On 5/22/2014 12:41 PM, Steve Murphy wrote:
So, these defenses can be employed to stop/ameliorate such
hacking efforts:
1. Keep your phones behind a firewall. Travellers, beware!
Never leave the default login info of the phone at default!
2. Never use the default provisioning URL for the pho
On 5/1/2014 10:38 AM, Richard Kenner wrote:
Please go ahead and open an issue and attach the refs log and the full DEBUG
log. That will allow us to understand what's occurring here.
I need to wait until I'm sure this isn't something I caused somehow,
so I need to first understand why I'm seeing
On 4/23/2014 12:20 AM, Nick Cameo wrote:
That's about as simple as it gets.
A call file that goes to the dialplan.
A dialplan that consists of Read (which would play the message)
followed a GotoIf into a mailbox (either voicemail or Dial() to an
external number).
One h
On 4/22/2014 5:54 PM, Nick Cameo wrote:
Hello Everyone,
Thank you all for your response. The people I am doing it for run a
non-profit charity,
and are legally able to reach out to their customers. I will wire it
up to the DNC
however, for starters, I would like to get asterisk to:
i) Iterate t
On 4/21/2014 3:58 PM, Nick Cameo wrote:
On Mon, Apr 21, 2014 at 2:01 PM, James Sharp mailto:ja...@fivecats.org>> wrote:
On 4/21/2014 1:47 PM, Mitul Limbani wrote:
Use vicidial for achieving the same.
Or call files (or AMI originate), a short bit of dialplan logi
On 4/21/2014 1:47 PM, Mitul Limbani wrote:
Use vicidial for achieving the same.
Or call files (or AMI originate), a short bit of dialplan logic, and
maybe a call to Queue().
--
_
-- Bandwidth and Colocation Provided by ht
On 3/26/2014 12:20 PM, Michelle Dupuis wrote:
If this is to 972 area code then the next digits should be 0X or 0XX but
they are not. This differs from what I found documented for that area
code - I thought someone from the region might add to the discussion.
Not sure if this reflected a premiu
On 3/18/2014 6:58 PM, Paul Belanger wrote:
On Tue, Mar 18, 2014 at 1:02 PM, James Sharp wrote:
Putting a whole bunch of people into a listen-only/muted Confbridge
conference or getting the broadcaster audio into a MOH class and then just
having callers attach to that MOH class?
Does the the
Putting a whole bunch of people into a listen-only/muted Confbridge
conference or getting the broadcaster audio into a MOH class and then
just having callers attach to that MOH class?
Does the the muted side of a Confbridge Room still try to mix in audio
from the muted channels or does it just
On 2/18/2014 2:09 PM, Eric Wieling wrote:
No. Asterisk will accept calls from unregistered devices, but you have to
enable guests I sip.conf and hope your dialplan is secure. No sane person does
this.
Asterisk cannot send calls to a device unless it knows the address from a
register or
On 2/5/2014 12:09 PM, G. Paul Ziemba wrote:
I'm using asterisk 1.8 as an answering machine. I'd like to
hear the calls it answers aloud in case I want to pick up and
interrupt the call.
There are a few articles describing, for example, three-way
calling a monitor phone set to auto-answer, but I
On 1/15/2014 5:50 AM, Gareth Blades wrote:
On 15/01/14 09:39, Francesco Namuri wrote:
Hello James,
thanks for your answer, I supposed this too, but my provider answered me
that as
m=audio 43718 RTP/AVP 8 18 3 101
^ ^ ^ GSM proposal
On 1/15/2014 3:59 AM, Francesco Namuri wrote:
Hello,
I'm having this issue on my pbx, it appears that asterisk is refusing
the codecs that my providers is proposing.
My trunk configuration is:
Pretty simple -
---
username=5x5x7x9x0x3
type=friend
secret=CRcxn7sqwm
qualify=yes
port=5060
insec
On 12/28/2013 7:04 AM, Shahid H wrote:
Thanks Daniel, that was useful, I will check those links :)
I am pretty good with PHP and jQuery. So I guess learning Node.js
shouldn't be too difficult.
