-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
I tried to modify the extension but * soon receive the extension and the
other hand get busy
i.e
First:
The caller digit 123456 (at the fourth digit connect get the ring back)
exten = _1234,1,dial,sip/25
exten =
Good day all
I have asterisk on a box with one network card
I have a 2 companies setup on the system.
To keep all apart I binded a different ip to the interface,i,o,w eth0
192.168.0.254 and eth0:1 192.168.1.254
And in sip.conf I took the bind setting out
So each company's phones are on a different
My experience with auto-rebooting schemes is that reliability doesn't
improve.
I also reported the non-registration of the Snom phones as a bug. On a few
occasions I found that the phone lost registration, and rebooting or power
cycling the phone didn't help (athorization failed at the *
Title: Fax problem, missing/compressed lines
Hi All,
On my * the faxes in- and outbound are of poor quality. Im missing some lines. I tried different fxs ports, different fax-machines, different cables. Low speed on the fax.
Asterisk 1.0.9-BRIstuffed-0.2.0-RC8j
Zapata.conf:
Hi,
I live in Italy and I have an Asterisk 1.0.7 box with a HFC monoBRI
card connected to my cisco router which is connected to FastWeb provider:
does anybody knows why every time my cisco router turns off, my
telephone connection to Fastweb drops (while internet connectior is ok)?
Restarting
Title: Setting wcte11xp card to use IRQ
Hi, is it possible to set a wcte11xp card to use a certain IRQ? I've tried a few things but it always shares the IRQ with eth0 even though the system has 4 spare ones. I can't set it via the BIOS.
Regards
Lee
Giorgio Incantalupo ha scritto:
I live in Italy and I have an Asterisk 1.0.7 box with a HFC monoBRI
card connected to my cisco router which is connected to FastWeb provider:
does anybody knows why every time my cisco router turns off, my
telephone connection to Fastweb drops (while internet
We schedule a reboot each night at 11/12 so it clears any errors,or
hanging channels,in case for save keeping
Im just not sure then with the 360 that it would keep the panel lights
working.
Got those problems your talking about,lost with the 220's
On Fri, 2005-09-02 at 09:39 +0200, Remco Barende
In article [EMAIL PROTECTED],
C F [EMAIL PROTECTED] wrote:
I believe the c option will do what you want, but to the best of
knowledge is only available for zap channels:
http://www.voip-info.org/wiki-Asterisk+ZAP+channels
Thanks. I knew I'd seen something like it apart from with agents.
On Friday 02 September 2005 10:05, altus wrote:
We schedule a reboot each night at 11/12 so it clears any errors,or
hanging channels,in case for save keeping
Im just not sure then with the 360 that it would keep the panel lights
working.
That depends on the behavior of * when restarting. If *
Did you tried to turn off te option Filter Packets from Registrar on the
Advanced page?
That is the only case, when a snom sends a 403 (= Forbidden).
Regards
Nils Ohlmeier
On Friday 02 September 2005 09:11, altus wrote:
Good day all
I have asterisk on a box with one network card
I have a 2
Sorry, but we are unable to reproduce your problem description here locally
(without an *). If you are able to reproduce it, we guess it must be related
to the *. In this case it could help if you send us SIP-traces of that
scenario, so that we can look into this problem.
Regards
Nils
Hi there,
Im having a problem configuring Routes in ASTCC.
I want to have more than one Trunk in each Route so if
one Trunk is busy the call should be forwarded to the
next available Trunk. This is done in astcc-admin.cgi
by choosing more than one Trunk when creating a Route.
The problem is
Hi everybody :)
I am a new member here and hope that someone gives me a hint for my problem:
Let's say I am at work and my SIP phone (KPhone in my case) is connected to my
private Asterisk. I want to call my wife at home so her SIP phone rings. She
does not pick up the phone (maybe she is
Hi,
We are facing an issue with ALL calls simply dropping during peak times
(this is happening upto 10-13x an hour) on certain gear:
We have a setup like this:
Client --- SIP --- Asterisk --- IAX --- Asterisk --- ISDN ---
Provider
Any ideas?
