Hi All,
I'm trying to use the orginate cmd.
I have it working if originate is from a user e.g. SIP/
originate SIP/ extension 987654...@outbound-route
What i'd like to be able to is instead of a local extensions i would
call an outside number then connect it another outside number.
On Tue, Mar 24, 2009 at 2:14 AM, Michael Graves mgra...@mstvp.com wrote:
Amen to that! Unles you have some compelling reason for VoWifi it's not
worthy of consideration. Especially for SOHO or small biz use. Too
costly to do well.
I have never understood why anyone would use wifi just to get
On Tue, Mar 24, 2009 at 2:11 AM, Zeeshan Zakaria zisha...@gmail.com wrote:
I was wondering if somebody maintains a list of these IP addresses which
everybody can block in their firewalls. And is there a place I can publish
these IP addresses?
We were just talking about this and I remembered
On Monday 23 March 2009 20:11:45 Zeeshan Zakaria wrote:
Hi,
In last one week I have seen two servers of our organization successfully
hacked and some other under attack from some other IP addresses. We would
block one IP address on our firewall and after a few hours, they would
start getting
Hello,
Does anyone know why I am unable to retrieve the Redirecting Number?
I've done a pri debug span 1/1 and can see the number being passed
correctly to Asterisk:
Redirecting Number (len=15) [ Ext: 0 TON: National Number (2) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1)
Hi,
i do have a request for an installation with about 1800 sip extensions -
as addon to a exisiting system - connected to it using qsig. The
requirement here is also that the system should have SIP over TCP with
TLS and SRTP (snom phones should get supported)
I know there are patches out there
On Tue, Mar 24, 2009 at 8:10 AM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
There are 4 billion possible IP addresses. To successfully block all possible
hackers, you must block 4 billion of them. Seriously. Even your own computer
is a possible source of hacking to other
On Mon, 2009-03-23 at 23:15 +, Steve Howes wrote:
On 23 Mar 2009, at 22:44, Hans Witvliet wrote:
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
concrete shielding wireless phones.
So i was thinking to
About two and a half years ago, I upgraded a small call centre from
corded handsets to X-Lite with Plantronics CS60 USB headsets.
X-Lite lasted about two or three months before we ditched it in favour
of Eyebeam. X-Lite disables too many features to be useful. With the
Plantronics headset,
Hi to all the ML. I'm new here. I start to use asterisk with realtime
configuration, with pgsql backend connected via odbc. The connection
between asterisk and pgsql works fine. I create a table sip_conf with
2 user (for testing purpose), 1401 and 1501. Those are the records:
asterisk= SELECT
Hello,
I configured both asterisk and grandstream 2000 accourding to howtos on
the web..
And everything seems working fin.
But if i reload asterisk grandstream stops working with BLF.
I need to restart the phone to enable BLF again.
Any clues??
___
--
On 23 Mar 2009, at 19:42, Gordon Henderson wrote:
Anyone connected up to it yet?
http://www.skypeforsip.com/
It would seem to make Digiums chan_skype rather pointness, or am I
missing
something?
Or is this Digiums chan_skype in a hosted box somewhere?
Gordon
There are fewer
Hi, is there a Asterisk 1.6 billing software that i can use?
Prepaid supporting ones are more acceptable for me right now but,
post-paids are also welcome if they are available.
As i see most of the softwares are designed for 1.2 and 1.4
___
--
On Tue, Mar 24, 2009 at 9:37 AM, Tim Panton t...@westhawk.co.uk wrote:
There are fewer limitations to SFA than SFS. SFA gets presence and full user
info, plus it can
make calls to Skype users, which SFS cant.
I'm hoping that Digium will extend this difference by adding support for
text and
On Mon, 23 Mar 2009, Hans Witvliet wrote:
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
concrete shielding wireless phones.
So i was thinking to use one of those usb-phones, and plug it into one
of my servers
On Mon, 23 Mar 2009, Zeeshan Zakaria wrote:
Hi,
In last one week I have seen two servers of our organization successfully
hacked and some other under attack from some other IP addresses. We would
block one IP address on our firewall and after a few hours, they would start
getting hits from
On 24 Mar 2009, at 09:52, Gordon Henderson wrote:
On Mon, 23 Mar 2009, Hans Witvliet wrote:
While reading the thread about recommending usb-phones...