If I decided to use Node.js - what is the best way to communicate with a
browser to AMI process? Send
On 12/3/2013 10:11 AM, Don Kelly wrote:
In the php routines, I would like to use the persistent connection
that is established in the dialplan, rather than creating a new
connection each time they run. How can I do this?
You can't, they are completely separate processes and code.
Joshua Colp
On 11/24/2013 2:47 AM, Todd R. wrote:
Did you have the externalip setting in sip.conf set to the Elastic IP?
I believe I did. But I didn't really get a chance to plow into it too
much, I had a client holding me at gunpoint.
--
__
On 11/22/2013 12:52 PM, Todd R. wrote:
Just checking one more time to see if anyone has an opinion on this. I
am primarily interested in using a cloud type setup such as Amazon AWS
for the redundancy, easy backup and recovery options. It's not about
price but the idea that it will be very hard fo
On Aug 26, 2013, at 8:11 PM, Manolo Quijano wrote:
> Hi all,
>
> This my first mail in the community. My name is Manolo. I'm new in
> Asterisk. My objective is to control some radio using Asterisk via web.
>
>In google I could see that the application app_rpt had this goal, but
> curre
On 07/10/2013 01:04 PM, bilal ghayyad wrote:
Hello;
To let the Phone answer automatically, this can be configured from
asterisk (at the sip.conf for the phone)? Or it has to be from the IP
Phone? Because, some phones does not support auto answer, also we do not
need to do it for each Phone.
De
Yes, you will be able to transfer calls between the E1 ports without a problem.
As far as timing goes, the PBX will not be a clock source. You need to
configuring your timing setup so that Asterisk takes timing from the Telecom
PRI and then sources clock to the PBX. The PBX will take timing f
On Jun 16, 2013, at 4:27 PM, Nick Khamis wrote:
> Anyone try this? I saw a post here:
>
> http://www.elastix.org/index.php/en/component/kunena/1-installation-issues/94041-setup-of-sangoma-a101-in-my-elastix.html
>
> But not sure if it's possible. What I am asking is if there are any T1
> cards
On 11/7/2012 2:01 PM, martin f krafft wrote:
Dear list,
we would really like to be able to "invite a third and fourth party"
to our current one-on-one call. At the moment, we have to agree to
dial into MeetMe 10 minutes later, then make calls to the third
parties, and hope it all works out.
I h
On 10/9/2012 3:52 PM, Niccolò Belli wrote:
http://www.traverse.com.au/geos21-dual-adsl2-x86-router-appliance
I achieved fallback in less than 10 seconds flushing routing cache and
nat tables with nearly zero false positives (I can do even better but I
prefer having less false disconnections).
I
On 9/28/2012 12:42 PM, Mitch Claborn wrote:
I want to put a "call me now" button on the web site that will place the
request into an asterisk call queue and then when an agent picks up the
call in the queue, place the outbound call to the customer.
The following AMI command works, but it calls t
On 9/18/2012 3:41 PM, Ahmed Munir wrote:
Hi all,
I would like to know, is there a way to trigger Asterisk after data
inserted into mysql DB? Like here what I'm trying to do, when the new
data inserted into MySQL DB, it sends the request to Asterisk along with
the new data (that is inserted in D
On 9/13/2012 2:31 PM, equis software wrote:
After exchanging the cable with other equipment was running smoothly in
my computer the problem persisted while the other team the cable that
could be bad worked.
With this test done, now suspect the problem I have it on the Digium card.
I perform the t
On 8/12/2012 3:57 AM, Steve Underwood wrote:
On 08/12/2012 10:32 AM, James Sharp wrote:
On 8/11/2012 8:05 AM, virendra bhati wrote:
Hi team,
I want to configure fax with asterisk. there a lot of fax link i found
by google but not working perfectly. my setup as follow
asterisk 10.x
centos 5.8
On 8/11/2012 8:05 AM, virendra bhati wrote:
Hi team,
I want to configure fax with asterisk. there a lot of fax link i found
by google but not working perfectly. my setup as follow
asterisk 10.x
centos 5.8
Want to used T.38 with SpanDSP...
Please suggest me the best way. and how to test FoIP .
From my experience with xlite, the soft phone itself must be configured for
auto answer. There is no way for the dialplan to control this.
