Gentelmen (and ladies too of course),
Just a quick question.
I run an internet provider here in Japan and we want to start offering
US DIDs to some of our US military customers.
Does anyone have a link to some good information about DIDs and setting
them up under asterisk? Also, perhaps a
Sorry Patrick,
I was mistaken here.
The Diva Server for Linux drives currently only support Little Endian machines.
Unfortunately the PPC based chipsets use Big Endian.
There is a discussion about this here:
http://www.cs.umass.edu/~verts/cs32/endian.html
Thanks
David
-Original
My local telephone provider on PRI lines gives billing signalling. Is there any
way to use this signalization? I would like to store those information's in
database (MySQL). Has anybody done something similar? So far I have export CDR
that Asterisk generates, in MySQL. Those information I'll
On 12/8/05, Alphonse Ogulla [EMAIL PROTECTED] wrote:
Greetings All,
I intend to buy a Polycom IP 500CS with part number 2201.11500.001 but
I'm not sure if it will work with Asterisk. Is this a SIP capable
phone? Moreover, what does CS stand for? Done some research at
Found it! For some reason [EMAIL PROTECTED] had chosen to build itself without
app_meetme.so!
After building this module by hand, all worked! :-)
Evert
Evert Meulie wrote:
Read before you reply... ;-)
To be 100% clear on zaptel/ztdummy, here's the output of my lsmod:
[EMAIL PROTECTED]
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
I'm running asterisk on FC4. All works fine, including musiconhold.
I tried installing ztdummy as directed, since the documentation
indicates that ztdummy is required for good music quality.
However, installing ztdummy on FC4
In article [EMAIL PROTECTED],
Ryan Laginski [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
-=-=-=-=-=-
Hi,
Ever since upgrading to 1.2.0, Asterisk occasionally core dumps. I'm
currently on 1.2.1 with the same problem.
It crashes when an incoming call (zap) dials an extension. It will ring the
Hi list.
I just want to share this information with all the
Aastra IP phone users that has or am going to switch to FirmWare version 1.3.x.
Ive just installed a bunch of 480i phones
connected to a local Asterisk 1.0.9. Using the pre installed 1.2.x firmware the
sound quality and the
i have an asterisk box behind the NAT ,when i try to
send calls through Sip to the voip provider server the
call is answered but in a one way calling,I hear the
voice of the other side just for 4 seconds and then
stop but the call do not hangup.
SOmetimes this can be due to the client using
Hi all,
Im testing canreinvite = yes in my sip.conf
with snom190 and a Atcom320.Atcom320 seems support re-invite, but the snom190?
Does anyone known if this phone support it?
How I can be sure that it works?
Giordano
___
Dakota,
Looking at it objectively, Asterisk has many benefits over traditional PBX
systems, yet you should be aware of some of the limitations.
Benefits:
1. Open source / low-cost of ownership / operates on cheap PC hardware. You
get voicemail, IVR, hunt-groups etc. without additional fees. Last
Hi Rob,
you could build a simple Perl or Python script to create incoming calls
using callfiles. We have used such a strategy and it seems to be working.
l.
On Thu, 08 Dec 2005 14:15:50 +0100, Rob Hillis [EMAIL PROTECTED]
wrote:
I'm currently starting development of an add-on to a program
Tobias Jönsson wrote:
On Thu, 17 Nov 2005, Frederic Steinfels wrote:
Last January I told KPJ that I can still not use my Simens Gigagaset
cordless phones and sent him some bug reports. He promised me to fix
this bug several times but nothing happened. The problem is that the
phone is
Hi list,
I m wondering is there a bug regarding the zaptel/ztdummy? or it is
just my misconfiguration?
As the log shown zaptel is refering to the /dev/zap/ctl when it suppose
to refer to /dev/zapctl as i m concering.
I m using
1) zaptel-1.2.1
2) kernel 2.6.12-1.1381_FC3
3) no zap's PCI
i updated to actual
sVN but now when i call with my phone i get a hangup when the clal should be
ringing.
with the branch all
is fine.
___
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Asterisk-Users mailing list
To UNSUBSCRIBE or update
Hi
Is there a way to get incoming calls to go to the same agent that handled
them previously, based on the Caller ID? This would be great for support /
helpdesk, since the caller doesn't have to explain the whole problem to each
agent.
