ip to be accepted.
Log in to gmail with that account and check
Sent from my iPhone 5
On Feb 5, 2013, at 4:31 PM, Joshua Colp wrote:
> Josue Freitas wrote:
>> Thank you!
>>
>>
guess the next step is to maybe use AGI
On Mon, Jan 21, 2013 at 5:10 PM, Al Efron [gmail] wrote:
> Hi All,
>
> Anyone know how to use the function DB_KEYS()?
>
> Info on this is non-existant on the net incl. the wiki and there are
> absolutely NO examples of it anywhere. I was
Hi All,
Anyone know how to use the function DB_KEYS()?
Info on this is non-existant on the net incl. the wiki and there are
absolutely NO examples of it anywhere. I was hoping that unlike the other
DB functions, this is able to get the Key for a given Value OR at least
list ALL keys of a given Fa
Sometimes just the act of collecting performance data degrades the quality
Sent from my iPhone 5
On Jan 6, 2013, at 6:00 AM, XBrian wrote:
> Thanks
>
> What would you use to measure jitter / packetloss in real time?
>
>
> --
> _
Asterisk "sip show peers" lists the qualify value in ms (milliseconds).
Please read up on this and the setting for it in sip.conf config file
Sent from my iPhone 5
On Jan 5, 2013, at 5:30 AM, XBrian wrote:
> Joachim, thanks for the reply
> - delay you can somewhat estimate prior to the call (w
Good luck! Finding the right person at VZ has always been a beef of mine
Sent from my iPhone 5
On Jan 5, 2013, at 11:12 AM, Logan Bibby wrote:
> Does anyone have a good contact for their sales? I've attempted calling their
> Enterprise sales a few times and was just spun around in circles. Ha
isk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
> *Sent:* Saturday, May 26, 2012 5:13 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion; FreeSWITCH
> Users Help
> *Subject:* [asterisk-users] Common/Reas
Hello All,
just throwing this out there. What are people generally using these days
when designing their services, esp. those that require a user to call a DID
to access their system, similar to calling card services. There was a time
when this used to be 50 to 1 for DIDs, and about 10 to 1 for nu
On Mon, Mar 12, 2012 at 6:52 PM, Markus wrote:
> Hi,
>
> this question is not Asterisk specific, but since there are so many
> experts present on this list, maybe its OK to ask anyways.
>
> I'm having a hard time "normalizing" rate sheets from different providers.
> What I mean with this: the goa
mples on the web,
but nothing happens. I'm on 1.8.5.0.
BTW, there is no Google Voice involved. and I'm calling from from a gmail
based gtalk client. Also, I can successfully make an outbound call. Just the
inbound isn't working :( Any help please?
Currently my incoming dial-plan i
On Thu, Jun 23, 2011 at 7:58 AM, Tim Panton wrote:
>
> On 15 Jun 2011, at 23:29, Kevin P. Fleming wrote:
>
> > On 06/15/2011 04:40 PM, Elliot Murdock wrote:
> >> Hello,
> >>
> >> Yes, the issue I am having is currently only with Google Talk. Wonder
> >> if what development will be made to fix th
On Wed, Jul 6, 2011 at 7:50 AM, Tzafrir Cohen wrote:
> On Wed, Jul 06, 2011 at 07:11:26AM -0400, A E [Gmail] wrote:
> > On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen >wrote:
> >
> > > On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:
>
> > > &
On Wed, Jul 6, 2011 at 7:02 AM, Tzafrir Cohen wrote:
> On Wed, Jul 06, 2011 at 06:15:26AM -0400, A E [Gmail] wrote:
> > On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen >wrote:
> >
> > > On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
> > > > hel
On Wed, Jul 6, 2011 at 3:21 AM, Tzafrir Cohen wrote:
> On Tue, Jul 05, 2011 at 08:30:52PM -0400, A E [Gmail] wrote:
> > hello people,
> >
> > I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
> some
> > reason I have noticed that only after
;m going to upgrade the version to 1.8.4.4 and
see what happens
> **
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
> *Sent:* Wednesday, July 06, 2011 9:50 AM
> *To:* Asterisk Users Maili
k?
I want to see if that's the cause of the CPU usage and I'll lose that if I
restart Asterisk.
Thanks
> 2011/7/5, A E [Gmail] :
> > hello people,
> >
> > I am running v1.8.4.2 on debian squeeze on a sparc platform...and for
> some
> > reason I have n
hello people,
I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some
reason I have noticed that only after a few test calls, the asterisk process
is running between 95% - 99.9% CPU when there's absolutely nothing on the
system. This is a clean Asterisk system in an internal net
On Mon, Jun 6, 2011 at 10:10 AM, Steve Edwards wrote:
> On Mon, Jun 6, 2011 at 2:26 AM, Steve Edwards
>> wrote:
>>
>> I strongly suggest using an existing library for the language of your
>> choice.