On Jul 13, 2012, at 2:35, upendra wrote:
> i am using a x lite phone.
>
>
> regards
> upendra
>
>
> On Fri, Jul 13, 2012
Different phones use different methods. What kind of sip phones do you have?
On Jul 13, 2012, at 12:17 AM, upendra wrote:
> Hi,
>
>
> i wanted to make dial plan in such a way that the any incoming call to the
> sip phone should auto answer.(auto pickup) .
> Help.
>
>
>
>
> regards
> Upe
On 7/2/2012 5:15 PM, Carlos Alvarez wrote:
We are a hosted PBX service provider using Asterisk (primarily 1.6,
moving to 1.8 soon). In the past, when we've been asked to provide call
recording, we deploy a custom server just for that customer. I'd like
to bring call recording to our standard ho
On 6/28/2012 3:53 PM, Ernie Dunbar wrote:
We have three Asterisk servers, Voip1, Voip2, and Voip3. Voip1 has a
Wildcard TE405P (3rd Gen), Voip2 has a Wildcard TE410P/TE405P (1st Gen),
and Voip3 has a Wildcard TE110P. Voip1 is the server that handles our
PRI to the PSTN and we hope will allow us t
Does your VoIP provider support t.38?
Sent from my iPad
On Jun 22, 2012, at 11:05 AM, Ahmed Munir wrote:
> Hi,
>
> I recently configured T.38 on Asterisk 10.4.2. When I send the fax to
> Asterisk, it gives the errors as listed below;
>
> WARNING[25986]: app_fax.c:442 transmit_audio: channel
On 6/18/2012 11:52 AM, Thorsten Göllner wrote:
Hi,
I am trying now for over 4 hours setting up cdr-logging via odbc into a
mysql database. But with no success. Do you have any hint for me?
*SNIP*
But after a call hangup I get the following error:
cdr_odbc.c: Unable to retrieve database hand
On 5/16/2012 12:07 PM, Tim Nelson wrote:
- Original Message -
Hi,
I'm facing a strange situation.
Though it's not directly related to Asterisk, I do think it is
interesting to this mailing list.
The setup is a single line which is split between an ADSL
modem/routeur and a fax machine
On 5/4/12 1:57 AM, Bruce B wrote:
James,
That is amazing details. I can use all of this. Thank you for sharing.
I am assuming you installed res_fax from repository?
*yum install asterisk18-res_fax_digium.i386*
No. I built Asterisk 10 from source. Once I installed spandsp, the
menuconfig ha
On 5/3/12 9:16 PM, Bruce B wrote:
Lee,
Much appreciated for the input.
I am running all VoIP. SIP and IAX2 to our ITSPs. Please elaborate on
HylaFax and IAXmodems. Is there a guide posted on to get it running, or
is it part of the repository? Once installed how would one send .pdf as fax?
You
On 3/13/12 5:53 PM, Danny Nicholas wrote:
Ping the phones, then run arp.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
resea...@businesstz.com
Sent: Tuesday, March 13, 2012 4:52 PM
To: asterisk-users@lists.d
On 02/15/2012 03:03 PM, Olivier wrote:
Hi,
When someone says "T.38 is not reliable on a (normally loaded and
managed) LAN", would you rather agree or disagree ?
In this case, fax calls are coming in through an analog gateway,
passing trough Asterisk and then going out to ISDN through a digital
g
I run my Asterisk system on a quad core Opteron system running VMWare ESXI 5.
On Feb 10, 2012, at 21:18, Carlos Rojas wrote:
> Hello everybody
>
> someone in this list, has installed asterisk, in a virtual server like
> proxmox? I'm thinking install some asterisk servers in a machine dell
On 02/01/2012 02:17 PM, bilal ghayyad wrote:
Hi All;
I heard from some friends that there are a dsl router that has Linux OS
and it has asterisk on it, so the ip phone can register on this router,
also if the router has FXS or FXO ports then it can be used to place
calls through them.
Is it rea
On 01/09/2012 02:44 AM, Eyal wrote:
Hi,
I try to create a new table using MYSQL command in asterisk.