Does anyone know?
We're using [EMAIL PROTECTED] 1.5,
Does the CIDNUM and
CIDNAME is not any longer working?
How do i get the
parts from the CALLERID?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
In article [EMAIL PROTECTED], L [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
Hi list,
I m wondering is there a bug regarding the zaptel/ztdummy? or it is just my
misconfiguration?
As the log shown zaptel is refering to the /dev/zap/ctl when it suppose to
refer to
/dev/zapctl as i m concering.
Hi Tony,
I have miss the README.udev part and this is user mistake.
Thanx for pointing me to the right direction.
Regards
-L-
Tony Mountifield wrote:
In article [EMAIL PROTECTED], L [EMAIL PROTECTED] wrote:
-=-=-=-=-=-
Hi list,
I m wondering is there a bug regarding the
Are you running udev? If so, you need to follow the directions in
README.udev...
http://www.voip-info.org/wiki-Asterisk+Fedora+Core+3
Help!: line 146: Unable to open master device '/dev/zap/ctl'
You are probably running udev and don't know it.. were you paying
attention during the make?
Hi!
I can't have Asterisk do a DNS SRV lookup because Asterisks SRV
lookups are broken. If I issue a series of Dial commands, such as
this:
exten = 1_.,1,Dial(SIP/[EMAIL PROTECTED],20,tr)
exten = 1_.,2,Dial(SIP/[EMAIL PROTECTED],20,tr)
How about you use ChanIsAvail() before each dial
HI!
how are you people. i am a newbie in asterisk and
voip.
i need your help.
the scenerio is like this.
1.all local SIP users will be connected to asterisk
via IP.
2.PSTN will be connected to AS5300.pstn will give us a
local prefix like 333. so any one calling at
333 will go to my
On Fri, 9 Dec 2005, David Waugh wrote:
Sorry Patrick, I was mistaken here. The Diva Server for Linux drives
currently only support Little Endian machines. Unfortunately the PPC based
chipsets use Big Endian.
The melware.net drivers (part of kernels 2.6) do support Big-Endian.
Armin
Your other option is to setup the OpenSER boxes in a truly redundant
configuration using Linux HA (www.linux-ha.org). That way you setup all
your PSTN calls to forward to one shared virtual IP between the boxes. One
of the boxes is the Master, the other is the Slave. There is a heartbeat
between
On Fri, 09 Dec 2005 00:36:18 -0500
Matthew matthew@zeut.net wrote:
Hello, has anyone taken their cell phone number and
ported it over to a voip provider? If so, what voip
provider and what was your experience?
Matt
Matt,
I have done this. I had a cell number with ATT
Wireless and
Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
It sounds like you are using a version compiled without USE_RTC on one
of the newer kernels that has fewer than 1000 jiffies per second.
I'm using zaptel from CVS (cvs.digium.com). I just did an update (many
files
Hi,
I don't think it is impossible, though not yet supported by Asterisk
out-of-the-box. You could have a general queue plus a queue per each
agent, and you would route the call to each agent based on the caller*id.
This might end up spoiling the advantage of a queue, meaning that you
Hello, has anyone taken their cell phone number and
ported it over to a voip provider? If so, what voip
provider and what was your experience?
Matt
Matt,
I have done this. I had a cell number with ATT
Wireless and first ported it to Broadvox Direct. There
service was ok
I chose this method and have been happy with the results.
-Jonathan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:51 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
We've just seen that one of our servers is an hour out (it reckons that
it's 15:02 instead of 14:02).
Can I change the time when * is running ? I don't want to try just in
case it causes * some grief.
Julian.
___
--Bandwidth and Colocation
Rich Adamson wrote:
Just an FYI... not all cell numbers are portable.
Do you have any more information on this? I read somewhere that
sometimes you can port a number to a VoIP provider but not be able to
port it back to the PSTN because not all PSTN providers will take
numbers from VoIP
An elegant wat to do this would be to have the caller ID and agent the call was sent to stored in mysql (Asterisk can do systems calls to this) and when calls come in do a quick check to the database, if it's matched put it through to the same agent, if the agent is busy it can revert to the
There has been a fair amount of converstaion about this, but I'm not
sure anyone really has this working. I had exactly the same problem
that the button got remapped to a volume up function. The only button
remapping I got working was to map the Transfer button to the # key so
that when you
Julian Lyndon-Smith wrote:
We've just seen that one of our servers is an hour out (it reckons that
it's 15:02 instead of 14:02).
