I think I'm looking for a carrier who's implementing Asterisk Matt. We're a
carrier, and that probably explains a lot of the frustration, and hair tearing
out we've had with Asterisks resundancy. Ie, it wasn't built with carriers in
mind. It was build with IT implementations in mind.
When you find out, let us know, because it isn't documented anywhere! We had to
drop ARA because it's behaviour was very unpredictable.
-Original Message-
From: unplug [mailto:[EMAIL PROTECTED]
Sent: Thu 7/20/2006 7:49 PM
To: Asterisk Users Mailing
Do you mean to drop AST DB? It is what I found in my previous mail.
NAT enabled
Register method:
1. log external IP and port in ARA DB
2. log external IP and port, internal IP and port in AST DB
invite (out bound) method:
1. construct a invite message using AST, (not sure it will involve ARA).
There is a patch available for the quiet voicemail volume issue (bug 6237)
but it isn't intended to work with 1.2.9. The patch below will give you this
functionality for 1.2.9. Add the volgain= parameter to voicemail.conf and
make sure sox is installed.
--- apps/app_voicemail.c.backup 2006-07-18
I think that context=incoming-[number] in firends definion is used
rather for determinig context for outgoing calls.
In sip.conf [general] section there is context= and
register=/[extension]
i think that extension in register line should be in context specified
in [general]
Delca wrote:
Fixed, i'm the kind of guy who ask and later find the solution :$ it
is a Linksys PAP-2 ATA setting in Regional - Control Timer Values -
Interdigit Long Timer (this is in advanced mode).
Sorry :)
Santiago
On 7/20/06, Delca [EMAIL PROTECTED] wrote:
Hi, I'm using a Linksys PAP-2 to
in [asterisk_sources_dir]/configs/extensions.conf.samples
(for asterisk-1.2.4 there was [mainmenu] and [submenu] contexts
describing very basic ivr - it was first to try for me)
I used to build interactive voice response systems for a university. I
have someone who would like a small system
Nobody is going to pay much attention to your help requests if you can't
even figure out how to set it up so the subject header reflects what the
problem is all about.
Why don't you try again with an appropriate subject?
B.
--
This message has been scanned for viruses and
dangerous content
Woodoo People .pGa! wrote:
I have a Voismart GSM card. I have calls through going fine. But in the
cdrs, all the calls have disposiotion of NO ANSWER and the billsecs are 0.
I am using Asterisk 1.2.7, visdn 0.16, kernel 2.6.11-12, on CentOs 4.2
that's call received via vgsm interface
*CLI extensions reload
*CLI) sip reload
will do what you want
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On Jul 21, 2006, at 12:43 AM, RR wrote:
*CLI extensions reload
*CLI) sip reload
will do what you want
If you reread his post it appears that he is saying that didn't work
for him.
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There might be better ways of doing this through the TUI but if you
have access to the CLI then you can do the following
*CLI meetme list 1000
this will show you that Channel: SIP/200 holds (say) userID = 03, then
you can do
*CLI meetme kick 1000 3
2006/7/20, voiplist [EMAIL PROTECTED]:
On 7/20/06, Mat Stace [EMAIL PROTECTED] wrote: I'm not exactly sure on the /how/ * mathes items from the sip.conf (I suspect it goes to the latter for whichever provider), but the way
configured my extenions.conf to handle multiple incoming accounts from
Haha sorry, well beats the hell out of me then.
The only guess I can make is he's doing the mods in some other file,
e.g. in trixbox using FreePBX, which makes all the mods in
extensions_additional.conf etc. and it's not being included by
extensions.conf. and since sip and extensions both are
I'm still in the testing stages of Asterisk, I haven't began with additional extension files. Or maybe I just discovered a bug, I'll try downloadiing a fresh copy and reinstalling Asterisk.
Regards
-Kevin
On 7/21/06, RR [EMAIL PROTECTED] wrote:
Haha sorry, well beats the hell out of me then.The
Dear
I want to make a billing recharge through receiving digits from IVR
through dtmf which will
be inserted at mysql a2billing database after validation the digits entered ,
How can todo
that ?