Once in a while, i'm in a data-centre, no normal phones, and too much
concrete shielding wireless phones.
So i was thinking to use one of
On Mon, 23 Mar 2009, Kelvin Chan wrote:
One of our local companies here in the UK are trialling a new conference
phone - the Konftel 300IP SIP however it's still as expensive as a
Polycom, but that might be the $/£ exchange - might be cheaper where you
are?
It seems like an interesting
Good morning everybody.
My question is simple.
Is there a way to perform relay register with Asterisk ?
More precisely, I want my clients regiter to a Proxy Registrar
(OpenSIPS/Kamailio) through my Asterisk :
REGISTER REGISTER
Client Asterisk
I am not really sure, but apparently they guessed a SIP username/password.
But what I don't understand is they even though I deleted that extension all
together, still 'sip show peers' showed that extension. Then I figured out
an easy to guess manager user and password, which I also deleted. I
Not sure about this. It seems you are trying to find a solution to a
problem which you do not actually describe.
I.E, you have problem X, you think that doing Y might be the solution,
but you don't know how to do Y (and in this case, neither do I).
How about exposing underlying problem X to the
Use the Local/ channel type(?)
Local/0123456...@outbound-route
2009/3/24 Nhadie nha...@gmail.com
Hi All,
I'm trying to use the orginate cmd.
I have it working if originate is from a user e.g. SIP/
originate SIP/ extension 987654...@outbound-route
What i'd like to be able to is
Hmm no, it is exactly what I want to do, not in order to solve an other problem.
In a more global context, I am trying to study if asterisk can act as a Session
Border Controller.
If I ask Asterisk in the sip.conf file to manually register to the Proxy
Registrar, it works for incoming and
Then, I don't know :-)
Seems you are looking for a way to have a distributed architecture.
The way I would do it is to let asterisk handle the registrations and
then use something like ENUM or DUNDi (more likely ENUM since it's a
more recognized standard) to know where the call should be going.
Yes. Grandstreams suck.
Oguzhan Kayhan wrote:
Hello,
I configured both asterisk and grandstream 2000 accourding to howtos on
the web..
And everything seems working fin.
But if i reload asterisk grandstream stops working with BLF.
I need to restart the phone to enable BLF again.
Any
-Original Message-
boun...@lists.digium.com] On Behalf Of Rob Hillis
Yes. Grandstreams suck.
[Cary Fitch] We are not entitled to your opinion.
[Cary Fitch] On a small development/production system, undergoing intensive
development, we reload continuously, 50
We get this error message
[Mar 23 10:10:09] WARNING[4325]: chan_iax2.c:1056 __send_lagrq: I was
supposed to send a LAGRQ with callno 14034, but no such call exists (and I
cannot remove lagid, either).
-- Channel 0/1, span 1 got hangup request, cause 16
-- Hungup 'IAX2/brandx-14819'
==
On Tue, 24 Mar 2009, Zeeshan Zakaria wrote:
I am not really sure, but apparently they guessed a SIP username/password.
But what I don't understand is they even though I deleted that extension all
together, still 'sip show peers' showed that extension. Then I figured out
an easy to guess
Hello,
In my scenario, the asterisk machine is installed behind a CISCO
mediaGW in order to be able communicate with the PSTN. Asterisk is
configured to use T.38 to send and receive faxes.
I'm trying to receive a fax from a fax machine located in the PSTN.
Apparently everything goes well: the
On Tue, Mar 24, 2009 at 8:10 AM, Tilghman Lesher
tilgh...@mail.jeffandtilghman.com wrote:
There are 4 billion possible IP addresses. To successfully block all
possible
hackers, you must block 4 billion of them. Seriously. Even your own
computer
is a possible source of hacking to other
On Tue, 24 Mar 2009 01:51:36 + (UTC), Jeff LaCoursiere wrote:
On Mon, 23 Mar 2009, Michael Graves wrote:
On Mon, 23 Mar 2009 20:01:51 -0400, Dean Collins wrote:
Siemens make a range of DECT handsets under the Gigaset model range.
Yes they shit all over every wifi handset I have ever
On Tue, Mar 24, 2009 at 9:21 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
WARNING[12229]: app_fax.c:650 in transmit: Transmission error
and the ReceiveFax function ends abruptly.