Regards,
Sahil Gupta
VoiceValley
LTenorio,
I am not sure what you mean between Terminal and Gateway.
The voip providor in China sell the H323 service as a package where
you get a h323 compatible handset and a landline number, the phone
comes preconfigured to connect to their gatekeeper to make and receive
calls.
So what I
On 10:44, Fri 02 Sep 05, Hauke Zuehl wrote:
Hi everybody :)
I am a new member here and hope that someone gives me a hint for my problem:
Let's say I am at work and my SIP phone (KPhone in my case) is connected to
my
private Asterisk. I want to call my wife at home so her SIP phone rings.
Hi Michiel :)
Am Freitag, 2. September 2005 11:25 schrieb Michiel van Baak:
We do something like this:
exten = michiel,1,Dial(SCCP/michiel,15,t)
exten =
michiel,2,Dial(SCCP/michielIAX2/vanbaak/${CELL_MICHIEL},40,t)
Aehhh...SCCP? Never heard about that...but I will google for this so thanks
On 11:29, Fri 02 Sep 05, Hauke Zuehl wrote:
Hi Michiel :)
Am Freitag, 2. September 2005 11:25 schrieb Michiel van Baak:
We do something like this:
exten = michiel,1,Dial(SCCP/michiel,15,t)
exten =
michiel,2,Dial(SCCP/michielIAX2/vanbaak/${CELL_MICHIEL},40,t)
Aehhh...SCCP? Never
Hauke Zuehl ha scritto:
Aehhh...SCCP? Never heard about that...but I will google for this so thanks
for your answer :)
protocol for cisco phones
http://chan-sccp.berlios.de/
http://chan-sccp.org
Sergio
___
--Bandwidth and Colocation sponsored by
Try amportal which is a sourceforge project, but remember if a cell is
one of the numbers there will always bea delay while it is dialed and
it might go to voicemail anyway.
on Friday 09/02/2005 Hauke Zuehl([EMAIL PROTECTED]) wrote
Hi everybody :)
I am a new member here and hope that
Hello BJ,
Thursday, September 1, 2005, 2:06:43 PM, you wrote:
BW Issue #3644 has recently been committed to CVS-HEAD which
BW allows for full device state notification via subscriptions for
BW Snom 360 and other supporting phones w/o the need for additional
BW patches.
Leds are working using
I am having the exact same issues. I even tried to madk my IAX peer
account in both the database, and in the iax.conf file (with different
names, but same info) and the static one works, but not the database
one. I am using 1.2.0-beta1.
If I specify the user:[EMAIL PROTECTED] on the
Hello,
These are error messages I get when I try to call a number over CAPI channel.
-- Executing SetCallerID(SIP/xlite1-3b80, 0) in new stack
-- Executing Dial(SIP/xlite1-3b80, CAPI/hfcpci/b17) in new stack
data = hfcpci/b17
capi request for interface 'hfcpci'
==
On Fri, 2 Sep 2005, Konrads Smelkovs wrote:
Hello,
These are error messages I get when I try to call a number over CAPI channel.
-- Executing SetCallerID(SIP/xlite1-3b80, 0) in new stack
-- Executing Dial(SIP/xlite1-3b80, CAPI/hfcpci/b17) in new stack
data = hfcpci/b17
Hi!
Asterisk 1.0.9 (maybe also earlier versions?) contains a bug the
effectively disables subscriptions for phones with multiple
regististrations in place.
When processing a nonce response as a result to an 407 authentication
request Asterisk with SIP DEBUG reports Found peer YYY even though
There is a patch that Frank Sautter has been working on to get call pickup and the record button on the Snom's working. It's 5014 in Mantis. The current patch available there will likely needmanual interventionto get it to merge with the current CVS-HEAD now that 3644 has been committed.