Regards
*
No employee or
CDR
Hi
Please how can I get the user ip address and put it at cdr
,its too important
Thanks
*
No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail
On Jul 21, 2006, at 1:08 AM, Kevin Class wrote:
I'm still in the testing stages of Asterisk, I haven't began with
additional extension files. Or maybe I just discovered a bug, I'll
try downloadiing a fresh copy and reinstalling Asterisk.
That seems silly if it's running ok... Make sure
That seems silly if it's running ok...Make sure the conf files youare editing are in the proper place.. That is an easy way to not get
any results from a reload ;~)Marty
All my users fall under the same context which is [default], in both my sip.conf and extensions.conf. And again, I found out
There are some others out there, we did something similar at
http://www.asteriskguru.com/archives/
i prefer to read it in an email client myself though, but my mailclient
archive is not as complete when i want to search for something. (Google
is not really good at indexing them so google
I have two polycom phones. One on a slow link, and one on a fast one.
I'm trying to set the phone on the slow link to use G729 as it's first
preference, and the phone on the fast link to use G711 as it's first
preference.
sip.conf has:
[general]
allow=ulaw
allow=g729
[slow-link] ;
Kevin,
like I mentioned before, FreePBX will do all of that for you. It works
if you have a one-box (*) solution. If it spans multiple boxes then
you got a bit of an issue. ARA does a fair bit of it, but I think it
has issues if you're using (*) to lookup sippeers etc i.e. using * to
be your
Hi,
yes there's was a small typo that prevented answer to be detected
into the channel driver.
Please check the wiki (http://open.voismart.it) and try the 0-16beta1
version.
matteo.
--
Matteo Brancaleoni
RD Director
Tel +39.02.70633354
Sip [EMAIL PROTECTED]
Iax2 [EMAIL PROTECTED]
On 20 Jul 2006, at 15:12, Moises Silva wrote:
Asterisk does not have softphone interfaces. You can write a softphone
to support some VoIP protocol supported by Asterisk, and voila, you
can connect to Asterisk. Supported and common protocols are IAX2, SIP
and H323. For IAX you have a library
I must say that for mailing lists Gmail seems to me just perfect! I
apreciate your integration into Forum. But Gmail seems to me even more
friendly!
Best regards,
Marco Mouta
On 7/21/06, zoa [EMAIL PROTECTED] wrote:
There are some others out there, we did something similar at
No, we aren't intending to check for available g729 codecs
that's why we wanted to have ulaw as a backup when no g729 codecs
where available.
That won't work. If it's trying to use G729, it will still try even
when the licenses are all in use. So you need to either force it g729
Just an idea:
Put this Slow-Phone sip account into sip realtime database, and
outside of asterisk manage to verify G729 licenses availability and
script it to your SIP-realtime.
This way every call to this SIP account will go to SIP realtime
database that is being changed by an external script
On 7/21/06, Tim Panton [EMAIL PROTECTED] wrote:
On 20 Jul 2006, at 15:12, Moises Silva wrote: Asterisk does not have softphone interfaces. You can write a softphone
to support some VoIP protocol supported by Asterisk, and voila, you can connect to Asterisk. Supported and common protocols are
I believe this bug is independent of handset and though I've only come
across it with SIP I think it affects all channels too. The transfer
method is that defined in features.conf.
Mike
Tong wrote:
Are you using a cisco 7960 with POS 8.2?
- Original Message - From: Mike Dawson
Hi,
I am experiencing a hard to solve problem with my VoIP provider. I can
make calls without any problem but I can not receive any. Actually,
calls arive to * but the phone just does not ring. I believe must
be a problem with NAT but I think I have a good config:
- Extensions have nat=always
Did you port forwar in your router RTP ports ? 1-2 to your *Box ?
On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote:
Hi,
I am experiencing a hard to solve problem with my VoIP provider. I can make
calls without any problem but I can not receive any. Actually, calls arive
to * but the
Hi Friends,I have installed Asterisk in my house with public ip. Now, I want to connect to my Asterisk server from my office system with XLite softphone. I was given my Asterisk server public IP in my XLite Domain field. But, it is not connecting. It is giving an error i.e., " Registration error:
Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box.
On 21/07/06, Marco Mouta [EMAIL PROTECTED] wrote:
Did you port forwar in your routerRTP ports ? 1-2 to your *Box ?On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote: Hi,I am experiencing a hard to solve problem with my
Could you post your sip.conf?