This is what i write:
*Query resultid ${connid} CREATE TABLE IF NOT EXISTS "conference_600"
("id" int(11) NOT NULL auto_increment, "channel_id" varchar(40),
"number_in_line" int(2), PRIMARY KEY("i
On 01/03/2012 09:42 PM, Raj Mathur (राज माथुर) wrote:
Hi,
I have a queue with a number of (static) agents. Is there an easy way
for an agent to indicate that she is away from her seat, so that her
phone is not rung when a call comes in? And the converse, of course: be
able to notify Asterisk w
On 01/03/2012 01:22 PM, Todd Routhier wrote:
Sounds perfect, I will need to look into how to blend them together like
that.
Put them in extensions conf like so:
[agentblends]
exten => bob,1,Dial(SIP/300&SIP/12102263232@myprovider)
Then put Local/bob@agentblends into your queue.
I wonder th
On 01/03/2012 01:06 PM, Todd Routhier wrote:
Happy New Year to all!
Asterisk 1.8.x
I have a queue to which I add agent channels like SIP/300 dynamically
using the manager interface. Once logged in, there SIP/300 of course
rings when a call is distributed to them.
How can I also get the agents
On 12/26/2011 04:15 PM, sean darcy wrote:
Thanks for the response. Home asterisk : 10.0.0 - Office: 1.8.8.0
So I thought I'd leave all the sip providers on udp, and move the
home-office to tcp.
And registration just work Just Worked over the default tcp registry
port - which I was surprised to
On 12/15/2011 01:43 PM, James Sharp wrote:
On 12/15/2011 01:33 PM, Tarek Sawah wrote:
Hello List,
I have customer with a 40 Agents call center. and is looking to
install a PBX switch that can serve those agents.
As per my experience i suggested Asterisk as i have tested it with
Call Centers
On 12/15/2011 01:33 PM, Tarek Sawah wrote:
Hello List,
I have customer with a 40 Agents call center. and is looking to install a PBX
switch that can serve those agents.
As per my experience i suggested Asterisk as i have tested it with Call
Centers, however he has been advised not to use it al
Build Asterisk with ODBC support and then use the ODBC functions to do the
database dips.
On Dec 12, 2011, at 13:44, Douglas Mortensen wrote:
> Any suggestions from people who have done this before?
>
> Thanks,
> -
> Doug Mortensen
> Network Consultant
> Impala Networks Inc
> CCNA, MCSA, S
On 12/12/2011 12:35 AM, Mike Diehl wrote:
Actually, I've configured the phones to use DNS SRV records to find the Asterisk
server, and this works very well. The problem is that when the router fails
over, the phones IP address changes and this causes them to be unavailable
from Asterisk's point
On 12/11/2011 07:22 PM, Mike Diehl wrote:
Hi all,
I've got a customer who is bringing up a second Internet connection for fail-
over. I've configured a WRT54 with 2 LAN ports and arranged for it to fail
over when one of the routes is no longer available. That works just fine at
the IP level.
I check in CLI
[Dec 6 17:46:58] WARNING[16264]: pbx.c:4088 pbx_extension_helper: No
application 'MeetMe' for extension (employees, 777, 1)
== Spawn extension (employees, 777, 1) exited non-zero on
'SIP/phone1-'
Plz tell me , where i am wrong in configuration.
Chances are you didn
On 11/16/2011 10:30 AM, eherr wrote:
But what is the correct physical setup of a CLEC.
Do you get rack space at a carrier hotel and equipment in there?
Do you get rack space at the local ILEC CO?; which is Verizon here.
What are the types of voice platforms used by CLECs?
Just as a point of
On 11/14/2011 09:57 PM, sean darcy wrote:
Unthinkable!! Used wireshark: I can see the REGISTER packets going out
from the home router, but nothing from home:5060 shows up at the office.
Bummer. Now I get to think about how to set up special ports between
home and office. A great evening activit
You're not going to get a telnet connection on port 5060, since that's tcp and
sip uses UDP.
Use tcpdump/wireshark on your office pbx to see if the packets are getting to
you. If not, then there's something wrong inbetween.