Can I change the time when * is running ? I don't want to try just in
case it causes * some grief.
It can cause some repercussions. I wouldn't recommend changing
Hello *-users,
this is my first mail and here I have my first big problem for you...
I use Asterisk with the Bristuff-Patches and a Digium TE405P plus a
quadBRI-card (european ISDN).
Everything I tried (calls with speech via BRI, SIP and POTS) was succesful
except the current task. I try to
On Fri, Dec 09, 2005 at 02:02:53PM +, Julian Lyndon-Smith wrote:
We've just seen that one of our servers is an hour out (it reckons that
it's 15:02 instead of 14:02).
Can I change the time when * is running ? I don't want to try just in
case it causes * some grief.
keep clocks in sync
Julian Lyndon-Smith wrote:
Kevin P. Fleming wrote:
Can I change the time when * is running ? I don't want to try just in
case it causes * some grief.
It can cause some repercussions. I wouldn't recommend changing the
time backwards by such a large amount while Asterisk is running with
Hi Lars!
I had similar problems when I tried to forward an UMTS video call
(H.324M). You can read the problems and mabye where to fix it on
http://bugs.digium.com/view.php?id=3891
klaus
Lars Poschitzki wrote:
Hello *-users,
this is my first mail and here I have my first big problem for
Hmm,
ls -l /etc/localtime
-rw-r--r-- 1 root root 1323 Nov 25 12:43 /etc/localtime
there's no symlink that I can see. This is CentOS 4.2
Julian
Gavin Hamill wrote:
Julian Lyndon-Smith wrote:
Kevin P. Fleming wrote:
Can I change the time when * is running ? I don't want to try just
in
Yeah,
I was going to change it tonight :) I'll wait until then.
Julian.
Kevin P. Fleming wrote:
Julian Lyndon-Smith wrote:
We've just seen that one of our servers is an hour out (it reckons
that it's 15:02 instead of 14:02).
Can I change the time when * is running ? I don't want to try
Julian Lyndon-Smith wrote:
Hmm,
ls -l /etc/localtime
-rw-r--r-- 1 root root 1323 Nov 25 12:43 /etc/localtime
there's no symlink that I can see. This is CentOS 4.2
OK, I just had a look in /usr/share/zoneinfo and the only files which
were 1323 bytes were for UK + Ireland, so if that's
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
Check ztdummy.h and ztdummy.c and make doubly sure USE_RTC is defined.
It sounds like you are using a version compiled without USE_RTC on one
of the newer kernels that has fewer than 1000 jiffies per second.
I'm using
Try removing the Answer() before the Dial... e.g.:
[spa2100]
exten = _X.,1,NoOp(SIP Call from SPA2100 to ${EXTEN})
exten = _X.,2,Dial(SIP/netvoice-102)
exten = _X.,3,Hangup
Regards
Julian J. M.
On 12/9/05, George Pajari [EMAIL PROTECTED] wrote:
Eric ManxPower Wieling wrote:
T/t/H/h and
What are you using to terminate the PSTN calls and do the SIP
transcoding?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jonathan
k. Creasy
Sent: Friday, December 09, 2005 8:45 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE:
On the CM, there is away to get the Device Information, Network Configuration, etc. by httping to the phones IP address. Is there away to do this via Astericks?
___
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To
If it's an electronic device, which this certainly is, and if it works on 100-240V, it will almost certainly work at either 50 or 60Hz. It probably gets converted to DC anyway but even if not, there wouldn't be much point in manufacturing a 100-240V power supply if it wouldn't work on both 50 and
Just an FYI... not all cell numbers are portable.
Do you have any more information on this? I read somewhere that
sometimes you can port a number to a VoIP provider but not be able to
port it back to the PSTN because not all PSTN providers will take
numbers from VoIP providers. Is
Ryan/Jonathan,
Couple quick questions regarding your setup?
Do you operate this in a strictly master/slave setup?