On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote:
Yes, of course. SIP, RTP and IAX ports are port forwarded to the * box.
On 21/07/06, Marco Mouta [EMAIL PROTECTED] wrote:
Did you port forwar in your router RTP ports ? 1-2 to your *Box ?
On 7/21/06,
From: Filip Dr?gowski [EMAIL PROTECTED]
Subject: Re: [asterisk-users] Two phone numbers, one SIP provider
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=ISO-8859-2; format=flowed
Here is my SIP.conf. (just replaced psswds with *)
Thanks.
[general]
port = 5060
bindaddr = 0.0.0.0
disallow=all
allow=ulaw
allow=alaw
context = from-sip-external
callerid = Unknown
tos=0x68
register=34700758288001:[EMAIL PROTECTED]/34700758288001
Hello,
Asterisk crash with chan_oh323.so
i use asterisk 1.2.9 asterisk-oh323-0.7.3
What's wrong ?
ACF|192.168.0.11:1720|4762_endp|2505|900:dialedDigits|903:h323_ID=903:dialedDigits=903:h323_ID=903:dialedDigits|false;
ACF|80.119.15.247:1721|4760_endp|2505|asterisk-gw:h323_ID|harry
gaillac
Hi,
I think i found your error. you are missing a context for your peer
PeopleCall , this way no context for incoming calls!
Am I wrong?
Hope it helps,
Marco Mouta
On 7/21/06, Jose Limeres [EMAIL PROTECTED] wrote:
Here is my SIP.conf. (just replaced psswds with *)
Thanks.
[general]
port
2006/7/21, David Cook [EMAIL PROTECTED]:
From: Filip Dr?gowski [EMAIL PROTECTED] Subject: Re: [asterisk-users] Two phone numbers, one SIP provider To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Message-ID: [EMAIL PROTECTED] Content-Type: text/plain;
-Messaggio originale-
Da: antonio [mailto:[EMAIL PROTECTED]
Inviato: martedì 18 luglio 2006 14.41
A: 'asterisk-users@lists.digium.com'
Oggetto: ooh323c - cdr problem
The configuration is this:
H323 -- ASTERISK --- SIP
ooh323.conf
amaflags = billing
Diffrent context:
[default]
= goto(|s|1)
= goto(|s|1)
then You can hadle incoming calls in diffrent contexts
register =
:[EMAIL PROTECTED]/
register = :[EMAIL PROTECTED]/
That won't help either. Context is always 'default', but what I
Hi all,
i have a little problem related to queue transfers. i have no idea how to solve it.
the scenario is, a caller has entered a queue. The agents picks up the
call talks and then he decides to transfer the caller into a conference
room and join the room himself with another person by inviting
Upgrading to ver 1.2.10 fixed it.
--- carl Lougher [EMAIL PROTECTED] wrote:
Hi,
Need to get the following working:
1. User calls ext 750.
2. If no answer or busy go elsewhere.
3. If answered and press 1 accept call.
4. If answered and not pressed 1 or timed out then
send call to be
Hi all,
I got the SPA-2100 and I can dial call from other extension to this SIP
ATA, however I got problem make out the call the Asterisk, I believe somethings
to do with the Dial Plan in side the SPA-2100
configuration file.
My setup simple topologies,
ATA connected to my LAN,
I'm sure there probably is other ways to do this but you could write a
script as a cron to use the manager API, filter the data you want, and
store it in a database or text file. But depending how often you run it,
you may miss some data.
Douglas Garstang wrote:
Thanks Johann. Yes, I wish
2006/7/21, Filip Drągowski [EMAIL PROTECTED]:
Diffrent context:
[default]
= goto(|s|1)
= goto(|s|1)
then You can hadle incoming calls in diffrent contexts
Thanks for the solution. This would be really a way to solve the
problem. Storing number=context pairs in the
I got started by experimenting for a replacement to the companies pbx, was sent to a boot camp and certified after several months of learning and have now implemented the system. Eventually it will come down to all of the IT staff being cross trained on how do do most of the maintance on *. On the
Hi,
I've got a bit of a puzzle with my setup. I've got 2 asterisks who are
connected using dundi over iax. Both asterisks use the realtime stuff to
fetch sip and iax client information from a mysql database. As long as all
clients are sip, everything works great. With iax clients however I've
Title: Message
That won't help either. Context is
always 'default', but what I want is a different context on any number. Maybe
oej'speermatch branch solves the problem. But I cannot compile it, There are lots
of ' merge right' tags in
chan_sip.c.