A firewall misconfig, perhaps. Or the unthinkable: your home ISP has
On 11/03/2011 09:16 PM, Nick Khamis wrote:
Hello James,
Thank you so much for your response. We just purchased an AudioCodes
MP124 for this job. And setting
up OpenSIPS as the proxy. As I mentioned earlier, Bell Canada is the
Telco here in Toronto. As for other
Telcos around the world, for examp
On 11/03/2011 07:20 PM, Nick Khamis wrote:
Hello Everyone,
Unlike going through DIDx, DIDLogic etc.., we have an option of
getting DIDs directly
from local telco Bell Canada. Currently our SIP Trunk provider
assigned a DID to us,
and as you know, they just redirect requests it to our PBX.
Howeve
On 10/15/2011 05:31 AM, Michael C. Robinson wrote:
[Oct 15 01:48:02] NOTICE[3747] channel.c: Dropping incompatible voice
frame on SIP/2006- of format ulaw since our native format has
changed to 0x8 (alaw)
[Oct 15 01:48:49] WARNING[3750] pbx.c: Channel 'DAHDI/1-1' sent into
invalid extensi
Then there's also the point where it makes more sense to drop a GSM card into
your Asterisk box and get a cheap unlimited mobile to mobile plan for a SIM and
use that to transit your calls to VoIP.
Although that won't help the original asker, though, since he mentioned
Verizon.
On Oct 11,
On 10/10/2011 10:31 PM, linux guy wrote:
On Mon, Oct 10, 2011 at 8:08 PM, Andres wrote:
I would recommend Acrobits. Not free but only a few bucks. It works fine
with ATT 3G.
This begs the question... which is more expensive (and where)...
making a regular cell call or making a SIP call ove
On 10/10/2011 05:35 PM, john Millican wrote:
Hello all,
Does anyone know of a good free/inexpensive 3G SIP client for the
iPhone? If anyone is using one that works good for them could you please
let me know.
I use VaxVoip on my 3GS and iPad. It works great over both 802.11 and 3G.
--
On 10/08/2011 02:38 PM, Ryan Wagoner wrote:
I signed up with Gafachi a few weeks ago to use them for T38 as well.
I haven't had any luck getting it to work. I have been mainly trying
to use Asterisk in T38 pass through mode and have tested with a
Linksys SPA2102 and Zoiper. Gafachi basically tol
On 10/07/2011 04:42 PM, Kevin P. Fleming wrote:
You shouldn't be *receiving* CNG, as you are the calling endpoint.
You're right. Hadn't even thought about that.
If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS,
etc. then something is badly wrong.
... and, that thing
On 10/07/2011 04:04 PM, Kevin P. Fleming wrote:
First, we can see that Gafachi's T.38 implementation still has some
breakage in it (although the problems are ones that Asterisk has been
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has
"a=T38FaxRateManagement:transferredTCFlocalT
On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use "sip set
debug on", "rtp set debug on" and "udptl set debug on"
No NAT involved and I shut off IPTables. Still no luck. Debug shows
the SIP invite, RTP
7 failed
faxes? I know 1 transmit didn't go through because I tried to place one
call while another was in progess and I only have one licensed channel.
Thanks,
James Sharp
ja...@fivecats.org
--
_
-- Bandwidth and Colocat
> Quickie: Does anyone out there have experience with PRI delivery of ANI II
> information?
>
> Specifically, I want to know if it's possible from within Asterisk to know
> if the inbound call (which may or may not be to an 800 number) came from a
> payphone or not. I know with some 800 providers i
> We have one other error (twice today) we get "Out of trunk data space on
> call number , dropping"
>
> How do I determine what is causing this error? we have a point-to-point
> T1
> between 2 * boxes, with 3 phone in the remote office. I have no idea how
> the trunk could be out of space.
> I am looking for a good case to house my Digium PCI cards, I was hoping to
> mount them in the front for cleaner access then in the back. Unfortunately
> I
> haven't found much, does anyone have a good recommendation for chassis to
> use up to six digium cards?
Probably not cost effective, but i
> Hello-
>
> I asked this question a LONG time ago (when I
> first got started with *), but seem to have lost
> the answer in between my multiple Windows XP
> "repairs".