Do you have anything(mon/ha's internal status/monitor options) that
actually monitors the asterisk process (to determine if it is hung). Or
is it only with total box failure to you
Adam,
An Audicodes Mediant 2000 gateway with a couple of PRI's.
Why?
Doug.
-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 7:59 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial
How do I set up extensions.conf to wait for x rings (ringing all
extensions) before answering? I'm trying to mimic a regular answering
machine on an multiple analog phone system. Currently, Asterisk picks
up after 1 ring and just tries to dial one extension. I want all
extensions to ring.
Howdy - This is my first post on the list, and from what I've seen of * I'm
very impressed. I had a question regarding everybodys experience with Teliax
or Broadvoice. I setup a Teliax trunk this morning, and had calls going out
it in about 5 minutes(Had to get more coffee). Has anybody had any
Yes, that's a great question. I'm wondering the same thing. Can these heartbeat
apps monitor applications as well as network connectivity? The heartbeat
utility at www.linux-ha.org talks about monitoring some standard apps like web
servers and such but isn't clear about other apps... like
Wath brands of modem or chip's work with asterisk?
Intel 537EP
Ambient MD3200
Motorola 62802
and i mix the 3 types of modem in 1 pcpbx? or how i do to mannage tree phone
lines?
thnks in advance
Vladimir
__
Visita http://www.tutopia.com y comienza a navegar más
I have a Digium TDM11B, I'm fighting an issue with with echo on the PSTN side. I run [EMAIL PROTECTED] 2.0. I have an analog phone on the FXS channel 1 and Telco on the FXS channel 4. I also have a coupe of softphones, 1 iax2, the other sip, and a LinkSys Sipura 941. I use a VOIP provider for long
chan_capi registers fine: **
[chan_capi.so] = (Common ISDN API for Asterisk) == This box has 1 capi controller(s). == Reading config for BRI1 -- ast_capi_pvt BRI1-pseudo-D (MSN1,MSN2,capi-in,0,2) (1,4,128)
--
Just an FYI... not all cell numbers are portable.
Do you have any more information on this? I read somewhere that
sometimes you can port a number to a VoIP provider but not be able to
port it back to the PSTN because not all PSTN providers will take
numbers from VoIP providers.
Doug,
We currently are using Digium TE410P boards directly into each Asterisk
server. I've been researching various gateways, up to DS3 capacity, to
convert PRI to SIP and then allocate the SIP among multiple Asterisk
servers. I've looked at Cisco AS5400 (), Lucent APX 1000 ($$$), and
Howdy - This is my first post on the list, and from what I've seen of * I'm
very impressed. I had a question regarding everybodys experience with Teliax
or Broadvoice. I setup a Teliax trunk this morning, and had calls going out
it in about 5 minutes(Had to get more coffee). Has anybody had
Hi I use a allied telesyn at-vp730
Works quite well
ash
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: 09 December 2005 16:01
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial
I think the Audiocodes boxes run at about $19,000 each.
-Original Message-
From: Adam Robins [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 9:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Asterisk Dial Failover
Doug,
We
Can you please explain?
Whats CM?
Whats Astericks?
On 12/9/05, James Horn [EMAIL PROTECTED] wrote:
On the CM, there is away to get the Device Information, Network
Configuration, etc. by httping to the phones IP address. Is there away to do
this via Astericks?
Hello,
After a long period of inactivity we are proud to bring you a new
version of the Queue Statistics.
Main changes in this version are:
- Fixed a nasty bug where calls can't longer than 999 seconds.
- Added the possibility to see reports for all queues.
- some code cleanups
In the next
${CALLERID(type)}
In the CLI:
type show functions to get a list of functions
type show function callerid to get a list of types
In any case read /usr/src/asterisk/docs/README.variables
On 12/9/05, René Enskat [Teamware GmbH] [EMAIL PROTECTED] wrote:
Does the CIDNUM and CIDNAME is not any longer
I did cvs update -A, which brought in new files.
make clean
make
make install
make config
/etc/rc.d/init.d/zaptel restart
lsmod | grep ztdummy
Ztdummy is loaded.