Frank
I was wondering on the choice of chan_capi and chan_misdn.
What should be the choice for a feature rich use of isdn-bri channel?
I just installed mISDN for my modem (ISDN BRI - I write about my
experience in another post).
mISDN is hailed as "the new ISDN stack of the Linux kernel version
Got a cheap ISDN modem with a so called Cologne chip for setting in TE
mode, with 1-S0 port.
I decided to use mISDN because it is the new interface for
linux.
My system is a Mandrake, sorry Mandriva, 10.2. already with TDM-400
card installed and working fine.
I spent a few hours trying to
Hi,
We're located in London (Ont) but have setup asterisk for
clients from Mississauga to Windsor (as well as lots of international clients
assisted remotely). We've also taken on roles from one extreme to the
other (i.e. deliver a turnkey solution including hardware, to just giving advice
Sorry
all - that was mean to go directly to the user, not the whole list
:)
From:Sent: Friday, September 02,
2005 7:54 AMTo: 'Asterisk Users Mailing List - Non-Commercial
Discussion'Subject: RE: [Asterisk-Users] Any one in Toronto / Canada
can help me!
Hi,
We're located in London (Ont)
Hello Goran,
Yes, you are right !
I have read too quickly your question, sorry.
I have checked myself with my * and same thing appears here.
If you will have the solution before me, please, post it :-)
For F.T. prices list, it is not realy easy...
You can start by browsing from here :
To be simple, you can register to any gk as Gateway or as Terminal,
some gk restricts the way the users could be registered as. Oh323 advertise
itself as gateways and if your provider does not support registration as GW
it will not work.
LTenorio
-Original Message-
From: Steve
That is correct. Normally the layer 3 switches include advanced features
such as QoS but they may be available on simpler layer 2 switches.
I think the key words to look for are 'Managed, QoS (802.1p) with
priority queues, VLAN, (802.1q)'...maybe even PoE if you go with some
SIP phones in the
This is an old problem, but I have been unable to locate a solution so
far :-(
I want to set-up asterisk so that it mimics a typical answerphone ie.
someone calls in - you call screen in case it's:-
a) a telemarketer
b) someone else you don't won't to talk to
c) someone you don't want to talk
On Fri, Sep 02, 2005 at 08:46:10AM +0200, Thomas Petersen wrote:
[EMAIL PROTECTED] wrote:
The caller digit 123456 (at the fourth digit connect get the ring back)
exten = _1234,1,dial,sip/25
exten = _123456,dial.sip/30
in this way the other hand get the first estension and does not pass
This is a well-known issue with the current drivers, although nobody's
really
stepped up to identify when exactly this started happenning (as the driver
code was changed) or why.
Do I understand that this would not be a problem with an 'older' version
of Asterisk ? If so, any idea
LTenorio,
Then this is my problem. I can only register to the gatekeeper as a
terminal, they do not allow me to register as a gateway.
Is there any other way I can get asterisk to register to the
gatekeeper as a terminal.
Thanks again for your help.
Kind Regards,
Steven Ducat.
On 9/2/05,
Has anyone else had
problems with users being able to press key tones during a voice prompt? I have
a few users complaining that some systems will not recognize key presses during
them.
using current
CVS-HEAD, linksys PAP2 UA's, rfc2833 dtmf mode.
Thanks
Sherwood
McGowan
I see this error when trying to make calls from my asterisk server. How
can I solve this problem.
Sep 2 08:25:03 NOTICE[1403]: -- Registration for
'[EMAIL PROTECTED]' timed out, trying again
___
--Bandwidth and Colocation sponsored by Easynews.com --
There was a FIX that corrected this (or at
least a very similar issue) about a month ago, it seemed to be during short
prompts like digits. I was using inband dtmf though.
If your code is more than a month old try
updating it.
From:
[EMAIL PROTECTED] [mailto:[EMAIL
Sherwood McGowan wrote:
Has anyone else had problems with users being able to press key
tones during a voice prompt? I have a few users complaining that
some systems will not recognize key presses during them.