That doesn't really help, other than that it seems your arrangement
defaulted to voice rather than using
Hi,
I need some help, getting to work asterisk MWI. I set up Asterisk as
voicemail server for Openser as this tutorial shows :
http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+1.4+With+OpenSER+1.3
. My voicemail system is working but, I can't get to work the message
waiting
Hi!
A customer of mine wants to connect an asterisk system with 240 to 480 lines
to a PSTN switch. To save the costs for E1 cards and the corresponding E1
mainlines he wants to connect the system to the switch by a SIP trunk.
Phones will be connected to the server through the same SIP trunk as
Sorry about that, I forgot to post them:
-extension.conf:
[fax-in]
exten = 9,1,Set(INCOMING_FAXFILE=/root/santi/fax/incoming.tif)
exten = 9,n,Answer()
exten = 9,n,Wait(3)
exten = 9,n,ReceiveFax(${INCOMING_FAXFILE})
exten = 9,n,NoOp(FAXSTATUS: ${FAXSTATUS}, FAXERROR:
First Issue to be addressed is how many simultaneous calls and bandwidth
availability.
Number of lines (numbers) is not a limitation in it self unless they are
all in use.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
Hi all
I have a T1 connected and working. Can call cell phones and numbers no
problem.
I am using call files to place these calls and play wave files.
When the user wants to place a call to the intercom system the call is made,
internally the PBX routes that to an analog trunk. they say the
Here are a few look outs; Using conference rooms will increase your
bandwidth requirements. On board Network controllers will affect
performance in this high-use scenario. 250 simultaneous calls will use
about 7.5Mb of bandwidth depending on the codec(s) you use.
_
From:
2009/3/24 Christian Victor christ...@victormedia.de
Hi!
A customer of mine wants to connect an asterisk system with 240 to 480
lines to a PSTN switch. To save the costs for E1 cards and the corresponding
E1 mainlines he wants to connect the system to the switch by a SIP trunk.
Phones will
maybe it is not GS,we have the same problem with snoms 360
2009/3/24 Cary Fitch ca...@usawide.net
-Original Message-
boun...@lists.digium.com] On Behalf Of Rob Hillis
Yes. Grandstreams suck.
[Cary Fitch] We are not entitled to your opinion.
[Cary
We also use SNOM 360s on the same system. No issue with BLF being flakey.
Cary Fitch
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Leonja Cerebro
Sent: Tuesday, March 24, 2009 11:13 AM
To: Asterisk Users Mailing List
Hello,
I configured both asterisk and grandstream 2000 accourding to howtos on
the web..
And everything seems working fin.
But if i reload asterisk grandstream stops working with BLF.
I need to restart the phone to enable BLF again.
Any clues??
This is a publish/registration problem, not a
2009/3/24 Cary Fitch ca...@usawide.net
First Issue to be addressed is how many simultaneous calls and bandwidth
availability.
Number of “lines” (numbers) is not a limitation in it self unless they are
all in use.
Sorry for being a bit unclear in this point. What I meant was 240 to 480
2009/3/24 Danny Nicholas da...@debsinc.com
Here are a few “look outs”; Using conference rooms will increase your
bandwidth requirements. On board Network controllers will affect
performance in this “high-use” scenario. 250 simultaneous calls will use
about 7.5Mb of bandwidth depending on
2009/3/24 Grygoriy Dobrovolskyy megaho...@gmail.com
If the switch is fine why not ? But i wander why kind if switch is that
240-480 fxo ? ;)
Sounds like a big overkill.
And i dont see a problem with asterisk, if not too much transcoding
involved and with the right hardware.
It's an ISDN
I use a Dell with the 1Gb Ethernet on board, but had to clock it down to 100
Mhz because * has an issue with Dell on board Ethernet.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christian
Victor
Sent: Tuesday, March 24,
Our work around is to lower the registration expiration on the phones.
Under account settings in the web interface on the phones, we reduced
the Register Expiration from 60 minutes to 15.
This means the phones re-register every 15 minutes...and when they
register the BLF updates. Now when
On Tue, 24 Mar 2009, Danny Nicholas wrote:
Using conference rooms will increase your bandwidth requirements.
How does conferencing consume bandwidth differently than bridging?
On board Network controllers will affect performance in this high-use
scenario.
I know you've recently had
On Tue, Mar 24, 2009 at 11:33 AM, Santiago Gimeno
santiago.gim...@gmail.com wrote:
Sorry about that, I forgot to post them:
That all looks pretty good.