>>
>
> On Mon, 6 Jun 2011, A E [Gmail] wrote:
>
> C
On Mon, Jun 6, 2011 at 10:39 AM, Tony Mountifield wrote:
> In article ,
> A E [Gmail] wrote:
> > Hello,
> > using 1.8.4. using a very simple local AGI script in bash which has only
> one
> > line in it:
> >
> > echo -e 'STREAM FILE welcome 123 \n&#
STDIN.
>>
>
> On Mon, 6 Jun 2011, A E [Gmail] wrote:
>
> Right! I did read that, the problem is how do I do this in bash?? I tried
>> read the result in and just post a Noop kind of a thing just to tell that I
>> read something, but it didn't help. I also explicitly d
On Mon, Jun 6, 2011 at 2:06 AM, Steve Edwards wrote:
> On Mon, 6 Jun 2011, A E [Gmail] wrote:
>
> Hello,using 1.8.4. using a very simple local AGI script in bash which has
>> only one line in it:
>>
>> echo -e 'STREAM FILE welcome 123 \n'
>>
>
On Mon, Jun 6, 2011 at 2:06 AM, mahesh katta wrote:
>
>
> On Mon, Jun 6, 2011 at 9:42 AM, A E [Gmail] wrote:
>
>> Hello,
>> using 1.8.4. using a very simple local AGI script in bash which has only
>> one line in it:
>>
>> echo -e 'STREAM FILE wel
On Mon, Jun 6, 2011 at 12:12 AM, A E [Gmail] wrote:
> Hello,
> using 1.8.4. using a very simple local AGI script in bash which has only
> one line in it:
>
> echo -e 'STREAM FILE welcome 123 \n'
>
> dialplan:
> exten => 5150,1,Answer()
> same => n,Set
Hello,
using 1.8.4. using a very simple local AGI script in bash which has only one
line in it:
echo -e 'STREAM FILE welcome 123 \n'
dialplan:
exten => 5150,1,Answer()
same => n,Set(CHANNEL(language)=en_AU)
same => n,AGI(testagi.sh)
same => n,Hangup
console output:
-- Executing [5150@A
On Wed, May 18, 2011 at 9:39 PM, A E [Gmail] wrote:
> On Wed, May 18, 2011 at 9:29 PM, Paul Belanger wrote:
>
>> On 11-05-18 08:01 PM, A E [Gmail] wrote:
>>
>>> boxb*CLI> dialplan show Test
>>> [ Context 'Test' created by 'pbx_config&#
On Thu, May 19, 2011 at 3:19 AM, GNUbie wrote:
> Anyone? Please advice. Thank you.
>
> That's WAYY too much info for me to go through right now, and I don't know
anything about TLS registration but what I would ask for is if you have the
following lines in your sip.conf
domain=:
so in your case
On Wed, May 18, 2011 at 9:29 PM, Paul Belanger wrote:
> On 11-05-18 08:01 PM, A E [Gmail] wrote:
>
>> boxb*CLI> dialplan show Test
>> [ Context 'Test' created by 'pbx_config' ]
>> '' => 1. Answer()
>&
Hello All,
This is probably another one of those completely silly questions that I'm
going to hit myself later on, but I have the simplest issue right now but I
can't figure out why it's happening:
I have a trunk from one * box (box a) to another * box (box b)
the call comes in from box a with a
On Mon, May 16, 2011 at 10:20 PM, Shaun Ruffell wrote:
> On Mon, May 16, 2011 at 09:26:48PM -0400, A E [Gmail] wrote:
> >
> > following this advice, is there a quick and minimal way to install/use
> > res_timing_dahdi without having to build/compile/install the whole dahd
On Mon, May 16, 2011 at 10:27 AM, satish patel wrote:
> Thanks Leif,
>
> I had changed it to res_timing_dahdi and since last few days it seem good.
>
> -S
>
> > Date: Sun, 15 May 2011 15:48:03 -0400
> > From: leif.mad...@asteriskdocs.org
> > To: asterisk-users@lists.digium.com
>
> > Subject: Re:
On Mon, May 9, 2011 at 7:58 PM, Markus wrote:
> Hi,
>
> > new to the list. Wondering if anyone has / knows of, a good rate importer
> > tool that can be used to standardize and normalize the ratesheets / rate
> > decks etc. obtained from various carriers so they can be analysed and
> > imported i
On Mon, May 9, 2011 at 3:05 PM, Jason Aarons (AM) <
jason.aar...@dimensiondata.com> wrote:
> I know most billing software sell this as a monthly service. You get
> cd-rom every month where they have collected the published tarrif tables
> filed with the FCC. You load it on the software to analyz
Hi All,
new to the list. Wondering if anyone has / knows of, a good rate importer
tool that can be used to standardize and normalize the ratesheets / rate
decks etc. obtained from various carriers so they can be analysed and
imported into a DB or be saved as a CSV or something?