You are using Backgroudn() to play the prompts?
--
Andreas Sikkema
Yes, but source code that compiles to a client library should be OK,
isn't it?
For example
major open source scripting languages, PhP, perl, python etc. have Oci8
modules.
DB
On Thu, 2005-09-01 at 12:35 -0400, Kanuri, Seshu (Company IT) wrote:
And to add to what Kevin said, we don't
Hi, all
Still struggling.
Here is my iax.conf entry:
[voipbuster]
type=peer
host=iax.voipbuster.com
secret=MYSECRET
notransfer=yes
context=pignet
Here is my extension.conf entry:
exten = _0.,1,SetCallerID(CID Name CIDNUMBER)
exten = _0.,2,Dial(IAX2/[EMAIL PROTECTED]/00613${EXTEN:1})
(I am
Actually, the problem exists with outside services, not ours.
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-Andreas Sikkema
-Sent: Friday, September 02, 2005 9:07 AM
-To: Asterisk Users Mailing List - Non-Commercial Discussion
-Subject: RE:
Hauke Zuehl wrote:
Hi everybody :)
I am a new member here and hope that someone gives me a hint for my problem:
Let's say I am at work and my SIP phone (KPhone in my case) is connected to my
private Asterisk. I want to call my wife at home so her SIP phone rings. She
does not pick up the
On Friday 02 September 2005 07:47, Alex Ongena wrote:
This is a well-known issue with the current drivers, although nobody's
really stepped up to identify when exactly this started happenning (as
the driver code was changed) or why.
Do I understand that this would not be a problem
Just something I was thinking about today...
The communications infrastructure of New Orleans and surrounding
communities has been obliterated. It occurred to me that it may be
possible to provide outgoing phone capability using a few Asterisk
servers, connected wirelessly (routing provided using
Hi Jeremy,
I think the first place to start on this is some corporate funding by
one of the voip providers (you could pretty much guarantee US only
calls).
Once you have this, the rest (servers, generators, sip phones etc)
fairly easy to come by.
If you do get this off the ground I would be
Title: FW: defunct email kill list
Is there an email address that we can forward defunct emails like the one below to so that they can be taken off the mailing list. I appreciate it means work for someone but this email is probably being sent out to 400 original posters a day if that makes
is where anyone who can tell me how it's possible to set nationalprefix
internationalprefix for a single isdn-card and not for all installed cards
?
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
The national guard and/or army routinely implements VoIP over wireless
in situations where comm is lost, I did see an news release that the
Guard started this project in the south the day after the disaster hit.
The key is not the VoIP infrastructure, that is the easy part (one ss7
Sonus
Great idea Dean,
I would also suggest that maybe instead of looking to set up within the
disaster zones that you consider setting up in the relocation zone (eg
where people are being sent to) yes they have payphones there but having
a bank of 20 grandstream phones connected a T1 is a far smarter
Hi,
I was running asterisk 1.0.7 but we've upgraded now to CVS-HEAD.
I've noticed this.. and several people have commented that audio
quality seems to have gone down hill. Just going
phone--asterisk--PRI. I've not changed the configuration files
during the upgrade.
sip.conf is:
allow=ulaw
[EMAIL PROTECTED] wrote:
I am
now looking for a small(?) paging amp, that I could hook 3 or 4 horns to.
I would like to just have the * extention be routed to a soundcard and out
an output, so I would like an amp that is voice signal activated.
The Parasound Zamp Zone amplifier would
Zeeshan Zakaria wrote:
Hi,
Ive Sipura 1001 phone
adapter. In the settings it has
separate Line 1 and Line 2 tabs, which apparently means it can control
two
separate phone lines. Ive [EMAIL PROTECTED] server and want to setup two
different extensions for two
I just noticed one of my phones on a call as:
400 e8516be0704 00101/02552 ilbc No Rx: ACK
Why would it default to ilbc when ulaw is available (worked with 1.0.7), etc?