>
> Has anyone experimented with or achieved PLAR
> (private line auto ringdown) capability with
> asterisk?
Its fairly easy if
> Steven,
>
> Perhaps I should have posted my question differently to the list:
>
> After installing the CVS version of Asterisk, I type, "modprobe xct1xxp."
> The machine accepts the command but the LED on the T100P does not flash.
> How do I know that the T100P module has loaded correctly?
Do y
> Is it possible to have the system outdial and take surveys. either by
> receiving DTMF or voice?
Yup. Just have the system use the outgoing queue (see sample.call) and
have it call an AGI script upon answering.
___
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[EMAIL
> Hi,
>
> Please excuse me if my question seems too simplistic. I have been reading
> the mailing list for some time and I am still a bit confused. Here is the
> scenario that I would need to achieve and am wondering if asterisk is the
> correct software to use.
>
> (h323) (h323/SIP
> On Tue, 3 Feb 2004, Chris Albertson wrote:
>
>> "Smallest" Asterisk server? No. That old Gateway box must
>> be about 2 cubic feet. 1.5 ft^3 at a minimum. I've got one
>> that is about 0.2 ft^3 a factor of maybe 10 smaller.
>
> Hehehe.. As far as "Form Factor" goes, I'm sure there are smaller
> Now, here's the real question: can you install it on a toaster?
It builds and runs on NetBSD, minus the hardware part (for the
moment)...so yeah.
Asterisk on NetBSD/Vax. Hrm.
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http://lists.digium.com/m
>> > exten => _.,1,Dial(Zap/1/$EXTEN)
exten => _.,1,Dial(Zap/1/${EXTEN})
Gotta put the name of the variable in brackets for it to work.
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http://lists.digium.com/mailman/listinfo/asterisk-users
To UNS
> Thanks John,
>
>
> I think it is not that simple. I am not using a phone but a Cisco ATA.
>
> The scenario: -
>
> User--(Multitech VOIP MVP200 (FXS))--Internet--(Multitect VOIP MVP100
> (FXO))--Cisco ATA--Asterisk--Any extension
Any reason you can't use the H.323 load for the MVP200? I've not t
> Hello All,
>
> I've mostly solved my DID problem from a few days ago. Apparenly the
> lines weren't configured properly. Now heres the next question. 12 E&M
> wink lines from telco. I have them all plugging into an Adtran 750 with
> FXS cards. The Adtran ports are configured DPO. How do I
> I'm evaluating * to replace the crap set of peered "smart" phones we
> have now in our small office, but I haven't been able to find out about
> this anywhere yet: I need to know if * can discriminate _incoming_ FAX
> calls on a voice line and route them to a specific extension?
Yes, it can.
_
> ; FXS Port 1
> context=local
> signalling=fxs_ls
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=yes
> ;
> ;FXS Port 2
> context=local
> signalling=fxs_ls
> usecallerid=yes
> echocancel=yes
> echocancelwhenbridged=yes
Change the signalling here to fxo_ls. Its gotta match what's in z
>
> The solution to the problems with the Grandstream 1.0.4.39 firmware is
> to use inband (in-audio) DTMF. Neither the RFC2833 nor INFO seem to
> work.
Don't the Grandstreams send a DTMF 'F' INFO message on a hookflash?
Shouldn't be that hard to change chan_sip to register an 'F' as an
AST_FLAS
> # ifconfig xl0
>
> xl0: flags=8843 mtu 1500
>
> address: 00:01:02:78:11:e8
> media: Ethernet autoselect (10baseT)
> status: active
> inet 203.219.167.126 netmask 0xfffc broadcast 203.219.167.127
> inet
> I have a little more info on this. Following the suggestion of another
> post on
> this topic I tracked down an analog phone with lighted buttons powered by
> the
> phone connection. I directly connected the phone to one of my inbound
> lines and
> called it with my cell phone. Picked up the anal
> Can anyone help me with the term that SBC uses to refer to disconnect
> supervision? I have an Adit 600 channel bank which has helped improve the
> disconnect detection time down to about 8 seconds. This is still causing
> some
> issues in particular with call progress enabled in * we are having
> Does anyone else have 2 t1's plugged into their T400 ? If
> so, how are they synced ? This was just happening at night,
> but I lost the second span a dozen times already today, all
> within less than an hour earlier this afternoon.