How about a slight modification of my
I have a similar problem with the queues. After getting dialtone for
transfer, i dial extension for transfering , the agent hangsup, the
user is in the queue at this time but didnt get transfered. The cli
show this warning:
WARNING[2867]: chan_h323.c:691 oh323_indicate: Don't know how to indicate
I downloaded svn, downloaded the source files and include files from the svn.sourceforge.net/svnroot/iaxclient/trunk/iaxclient.
My problem is I get errors when I try to compile these projects, although I've included the lib directory in my project. I hope there's someone out there who has an
Hi,
I have problems with two trunks, ZAP3 and
ZAP4. ZAP4 is connected to PSTN line while ZAP3 is connected to analogical
switchboard.
The
system is able to redirect calls from ZAP4 to ZAP3, through an
IVR, but, hanging up doesnt work .
This is the CLI report where you can see, at
Yes you may be right and I going to investigate it bit I thought that
using the context from -sip-external was enough. Specially when I
have defined in extensions.conf that calls belonging to this contyext
should be sent to the extension I want to ring.
Anyhow, will try defining one unique
Hi,
What is the mechanism of asterisk DB? In folder /var/lib/asterisk,
there is a file called astdb. Does it used for storing data of
asterisk DB?
thanks,unplug
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I trying to find a way of using two different callerid numbers. I have the
callerid=Agent 1 33x (in the sip.conf) being the direct dial
phone number, but for internal calls I would like it to show the extension
number. My internal dialplan is.
exten = 3002,1,Set(CALLERID(NUM)=xxx)
exten
There is a nice one called RealTime. Works great and you can even get it to
store email in sql, but I haven't really tested it much.
http://www.voip-info.org/wiki-Asterisk+RealTime
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
-Original Message-
From: [EMAIL
Hi,
I am here in Germany connected to the telephone system with a PRI interface:
00:0b.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN
interface. To let it work, I have read that I have to set the options
overlapdial=yes and immediate=no in zapata.conf. here is my zapata.conf:
2006/7/21, Natambu Obleton [EMAIL PROTECTED]:
There is a nice one called RealTime. Works great and you can even get it to
store email in sql, but I haven't really tested it much.
http://www.voip-info.org/wiki-Asterisk+RealTime
http://www.voip-info.org/wiki/view/Asterisk+Voicemail+ODBC+storage
Hi,
forgot to mention, I have asterisk 1.2.7.1 and zaptel-1.2.5 running.
Sebastian
Sebastian Reitenbach [EMAIL PROTECTED],Asterisk Users Mailing List -
Non-Commercial Discussionasterisk-users@lists.digium.com wrote:
Hi,
I am here in Germany connected to the telephone system with a PRI
-Original Message-
From: Raymond McKay [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 20, 2006 10:16 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Typical Asterisk Company
I'm flying all over the world installing systems for
-Original Message-
From: Matt Florell [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 20, 2006 9:43 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Typical Asterisk Company
Any company. There is no typical company in my experience. I
Can someone point me
to some info or provide a summary on how to allow ODBC access from other hosts
to the Asterisk database?
Wesley A.
SchochetSenior Telecommunications
EngineerSelect Comfort Corporation(763-551-7757651-592-5441*[EMAIL PROTECTED]
All,
Is there a way to have Asterisk read queue.log on startup, or reload, so that
queue stats can be retained between restarts and reboots? It'd would be
especially nice on the reloads, as even a 'reload app_queue.so' clears all your
stats. That COMPLETELY sucks, as every time you make a
I'm working with a Sipura 3000 ATA here. I'm trying to get incoming PSTN calls
on the FXO port to go automatically to Asterisk. I have it working, but I had
to configure the ATA to register with Asterisk, which means that all calls are
being sent to Asterisk with a caller id of the username
They always have Linux Administrators already so they just need to
learn how to edit the conf files. Also, most of them are using
astGUIclient/VICIDIAL which has it's own web interfaces.
MATT---
On 7/21/06, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Matt
Does astGUIclient/VICIDIAL have all the bells and whistles though?