On 9/2/05, Matt [EMAIL PROTECTED] wrote:
Hi,
I was running asterisk 1.0.7 but we've upgraded now to
Dias Badekas wrote:
Yes, but source code that compiles to a client library should be OK, isn't it?
For example major open source scripting languages, PhP, perl, python etc. have
Oci8 modules.
I cannot say; I only know that I've been told that the Oracle client
library license makes it not
Hi,
I figured it out.
The problem was that I was using examples of how to use the rg-group macro from
an older asterisk, and it looks like they changed that macro. It used to take
arguments set in variables, now it takes them normally. If I define my
ringgroups like this it works:
exten =
On Fri, 2 Sep 2005, Dias Badekas wrote:
I was wondering on the choice of chan_capi and chan_misdn.
What should be the choice for a feature rich use of isdn-bri channel?
It depends on the card you are using (e.g. maybe a special capi driver for
that card exists like FritzCard) and on the
Do you ever wish you could just revoke a question you asked *ugh*...
someone changed the setting on the PHONE grr
On 9/2/05, Matt [EMAIL PROTECTED] wrote:
I just noticed one of my phones on a call as:
400 e8516be0704 00101/02552 ilbc No Rx: ACK
Why
let's suppose I have this dialplan :
exten = _X.,1,Playtones(ring)
exten = _X.,2,Dial(CAPI/contr1/${EXTEN},,g)
exten = _X.,3,AGI(update)
where update updates some db tables we have based on the type of extension
Now, from the wiki :
If the /g/ option is specified, and the called party hangs
On 9/2/05, Chris A. Icide [EMAIL PROTECTED] wrote:
I am having the exact same issues. I even tried to madk my IAX peer account in both the database, and in the iax.conf file (with different names, but same info) and the static one works, but not the database
one. I am using 1.2.0-beta1. If I
Hello
My asterisk is stoping. i am using asterisk with ser
on same mechine
here is the asterisk trace
-- Setting call duration limit to 3000 seconds.
Sep 2 15:58:12 WARNING[10334]: rtp.c:852
ast_rtp_new_with_bindaddr: Unable to allocate socket:
Hi all,
I just got my long awaited FXS modules for a TDM400 card, and was a bit
puzzled about how my phones and the FXS interface worked together. Here
goes:
In Sweden (that's where I am located) phones traditionally are connected in
daisy chain configuration. This means that a phone in
Johan van Tongeren wrote:
On my * the faxes in- and outbound are of poor quality. I’m missing
some lines.
TxFAX and RxFAX don't yet support ECM, and so you're going to miss an
occassional line. The frequency will depend upon the quality of the
connection between the two endpoints.
Lee.
In article [EMAIL PROTECTED],
Simone Cittadini [EMAIL PROTECTED] wrote:
So I may use the h extension, but also from the wiki :
Use with great care: Apparently some channel variables get destroyed
when the call is hung up, and those variables aren't available anymore
(or have inconsistent
On Friday 02 September 2005 09:33, Martin wrote:
On Friday 02 September 2005 07:47, Alex Ongena wrote:
This is a well-known issue with the current drivers, although
nobody's really stepped up to identify when exactly this started
happenning (as the driver code was changed) or why.
Probably already been covered but you do have the power connector on the
board connected right?
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Martin
Sent: Friday, September 02, 2005 12:32 PM
To: Asterisk Users Mailing List - Non-Commercial
On Friday 02 September 2005 10:43, Jonathan k. Creasy wrote:
Probably already been covered but you do have the power connector on the
board connected right?