So in your original post, you clipped it off before you got all the
useful no-op output at the end.
I'm also assuming your file was empty?
2009/3/24 Danny Nicholas da...@debsinc.com
I use a Dell with the 1Gb Ethernet on board, but had to clock it down to
100 Mhz because * has an issue with Dell on board Ethernet.
Ah - good to know. I think we will use SUN machines. But I'll keep that in
mind.
Chris
If life were only that simple. A lot of hacking passes through unsuspecting
intermediary computers, precisely to hide their tracks, not to mention IP
spoofing. People have offered for sale access to 10,000 computers to use for
propagating mischief. That's a lot of IPs to block!
I got hacked
Asterisk 1.6.0.6 with dahdi 2.1.0.4 is showing a strange Unrecognized
prilocaldialplan error with the SIP username when a SIP call is dialed to a
PRI trunk. The error shows up like this:
Unrecognized prilocaldialplan TON modifier: a
Unrecognized prilocaldialplan TON modifier: b
Hi,
I'm trying to enable sip.conf outboundproxy support in version 1.4.20.1 of
Asterisk, but for the tests I made, every calls, even internal SIP calls
between extensions are sent over the proxy that I have specified with the
outboundproxy=xxx.xxx.xxx.xxx in sip.conf.
I think this isn't the
Hello,
I want originate a call to some destination, and when B side answes to
play a prompt. Asterisk version is 1.6.0.5. But also I need to insert a SIP
header to Invite, that's why I'm using Local Channel. This is my
extension.ael:
context autodialer-local {
_X. = {
Hello,
The NoOp output was not displayed at all. I'm assuming because of the
failure in the ReceiveFax application. In fact, the verbose output
was:
== Using SIP RTP CoS mark 5
== Using UDPTL CoS mark 5
-- Executing [99...@demo:1] Set(SIP/192.168.0.253-b7a96b70,
Wilton Helm wrote:
If life were only that simple. A lot of hacking passes through
unsuspecting intermediary computers, precisely to hide their tracks, not
to mention IP spoofing. People have offered for sale access to 10,000
computers to use for propagating mischief. That's a lot of
Not at all, just Dell :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, March 24, 2009 12:20 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
--- On Tue, 3/24/09, Ken Williams k...@intermountainelectronics.com wrote:
Our work around is to lower the
registration expiration on the phones.
Well, something's not working as I expect it to. My GXP2000 phones have
re-registration timeout of 2 minutes.
In my example below, extension
Vieri wrote:
I see much the same except I think if you investigate further, the light
will be green whether the phone ever registered or not.
--- On Tue, 3/24/09, Ken Williams k...@intermountainelectronics.com wrote:
Our work around is to lower the
registration expiration on the phones.
Hello,
is anyone on the list using a normal cell/mobile phone which is able to
act as a SIP client over WLAN?
Or has anyone heard of a SIP client for cell/mobile phones running
windows mobile 6.x?
The phone should use SIP, when the asterisk server is reachable and
should automatically switch
I REALLY like the Snom M3 DECT SIP base.
You can have up to 3 simultaneous calls through the base
and you can have up to 8 phones registered with it.
It's all web managed as well as http/s provisionable and has
this nice phone to line matrix where you can set which phones
ring on inbound calls and
On 24 Mar 2009, at 17:51, Ricardo Carvalho wrote:
Hi,
I'm trying to enable sip.conf outboundproxy support in version
1.4.20.1 of Asterisk, but for the tests I made, every calls, even
internal SIP calls between extensions are sent over the proxy that I
have specified with the
I'm afraid this isn't a GXP2000 bug but a broken feature in Asterisk 1.4,
unless I'm overlooking something (it should affect any phone with BLF; I only
have GXP2000 sets).
I don't have 1.6 yet so I can't see if BLF behaves the same way.
--- On Tue, 3/24/09, Jon Pounder j...@inline.net wrote:
Stefan Guenther schrieb:
is anyone on the list using a normal cell/mobile phone which is able to
act as a SIP client over WLAN?
Nokia N95 and some Nokia Exx (E90, E71, E66, E65 ?) I think.
The phone should use SIP, when the asterisk server is reachable and
should automatically switch to a
Hi,
We have a customer who used a strong quad-core Xeon box to convert up
to 800 simultneous calls from SIP to IAX and trunk them to another
box.
So your requirement doesn't look like a big problem.