/etc/rc.d/init.d/asterisk restart
lsmod | grep ztdummy
ztdummy 7816 0
wcfxo 17440 0
wcte11xp
WEll I personally have not implemented a Linux-HA cluster mainly because I
don't have the resources to do so. I study Asterisk purley as a hobby
(nerd.. yeahI know) because it is an awesome OSS product. Anyways, after
some searching around I think it would not be TOO difficult to implement a
we are beginning to test asterisk for our office
one of the features of the current phone system that is very heavily
used is overhead paging
now i came accross this post
http://forums.digium.com/viewtopic.php?t=2844highlight=features
that basically says it is not possible with asterisk.
Robert La Ferla wrote:
How do I set up extensions.conf to wait for x rings (ringing all
extensions) before answering? I'm trying to mimic a regular answering
machine on an multiple analog phone system. Currently, Asterisk picks
up after 1 ring and just tries to dial one extension. I want
Sorry, forgot to follow directions :-)
linux cvs status ztdummy.c
===
File: ztdummy.c Status: Up-to-date
Working revision:1.15
Repository revision: 1.15/usr/cvsroot/zaptel/ztdummy.c,v
Sticky Tag:
I use Teliax. I think the sound quality is really very good. I get
about an 80ms ping with them, but a 20ms ping to Junction Networks.
Some how calls to/form Teliax sound better.
With Junction Networks I get great customer service, with Teliax I get
Okay to good customer service (depending on who
On Fri, 9 Dec 2005, Tzafrir Cohen wrote:
On Fri, Dec 09, 2005 at 02:02:53PM +, Julian Lyndon-Smith wrote:
We've just seen that one of our servers is an hour out (it reckons that
it's 15:02 instead of 14:02).
Can I change the time when * is running ? I don't want to try just in
case it
Not sure I completely agree with all of these.
Looking at it objectively, Asterisk has many benefits over
traditional PBX systems, yet you should be aware of some of
the limitations.
Benefits:
1. Open source / low-cost of ownership / operates on cheap PC
hardware. You get voicemail,
Overhead paging is totally possible, there are several articles available on
how to do it. But you cannot have multiple zones today unless you use a sip
device that has autoanswer.
Easiet way to remove that message is to replace the file with one that only
has a split second of silence.
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Thanks. This will call/ring multiple extensions but what about waiting
for X rings before going to voicemail? How do I do that?
That article is about shared call appearance. I have this working using
Grandstream GXP-2000's. It's a great new feature.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stas Khromoy
Sent: Friday, December 09, 2005 8:21 AM
To:
Create a ring group, put all the extensions into the ring group, set your
dialplan to go to the ring group first and then failover to a voicemail
extension.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Derek Whitten
Sent: Friday, December 09,
Can you please explain?
Whats CM?
I think this is for (Cisco) Call Manager
Whats Astericks?
Maybe it's in the same part as Atérisk (only the french-speaking
will laugh this one)
;)
___
--Bandwidth and Colocation provided by Easynews.com --
He maybe referring to Cisco's Call Manager
-Original Message-
From: C F [mailto:[EMAIL PROTECTED]
Sent: Friday, December 09, 2005 11:09 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Phone Information
Can you please explain?
Whats CM?
Whats
Hi guys, given 1.2.1 is out. How is the t38/fax support going on?
also, can someone point me to proven brands/configs with ip fax capable
machines? Fax machines with a lan port (i heard of them but havent
found them online).
or a fax machine plugged to a converter that actually works for heavy
In article [EMAIL PROTECTED], Jim Duda [EMAIL PROTECTED] wrote:
I did cvs update -A, which brought in new files.
make clean
make
make install
make config
/etc/rc.d/init.d/zaptel restart
lsmod | grep ztdummy
Ztdummy is loaded.
/etc/rc.d/init.d/asterisk restart
lsmod | grep ztdummy
On Fri, 2005-12-09 at 11:41 -0500, Robert La Ferla wrote:
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Thanks. This will call/ring multiple extensions but what about waiting
for X rings
Hey ho,
About a week ago i uploaded a channel.c jitter buffer on mantis, this is
the patch many of you have been waiting for. It's supposed to be rock
stable (stability wise), but needs some more audio quality testing.
This channel.c jitter buffer implementation is a channel independent
If you need more rings, increase 25 (that is in seconds, more or less).
Regards.
El vie, 09-12-2005 a las 11:41 -0500, Robert La Ferla escribió:
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Dear Aaron,
You can check out www.didx.org US did's start at 10 cents a number per month. uk start 30 cents per month.