-Jonathan
Yes, from my earlier email
Yes, the 4-pin power connector from the 400W PS is plugged in and there is
only one H/D and one
i'm having this error message when trying to run the agents-feature
Sep 2 17:37:40 WARNING[10445]: pbx.c:1645 pbx_extension_helper: No application
'AgentsLogin' for extension (from-internal, 28, 1)
while chan_agent.so is beeing loaded i still don't seem to have access to the
commands like
On Friday 02 September 2005 12:31, Martin wrote:
Port #1
Standing voltage = -48.4 VDC
Ringing voltage = 42.7 VAC
Port #2
Standing voltage = -48.2 VDC
Ringing voltage = 42.7 VAC
Port #4
Standing voltage = -44.4 VDC
Ringing voltage = 43.5 VAC
Sounds like you have defective modules or
I thought that I would try this on iax.conf as well however I still get
asterisk asterisk as the callerid name and num.
I have the latest CVS as of 8/17/05
Has anyone have this working with iax incoming?
Thanks
Ben
That worked. The following line also got rid of asterisk without
entering
Just upgraded to most recent CVS and now I get this:
pri_dchannel: PRI Error: We think we're the CPE, but they think they're
the CPE too.
when starting asterisk. Needless to say, I can't run asterisk without my
PRI.
Guess I'll start reverting code backwards day by day until I find the
On Friday 02 September 2005 10:51, Andrew Kohlsmith wrote:
On Friday 02 September 2005 12:31, Martin wrote:
Port #1
Standing voltage = -48.4 VDC
Ringing voltage = 42.7 VAC
Port #2
Standing voltage = -48.2 VDC
Ringing voltage = 42.7 VAC
Port #4
Standing voltage = -44.4 VDC
What Steve posted is the output of the command 'show application RxFAX' on
the Asterisk CLI.
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Woody Sturges
Sent: Wednesday, August 31, 2005 11:15 AM
To: Asterisk Users Mailing List - Non-Commercial
Just use the standard * H323, will let you register as terminal.
In my poor knowledge, base on a couple of test with both, H323 let you
register as terminal and OH323 as GW. Never tried the ooh323 on HEAD.
LTenorio
-Original Message-
From: Steve Ducat [mailto:[EMAIL PROTECTED]
Sent:
ChB wrote:
i'm having this error message when trying to run the agents-feature
Sep 2 17:37:40 WARNING[10445]: pbx.c:1645 pbx_extension_helper: No application
'AgentsLogin' for extension (from-internal, 28, 1)
Do you not see what is wrong here? AgentsLogin is not the same as
AgentLogin.
For those of you who wanted to know how the AG-468 (4xFXS) unit work (or
it doesn't work), here is my personal experience.
I had a problem from the start. The units ship from the factory set to
static IP 192.168.1.200 even though it has a DHCP option in LAN Setting.
So if you are on a different
I am using GSM end-to-end. I want to avoid transcoding at all costs. I am
looking for a low cost hard phone that's $150 with gsm. Anything that
anyone knows of that fits the bill?
___
--Bandwidth and Colocation sponsored by Easynews.com --
[EMAIL PROTECTED] wrote:
I thought that I would try this on iax.conf as well however I still get
asterisk asterisk as the callerid name and num.
I have the latest CVS as of 8/17/05
Has anyone have this working with iax incoming?
Thanks
Ben
Every since moving to CVS HEAD, CallerID
On Friday 02 September 2005 12:57, Martin wrote:
I would be really curious if anyone else has the capability (digital
multimeter) and enough knowledge to test their incoming CO (Central Office
line) and board outputs.
I will test mine at lunchtime today.
-A.
I have one GXP-2000 and i prefer the SPA 841 of SIPURA, (look their are same price) y the price is not a problem POLYCOM or SNOM
On 9/1/05, Joe McConnaughey [EMAIL PROTECTED] wrote:
Check out the Aastra 9133i. Fantastic phone for about $179. I have two of them and will be adding more. More bang
The new version of heartbeat (http://linux-ha.org/GettingStartedV2)
supports up to 16nodes. I was wondering if anyone has tried it with
Asterisk.