Steve
On 3/24/09, Christian Victor christ...@victormedia.de wrote:
Hi!
A customer of mine
Just a guess, but your outboundproxy statement is in the global section of
sip.conf, which is making it apply to all sip traffic. If you move that
line to the applicable sip extension (ie. prox...@sipprov.com), this will
probably fix the behavior, even if it doesn't resolve the problem.
Anyone connected up to it yet?
http://www.skypeforsip.com/
This service is vaporware. It's just surveyware at this point with no actual
service. An alternative is OpenSky which is a launched service which does
SIP to Skype and Skype to SIP so you can answer and make all your Skype
calls from
I have several Dell boxes running onboard Broadcom and Intel NICs any haven't
had any issues. It's preposterous to make a blanket statement like that about
all Dell hardware.
Maybe you should re-compile your drivers. Or have prosupport come put a new
mobo in for you :).
-Dave
snip
Not at
2009/3/24 Steve Gladden aster...@michiganbroadband.com
I REALLY like the Snom M3 DECT SIP base.
Yeah - it's such a pitty that you always have to buy it bundled with one of
these crappy handsets. Or is there a way to get only the base that I don't
know?
Chris
On Tue, 24 Mar 2009, Philipp Kempgen wrote:
Stefan Guenther schrieb:
is anyone on the list using a normal cell/mobile phone which is able to
act as a SIP client over WLAN?
Nokia N95 and some Nokia Exx (E90, E71, E66, E65 ?) I think.
Yup - Nokia E90 here.
The phone should use SIP, when the
Okay - I'm not shooting from the hip here. The driver in question is a
Intel E1000 on a Poweredge 1650. If you visit the Digium site and do other
googling, you will see that there is a specific issue with asterisk and this
hardware/driver combination. I'm not really a fan of Dell, but I'm not
Then say as long as you don't use an Intel E1000 on a Poweredge 1650, as
I and others have had issues. I also have many Dell Poweredge series
with onboard NICs and no issues.
j
On Tue, 24 Mar 2009, Danny Nicholas wrote:
Okay - I'm not shooting from the hip here. The driver in question is
On Tue, 24 Mar 2009, Danny Nicholas wrote:
Using conference rooms will increase your bandwidth requirements.
On Tue, 24 Mar 2009, Steve Edwards wrote:
How does conferencing consume bandwidth differently than bridging?
On Tue, 24 Mar 2009, Danny Nicholas wrote:
Not at all, just Dell :)
It's actually a E1000 on Any POWEREDGE. If yall want a rukus, I can trash
Dell all day. That's not really what I had in mind though.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jeff
LaCoursiere
Sent:
Guess I made that one up. Conference causes other concerns, but bandwidth
isn't one of them. That's why they pay you the big bucks, Steve. :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Hello,
I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo
cancellation. Every 30-60 minutes I experience PRI dropping.
@@@ /etc/zaptel.conf:
loadzone=dk
defaultzone=dk
span=2,1,0,ccs,hdb3,crc4
bchan=32-46
dchan=47
bchan=48-62
@@@
@@@ /etc/asterisk/zapata.conf
[channels]
And nevermind. I just noticed that I didn't have warnings this time,
and it's perfectly normal.
2009/3/24 Harry Vangberg ha...@vangberg.name:
Hello,
I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo
cancellation. Every 30-60 minutes I experience PRI dropping.
@@@
Is your PRI dropping calls, or are the unused B channels resetting? What is
your resetinterval in the /etc/asterisk/zapata.conf?
On Tue, Mar 24, 2009 at 3:21 PM, Harry Vangberg ha...@vangberg.name wrote:
Hello,
I have an ISDN-30 connection and a Digium TE-121 with VPMADT032 echo
On 25/03/2009 10:05 a.m., Danny Nicholas wrote:
It's actually a E1000 on Any POWEREDGE. If yall want a rukus, I can trash
Dell all day. That's not really what I had in mind though.
Hmmm, I've also had problems with the e1000 driver in the past but not
on Dell - I seem to remember reading
I downloaded the newest E1000 driver from the Intel site and tried it on a
1550 and 1650 with no joy. So this isn't an attack on Dell, just a
verification of information I found and was trying to pass on to the
questioner. It could just as easily be some function of SUSE 11.0 (a bone
for you
It was just B channels resetting. Yesterday I had them dropping, and
thus I just hurried sending a message today, without noticing that
they were just resetting.