Thanks so much in
On Tue, May 3, 2011 at 5:19 AM, Tzafrir Cohen wrote:
> On Tue, May 03, 2011 at 01:09:14AM -0400, A E [Gmail] wrote:
> > On Mon, May 2, 2011 at 9:45 PM, C F wrote:
> >
> > > Just from my experience with different DBs, stay away from BLOB data
> > > types as mu
On Tue, May 3, 2011 at 4:41 AM, Thorsten Göllner wrote:
> Am 02.05.2011 15:59, schrieb A E [Gmail]:
>
> On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] wrote:
>
>> Hello All,
>>
>> Probably a silly question, but we're wondering if people have had any
>> e
On Mon, May 2, 2011 at 9:45 PM, C F wrote:
> Just from my experience with different DBs, stay away from BLOB data
> types as much as possible.
>
> Hi CF,
any particular reason why? I've had a good experience with it, in fact
that's recommended by DB developers when it's a case of small files. The
On Mon, May 2, 2011 at 3:23 PM, Danny Nicholas wrote:
>--
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
> *Sent:* Monday, May 02, 2011 1:23 PM
> *To:* Asterisk
On Mon, May 2, 2011 at 2:30 PM, Danny Nicholas wrote:
>--
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *A E [Gmail]
> *Sent:* Monday, May 02, 2011 1:23 PM
> *To:* Asterisk
>
>
>> Just realised that this can better be described another way:
>
> What we're essentially trying to do is be able to do any one of these
>
> a) stream an audio/video file stored in the DB via AGI into the current
> channel so that it plays on the phone
>
> OR
>
> b) Do something like what Real
On Mon, May 2, 2011 at 3:15 AM, A E [Gmail] wrote:
> Hello All,
>
> Probably a silly question, but we're wondering if people have had any
> experience and have data to demonstrate if the performance of the Asterisk
> system might suffer in terms of latency etc. if we&#x
On Mon, May 2, 2011 at 12:07 AM, Kaushal Shriyan
wrote:
>
> Hi Jim,
>
> Thanks for the explanation, I have couple of questions here.
>
> 1) Does the xorcom box support *8 Port PRI E1 Interface*. ?
> 2) Also the Primary and Secondary Asterisk Server can be any server which
> will run Asterisk or As
Hello All,
Probably a silly question, but we're wondering if people have had any
experience and have data to demonstrate if the performance of the Asterisk
system might suffer in terms of latency etc. if we're to have it retrieve
sound files from a database using odbc as opposed to storing them lo
i have this configuration , An Asterisk server connected to my private LAN
192.168.10.0/24 when i do port forwarding for port 5060 so that i make a call
from Internet into Asterisk wireshark show the message "destintion port
unrechable"
i configured sip.conf for "nat=yes" and "qualify=yes" and
does 3CX compare to Asterisk in anyhow? it is based on windows and it seems
that it is more easier to configure than Asterisk , however i think the
complexity of Asterisk configuration comes with its flexibility , am i right?___
-- Bandwidth and Colocat
re also in sip.conf
(allow=ulaw,alaw,gsm,h263)
--
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of gmail
Sent: Thursday, June 25, 2009 12:57 PM
To
i am trying to make a video call on asterisk 1.6 , my configuration is an
- asterisk 1.6 on Centos on virtual machine VmWare
- Xlite softphone one windows xp (the Host operating system)
- X-lite client on another windows XP (the Guest operating system )
i put the paramtervideosupport=yes
channels no aparece nada.
2009/4/2 Brandon B. :
> nos muestran la configuración de sus líneas de /etc/dahdi/system.conf y
> /etc/asterisk/chan/dahdi.conf
>
>
>
> 2009/4/2 Manolet Gmail
>>
>> Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic
>>
>
On Thu, Apr 2, 2009 at 4:38 PM, Tzafrir Cohen wrote:
> On Thu, Apr 02, 2009 at 03:51:21PM -0500, Manolet Gmail wrote:
>> I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic
>>
>> i want to configure a x100p card an use it with asterisk, so i download,
>> compile
Tengo en una maquia ubuntu 8.10 el kernel es Linux 2.6.27-14-generic
Quiero configurar una tarjeta x100p i usarla con asterisk, asi que
descague compile e instale lo siguiente:
asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9
Sin embargo no logro configurar la tarjeta con exi
I have an ubuntu 8.10 machine. Linux 2.6.27-14-generic
i want to configure a x100p card an use it with asterisk, so i download,
compile and install:
asterisk-1.4.24
dahdi-linux-2.1.0.4
dahdi-tools-2.1.0.2
libpri-1.4.9
i try almost everything i found on the net but without success:
if i run lspc
Hi, I got a card from Digium TDM with 2 FXO modules (red ones). There is a
problem that has me quite upset and is that asterisk always detect tones
repeated two, three or more times.
i mean, if i press 123 on my phone. asterisk detects somethin like:
111223
or 112333
or things like that.