The biggest hurdle would be to configure multiple instances of
Asterisk on the same box. Anyone configure more than one copy of
asterisk on the same
Sounds like you are connected to a pbx instead of a carrier switch,
either asterisk or the pbx need to be set to network
Pri_cpe or pri_net in asterisk...
Caller id name might play a role in which you decide to set to network.
-Original Message-
From: [EMAIL PROTECTED]
I'm looking for some
information on how the CDR gets the data for the source and destination on the
records. My current system sets the callerid to private, via
SetCallerID(Private) in the dialplan. Unfortunately, this means there's
records in my CDR that have no source on them, and as such
Hehe, yes, fixed it in the meanwhile. after a long search i found that the
agent-module missing but when it still didn't worked i was so fed up with it
that i didn't recognize the typo ;-)
On Fri, 02 Sep 2005 11:01:33 -0500
Kevin P. Fleming [EMAIL PROTECTED] wrote:
ChB wrote:
i'm having
20050902
http://chan-sccp.berlios.de/
- added support for call forward (all and busy)
The call forward buttons are available on the ringout state (it is
because I need to verify that the extension exists) to activate it and
on the offhook state to deactivate it. There is a problem
How do you find out what company owns a certain 877 number? Currently it is
disconnected and I have a friend that wants acquire it and port it to my
system. I have Googled and have found people that do this for a living, but
surely there must be an easy way to find out without paying a couple
Hi,
Quick question. With an old phone
system a receptionist receiving a call has 1 button to push to transfer
calls to a specific extension, with Asterisk, a receptionist would actually
put the caller on hold, pick up another line, call the extension, ask if
the person is available, hang up pick
Dana Olson wrote:
Chris,
Thanks for the reply.
I checked those settings, and they were commented out, so I uncommented
them. I assumed you meant rtnoupdate=yes, so that's what I put, but
that didn't work. I tried rtnoupdate=no, and that didn't work either.
I do have a register
I h ad a similar problem. I have a number I want, which is unused.
Call verizon, ask them who the ld owning company is. Mine ended up as
mci, then call them and ask, worth a shot...
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of calvis
Sent:
Dean Collins [EMAIL PROTECTED] wrote:
(Article auto-converted from unnecessary HTML to nice plain text.)
Is there an email address that we can forward 'defunct' emails like the
one below to so that they can be taken off the mailing list. I appreciate
it means work for someone but this email
I would be willing to give time and be able to setup / manage some
devices. I was thinking that you could use standard POTS phones to a
adtran tsu600 asterisk t1 to fxo to pots. the wifi phones would be nice
but i think they would tend to walk off.
JasonOn 9/2/05, Damon Estep [EMAIL PROTECTED]
Quick question. With an old phone system a receptionist receiving a
call has 1 button to push to transfer calls to a specific extension,
with Asterisk, a receptionist would actually put the caller on hold,
pick up another line, call the extension, ask if the person is
available, hang up pick
:o)
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Hi,
I have 2 h323-gateway (clarent-h323 cisco
AS5x00-h323) with different fastStart/h245Tunneling
mode configuration.
I'm using fedora core 1 with
- asterisk-1.0.9
- asterisk-oh323-0.6.6
Anyone know how to configure oh323.conf with multiple
h323-gateway ?
regards,
So, with this i solve the issue on main office. But what about the two
remote? they are so little that they will not let me place another *
box there. The phones will be SIP and they are like this
INTERNET--PIX--LAN(machines and sip phones). The pixes in those two
offices have an ipsec tunnel with
I can
give you my experience: It was a hard hurdle to gain acceptance of Asterisk in
my office (6 companies) simply because a receptionist position is a weak point
in Asterisk. Old time receptionists used to Meridians hate learning anything
new, even FOP is not acceptable. However, what
I would like to have my asterisk ring my cell phone and let me know when a new
voicemail arrives. In fact if it would automatically put into the voicemail
menu that would be cool too. In the future I will probably want it to IM me.
Are there good examples somewhere of doing stuff automatically
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