>
If you've got 2 spans from the same provider, you should ju
> If some channel banks don't support this, how on earth do they know when
> the telco side of the call has hung up ?
They don't. They rely on either a timeout or the called party hanging up
to disconnect the call.
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> FYI there is a way to do 911 its called E-911 enhanced 911
> the user has to set it up with the local emergency services
> to it and you setup your pbx to xmit the data.
There's PS/ALI (Private Switch Automatic Location Information) that's
quickly becoming state mandated for all PBX systems. Th
> I had documented the Makefile modification in an email to the list. If you
> search for Sparc in the mailing list, you should be able to find it. If
> not, drop me a line and I'll see if I still have it.
>
I've got an Ultra 30 sitting here doing nothing. I'll see what I can come
up with for Li
> 1. Moving a physical interface (whether a T1, ethernet or 2-wire pstn) is
> mostly trevial, however what "signal" is needed to detect a system failure
> and move the physical connection to a second machine/interface? (If there
> are three systems in a cluster, what signal is needed? If a three-wa
>> Andrew Kohlsmith wrote:
I would set the "Enterprise Class" bar at five 9's reliability
(about 5.25 minutes per year of down time) the same
as a Class 4/5 phone switch. This would require redundant
design considerations in both hardware and software.
>>>
>>
>> To turn around, let'
> I am putting together a solution that will employ the Digium TE410P with
> one T1 going out the PSTN and the other front-ending a PBX. The idea is
> that based on a URL, Asterisk will dial an employee behind the PBX. When
> the employee picks up, Asterisk will dial the customer (detailed in the
>
> Occasionally I do NPA-NXX lookups for my local exchanges to see what other
> carriers have prefixes in my area. I used to use telcodata.us, but they
> seem
> to have gone offline. Usually, after you find the carrier's name, you can
> see info on the location and type of switch being used. I can't
>
> Hi all.
>
> Could it be possible that video frame buffering be causing problems
> even if the computer is not running X ?
Yes. There are known problems with systems running with either a frame
buffer console or a serial console. For best results, run a plain VGA
console.
What about having your VoIP gateway system placing a 911 call to the 911
answering center in the appropriate region and when the 911 operator
answers, have a message say "This is a 911 call from 123 Main Street,
Nowhere Nebraska" then connect the caller to the 911 operator. Legal?
Maybe. Dunno.
> Run using a serial console
> (http://www.tldp.org/HOWTO/Remote-Serial-Console-HOWTO/). No monitor,
> VGA adapter, keyboard etc needed. Use SSH to log into the asterisk box
> for any maintenance, etc. If the box gets hosed, connect the serial
> port to a working PC and fire up minicom and your
> It's just my lowly opinion but I too must agree when it comes to the
> consumer/soho (1 to 3 line) markets.
>
> CAUTION!!, DANGER!! Marketing Hat On!!
>
> Vonage, the most "visible" marketer of a voip consumer product must also
> agree. Vontage offers an ip "fax line". using cisco's ata. Vontage
It can be either.
> Does this card only work as PRI or can it be used like a standard T-1
> wired
> to a PSTN Switch?
>
> TIA
>
> -Seth
>
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
> Can someone give me an idea exactly what things are intended to be tested
> via RADIUS, or some other AAA system?
>
> Are we talking about building SIP/IAX/H323 entries from RADIUS?
>
This is where the PAM system I developed for * comes into play. I've got
most of it working at the moment, but
>> Case 1 and 2 are ties in my eyes, except the channel bank would
>> provably be cheaper to upgrade to 8 lines. I am just afraid of the
>> channel bank. I just don't know anything about them. If I buy the
>> wrong crap, it gets really expensive fast, plus adds another layer of
>> complexity.
It seems that there's a non-printable character at the beginning of the
DNIS stream I'm getting from the SUMA 4 switch. Once I chopped that off,
everything works right.
> Hi James,
>
> Try to do
> exten => _8005095639,1,Agi(ivr-main.pl)
>
>
> Quoting James
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