-Original Message-
From: Matt Florell [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 9:27 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Typical Asterisk Company
Hi!
Does anyone know how to connect 2 AAH
IPPBXs so that one extension in A IPPBX can use the PSTN trunk in B IPPBX for
dial out?
Thanks very much
Gidean
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To
On Jul 21, 2006, at 3:01 AM, Woodoo People .pGa! wrote:
No, we aren't intending to check for available g729 codecs
that's why we wanted to have ulaw as a backup when no g729 codecs
where available.
That won't work. If it's trying to use G729, it will still try even
when the licenses are
I want the real caller ID to be sent to Asterisk, which means I don't want the
ATA to register. The badly written Sipura docs aren't clear about how to do
this. Anyone set this up?
I am having the same problem...
Cheers,
Jorge Mauricio
--
blog
http://djmaucom.blogspot.com
On Fri, 2006-07-21 at 09:23 -0600, Douglas Garstang wrote:
I'm working with a Sipura 3000 ATA here. I'm trying to get incoming PSTN
calls on the FXO port to go automatically to Asterisk. I have it working, but
I had to configure the ATA to register with Asterisk, which means that all
calls
Check out DUNDi for what you're trying to accomplish. There's a few good guides around for DUNDi on AAH/Trixbox.I wrote one here: http://blog.thegoldfish.net/dundi-tutorial-for-asteriskhome/
Tijmen wrote a great one here: http://kneh.xs4all.nl/tijmen/asterisk/Using%20DUNDi%20with%20Trixbox.pdfAlex
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 9:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
Asterisk
On Fri, 2006-07-21 at 09:23 -0600, Douglas
Can't put it in a realtime database. We have multiple Asterisk boxes in a
cluster, and it's a well known fact that multiple Asterisk boxes using realime
cannot query a common MySQL database. Sounds crazy, but true.
Doug.
-Original Message-
From: Marco Mouta [mailto:[EMAIL PROTECTED]
Emphasis here being on 'a few'.
-Original Message-From: Alex Robar
[mailto:[EMAIL PROTECTED]Sent: Friday, July 21, 2006 9:58
AMTo: Asterisk Users Mailing List - Non-Commercial
DiscussionSubject: Re: [asterisk-users] How to connect 2
AAHCheck out DUNDi for what you're
Why sending documents privately if this issue could be a problem for
many users in this list? I think it would help much more if we share
knowledge.
On 7/21/06, Douglas Garstang [EMAIL PROTECTED] wrote:
-Original Message-
From: Dave Cotton [mailto:[EMAIL PROTECTED]
Sent: Friday, July
I don't think there is. It would be rather overkill for what the app_queue
does; there are a number of queue stats packages, commercial and free,
that will provide a better approach to gathering stats for the purpouse of
running a call center or an inbound queue.
l.
On Fri, 21 Jul 2006
Not sure what you mean by bells and whistles, take a look at the project site:
http://astguiclient.sf.net/
MATT---
On 7/21/06, Douglas Garstang [EMAIL PROTECTED] wrote:
Does astGUIclient/VICIDIAL have all the bells and whistles though?
-Original Message-
From: Matt Florell
I have clients in a remote location and therefore do
not have access to their Polycom phone to input the ftp information. Is there a
method to input the info without going through the phone?
Steve
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Yes, except that if the queue.log file was read by asterisk, and read at
restart/reload, they could be pulled with the Manager interface. We are running
three Asterisk boxes here in a cluster, and being able to pull the stats from
the Manager interface is relatively easy.
I was just looking at
security implications notwithstanding...
can you temporarily forward an external port on their router to port 80
of the phone and configure it via its webserver?
Stephen Murphy wrote:
I have clients in a remote location and therefore do not have access to
their Polycom phone to input the
Well, as I said below...
Can the user configure, on a per extension basis:
- incoming and outgoing black-lists and white-lists with hierarchical
management so that a company can set blocking and have it override user
settings etc.
- PIC codes, rate centers
- findme/followme with caller id based
pass via dhcp
if you have control of their router, just redirect dhcp requests to
your boot server and control all from there.
On Jul 21, 2006, at 11:24 AM, Stephen Murphy wrote:
I have clients in a remote location and therefore do not have
access to their Polycom phone to input the ftp
I'm using asterisk 1.2.7.1 on ubuntu dapper. It was working.