2009/3/24 Brandon B. bran...@brellsystems.com:
Is your PRI dropping calls, or are the unused B channels resetting? What is
your
Test
--Mensaje original--
De: tracinet
Remitente:asterisk-users-boun...@lists.digium.com
Para:Asterisk Users Mailing List - Non-Commercial Discussion
Responder a:Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [asterisk-users] [Asterisk-Users] Inter-Asterisk Using SIP
On 25/03/2009 11:08 a.m., darwin.sol...@gmail.com wrote:
Test
failed :)
--
Kind Regards,
Matt Riddell
Director
___
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
On 25/03/2009 10:54 a.m., Danny Nicholas wrote:
I downloaded the newest E1000 driver from the Intel site and tried it on a
1550 and 1650 with no joy. So this isn't an attack on Dell, just a
verification of information I found and was trying to pass on to the
questioner. It could just as
I recall having similar issues early in Asterisk 1.4...but currently running
1.4.17 and BLF works great with a phone expiration of 15 minutes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vieri
Sent:
Hi all,
Is it possible to have the Cisco 7960 dialing a SIP address to a service that
you are not registered with, for example:
sip:xxx...@x.org and asign that to some spee dial button?
I have heared that it should be possible to define in the dialplan.xml file,
but not sure. Any info is
hi
wich predicitive dialer are you using and wich one do you recomend?
a link to the project/product and a link to a how to will be VERY
apreciated.
Thanks
David
--
(\__/)
(='.'=)This is Bunny. Copy and paste bunny into your
()_()signature to help him gain world domination.
After passing authentication,
Then with this line,
extent = 361673,5,DISA(no-password calls-outbound)
As soon as the first digit of the intended number to be called is entered,
the system does a
Hungup 'DAHDI/1-1'
It has done that no matter what I have tried.
I am missing the boat
Danny Nicholas wrote:
Okay - I'm not shooting from the hip here. The driver in question is a
Intel E1000 on a Poweredge 1650. If you visit the Digium site and do other
googling, you will see that there is a specific issue with asterisk and this
hardware/driver combination. I'm not really a
Hello list,
I don't know if anybody faced this issue, but I finally found a
workaround. I am using an external program with an AMI connection to
originate outbound calls to Local/ channels, and on the dialplan context
I dial outside with the corresponding trunk according to the prefix of
the
Cary Fitch escribió:
After passing authentication,
Then with this line,
extent = 361673,5,DISA(no-password calls-outbound)
Please show the calls-outbound context to help you better.
As soon as the first digit of the intended number to be called is entered,
the system does a
Hungup
On Tue, 2009-03-24 at 23:05 +0100, Harry Vangberg wrote:
It was just B channels resetting. Yesterday I had them dropping, and
thus I just hurried sending a message today, without noticing that
they were just resetting.
If having them reset is causing problems, you can always set
Is there a way to make a call on a digital line T1/PRI
and dont WAIT for the signalling back that the call was answered?
I have a case where the T1 is working fine. I can dial work, numbers
cell numbers etc...
everything is fine. EXCEPT when I dial this internal trunk to access a
PA system.
The
In a SOHO environment I would agree with you, but not if your coverage area
needs to be tens of thousands of square feet. Deploying a complete overlay
wireless infrastructure doesn't make sense and is another infrastructure to
manage and maintain.
Frank
-Original Message-
From:
Frank Bulk wrote:
In a SOHO environment I would agree with you, but not if your coverage area
needs to be tens of thousands of square feet. Deploying a complete overlay
wireless infrastructure doesn't make sense and is another infrastructure to
manage and maintain.
did you think about
http://astguiclient.sourceforge.net/vicidial.html
On Wed, Mar 25, 2009 at 8:10 AM, David fire ddf...@gmail.com wrote:
hi
wich predicitive dialer are you using and wich one do you recomend?
a link to the project/product and a link to a how to will be VERY
apreciated.
Thanks
David
--
ok
-Original Message-
From: Frank Bulk frnk...@iname.com
Reply-To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] conference
Hi,
Can someone please answer this query.
We are planning to use Open IMS Core + Asterisk to make
Mobile to Land calls. Can you please let me know if the following setup is
possible.
Voip Client --- OpenIMSCore --- Asterisk
- PSTN
1) Can Asterisk
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