Have, i want to create a sip extension to a context in my dialplan.
how i can do that?
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net
as
Hi, i want to make a direct ip to ip call (without a sip proxy), what
software i can use (windows)? i try with xlite but dont understand how
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSC
Hola a todos, estoy creando una comunidad de asterisk en español que
se dividira en un blog y un foro, estoy buscando gente que quiera
ayudarme a escribir articulos para el blog, y claro, pueda participar
en el foro.
Si a alguien le interesa saber mas escribanme un mail.
[EMAIL PROTECTED]
__
Hi, im a new user to asterisk. i have configured one box using asterisknow.
now i want to enable *9 (or some code) to play for example tt-monkeys.
i read a lot in voip-info but cant do it:
i have this on my features.conf:
[applicationmap]
testfeature => *9,callee,Playback,tt-monkeys
extensions
Hello all,
this might be a crazy question
can I connect 2 FXS plugs to the same analog phone ?
my reason: I'm expecting that, with this setup, the phone could operate
transparently through the redundant FXS if the main FXS would fail... of
if asterisk is stopped on one of the servers...
the
Does anybody know how to off-load an Asterisk Box so that to distribute its
functions like IVR and VoiceMail or its PTSN gateway function into different
servers? in this case , will the installation of Asterisk on each server
differe and how these different servers will interact as a single log
Is Asterisk practically stable and reliable for a larg Enterprise has say a
1 phones , is there any case study like this?___
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Hi all,
I want to use a cell phone as my FXO line to Asterisk Box ,did anyone try
this and configured it and how to physically connect it to Asterisk server?___
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asterisk-users mai
Is there any ISDN PCI cards that can be used with Asterisk as a PSTN gateway
instead of using Diguim FXO cards?___
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Does anyone know how to off-load an Asterisk Box so that to distribute its
functions like IVR and VoiceMail or its PTSN gateway function into different
servers? in this case , will the installation of Asterisk on each server
differe and how these different servers will interact as a single logi
Does anybody know how to off-load an Asterisk Box so that to distribute its
functions like IVR and VoiceMail or its PTSN gateway function into different
servers? in this case , will the installation of Asterisk on each server
differe and how these different servers will interact as a single log
Hi marek,
gr8. I am working on chan_ss7 now..
Regards,
Joel
- Original Message -
From: "marek cervenka" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Monday, March 03, 2008 3:55 AM
Subject: Re: [asterisk-users] chan_ss7 0.10
>> Thanks for t
Hi marek,
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross
chan_ss7 but found sifira is not in active development. But is this
chan_ss7 stable and can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7 signalling u
Hi marek,
Thanks for the update.
I have Sangoma A104D and wanted to use ss7 signalling. I came accross chan_ss7
but found sifira is not in active development. But is this chan_ss7 stable and
can be used in production server implementation.
We are going to have 2 to 3 boxes with ss7 signalling
X-Lite do what you need...
On 8/6/07, Joao Pereira <[EMAIL PROTECTED]> wrote:
> Hello
> I need a Softphone with auto answer where users can't turn it off.
> Does someone knows a softphone where users can't turn the auto answer off?
> Or is there any way Asterisk could force the clients to answe
Do you have MySQL installed in your machine???
On 6/21/07, Khaled Chehab <[EMAIL PROTECTED]> wrote:
>
>
>
>
> No one faced a problem like this !!
>
>
>
>
>
>
> From: Khaled Chehab [mailto:[EMAIL PROTECTED]
> Sent: Thursday, June 21, 2007 12:37 AM
> To: 'As
In [general] section:
externip=your_extern_ip_address
localnet=your_local_net/bits i.e. 192.168.0.0/24
Try this...
On 6/12/07, Rob Schall <[EMAIL PROTECTED]> wrote:
We are trying to use a softphone from a location that is behind a firewall.
We are using x-lite as the softphone.
So far
Hello Drew;
Assuming your extensions is 105 let's see the dialplan:
exten => 105,1,Dial(SIP/105,30,Tt)
exten => 105,n,Hangup
exten => *XXX,1,Answer
exten => *XXX,n,VoiceMail(${EXTEN:[EMAIL PROTECTED])
exten => *XXX,n,Hangup
I think this should work for what you want.
Regards;
Leonardo Kamac
Yes from Brazil...
On 6/6/07, Ed Nuñez <[EMAIL PROTECTED]> wrote:
Is anyone else having trouble going into voip-info.org today?
___
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