But today, without changing nothing in the config, and without
connecting-desconnecting anything, it began to give me this error:
Jul 21 12:31:51 WARNING[6333]: chan_sip.c:12637 reload_config: Failed to bind
to 10.152.58.9:5060:
The webserver, I believe, does not have the ability to add the ftp info. I
can get into their system via vnc and can access our pbx via ssh.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Friday, 21 July 2006 9:30 AM
Can you provision the phone then ship it to them or are you shipping direct to them?On 7/21/06, Stephen Murphy
[EMAIL PROTECTED] wrote:
I have clients in a remote location and therefore do
not have access to their Polycom phone to input the ftp information. Is there a
method to input
or if you know the exact sip rom in the polycoms, you could tell the
clients the exact button presses they'd need, but that would be more
cumbersome and prone to error:
e.g.,
Menu,3,2,4,5,6,Enter,1,1,Down Arrow,Select,Down Arrow,Edit,Right Arrow
till 'Static' appears,Exit,Down Arrow, etc...
This will enable me to add the required ftp info to each phone? What
interface do I use - the web?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jerry Jones
Sent: Friday, 21 July 2006 9:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
You'll get no arguements from me on that front. It's why I wrote my guide... There's very little DUNDi documentation out there that explains what's going on. The only decent ones I found were config examples from VoIP-Info, but nothing was explained, and they didn't even work for me. DUNDi is a
They already have the phones and every
phone is setup on an individual basis. I have setup all the config files and
ftp site for provisioning but I can not input the ftp info on each of the
phones. I can always drive there however I would prefer to do it all remotely.
From:
I don't want to involve the clients as something is sure to go wrong.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mojo with
Horan Company, LLC
Sent: Friday, 21 July 2006 9:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
I think what he meant was that you can use your DHCP server to pass a TFTP boot server in a DHCP request. So if the remote router passes DHCP requests to your server, and your server returns the boot server, your phone should get it's config fine.
Unless I misunderstood what Jerry was saying. I
As long as the phones are currently configured to get their boot server
name via option 66 of the dhcp server's response, and you can control
the dhcp server to send the ip you desire, this will allow you to run
the ftp server wherever you choose.
Then you will construct or place in the ftp
Douglas Garstang wrote:
I'm working with a Sipura 3000 ATA here. I'm trying to get incoming PSTN calls
on the FXO port to go automatically to Asterisk. I have it working, but I had
to configure the ATA to register with Asterisk, which means that all calls are
being sent to Asterisk with a
I am looking for a windows softphone (IAX preferably) that supports
USB handsets?
I am trying to put a voip extension onto my wifes work computer... I
have navigated the IT guy and have port 4569 open on the firewall.
I was planning on/hoping to use DIAX, which is a nice compact IAX
Douglas Garstang wrote:
Can't put it in a realtime database. We have multiple Asterisk boxes in a
cluster, and it's a well known fact that multiple Asterisk boxes using realime
cannot query a common MySQL database. Sounds crazy, but true.
You spread some amazing well-known facts on this
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 11:20 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sipura ATA's Forwarding PSTN Calls to
Asterisk
Douglas Garstang wrote:
I'm working
Martin Joseph a écrit :
I am looking for a windows softphone (IAX preferably) that supports
USB handsets?
I am trying to put a voip extension onto my wifes work computer... I
have navigated the IT guy and have port 4569 open on the firewall.
I was planning on/hoping to use DIAX, which
-Original Message-
From: Brian Capouch [mailto:[EMAIL PROTECTED]
Sent: Friday, July 21, 2006 11:33 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Codec Negotiation
Douglas Garstang wrote:
Can't put it in a realtime database. We have
you can trunk the two boxes together with IAX. Check out trixbox.org and search, its been covered a few times.On 7/21/06,
Gidean Chan [EMAIL PROTECTED] wrote:
Hi!
Does anyone know how to connect 2 AAH
IPPBXs so that one extension in A IPPBX can use the PSTN trunk in B IPPBX for
dial out?
Douglas Garstang wrote:
Would you like me to dig up the posts from Keving Fleming stating that this is
known not to work Brian?
As I recall those posts have to do with the way your particular setup
required ARA to work with a failover/redundant cluster system you were
building.
Beyond
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