You get 2 free did's to test the didx services.
If you can PROVIDE us Japan did's that would be super cool.
Rehan
On 12/9/05, Aaron Anderson [EMAIL PROTECTED] wrote:
Gentelmen
Yes, I have other PRIs. problem is that there are
100 different fields to fill in on the M1, but only 20 on the zap/asterisk
side!
I was able to get this
going.Afewpoints:
1. I am using the Sangoma Card - works
great.
2. I have the M1
set for USR and the Asterisk set for NET
3. The
hehe, :-)
On 12/10/05, Lawrence Jovellanos [EMAIL PROTECTED] wrote:
He maybe referring to Cisco's Call Manager-Original Message-From: C F [mailto:
[EMAIL PROTECTED]]Sent: Friday, December 09, 2005 11:09 AMTo: Asterisk Users Mailing List - Non-Commercial DiscussionSubject: Re:
The other thing I'll say about my PBX is that there is no comparison between
my Nortel i2004 and any SIP phone I've seen. Yes, the cost is slightly
more, but for an instrument that I interact with constantly - there is no
SIP device to compare. I know there will be eventually, but not now!
CM is the Cisco Call Manager and Astericks is the Asterisk Software.
-- Forwarded message --From:C F [EMAIL PROTECTED]
To:Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.comDate:Fri, 9 Dec 2005 11:08:48 -0500Subject:Re: [Asterisk-Users] Phone
I've been using them for about 3 months and haven't experienced any
problems with them.
My only problem has been with my ISP. I get two calls going at the same
time and the ISP boggs down.
Regards,
Chris
- Original Message -
From: Rolf Brusletto [EMAIL PROTECTED]
To: Asterisk
On 12/9/05, Stas Khromoy [EMAIL PROTECTED] wrote:
we are beginning to test asterisk for our office
one of the features of the current phone system that is very heavily
used is overhead paging
Overhead paging can be done with asteirsk in anyway you want, you can
even do mutilple zones, all
On 12/9/05, Kerry Garrison [EMAIL PROTECTED] wrote:
Overhead paging is totally possible, there are several articles available on
how to do it. But you cannot have multiple zones today unless you use a sip
device that has autoanswer.
Why can mutilple zones not be done?, why
On Fri, 9 Dec 2005 09:46:39 -0600
Rich Adamson [EMAIL PROTECTED] wrote:
Just an FYI... not all cell numbers are portable.
Do you have any more information on this? I read somewhere that
sometimes you can port a number to a VoIP provider but not be
able to port it back to the
FWIW, we have replaced a Mitel 3300 with Asterisk - 170 users, mixed SIP/IAX
and cell (GSM gateway). The feature set that Asterisk brings to the table is
as good as or (more often) far better than the 3300 at a far, far cheaper
cost. I am doing stuff that my users quite frankly find amazing, and
Rich Adamson wrote:
Just an FYI... not all cell numbers are portable.
Do you have any more information on this? I read somewhere that
sometimes you can port a number to a VoIP provider but not be able to
port it back to the PSTN because not all PSTN providers will take
numbers from VoIP
I realize that it's a timeout but what's implicit in that is that
Asterisk can't detect # of rings just the amount of time spent ringing?
I have been looking at the reference manual on asteriskguru.com. They
say it's a timeout but they don't indicate the units. Is it
milliseconds,
Update:
I've determined that the problem is DTMF 9. I cannot get to
extension 6950 on the Nortel. The 9 is totally skipped. However, if
I dial 69501, I get connected to extension 6501. What is so special
about DTMF 9?
Thanks
On Dec 8, 2005, at 7:40 PM, Gaurav Naik wrote:
I'm
Tony,
I downloaded fresh versions of asterisk 1.2.1 and zaptel 1.2.1 from digium.
I now have USE_RTC in the zaptel files. I recompiled and installed both.
I updated /etc/rc.d/init.d/zaptel and removed the other modules as I don't
have any Digium cards in my system.
lsmod results show that
I'm trying to figure out why you changed the subject?
Anyhow, thirdlane makes something called asterisk PBX Manager. There
is also another tool but it doesn't work (AFAIK) over http, amongst
others that tool can:
* Show you all the contexts
* Show you all the extensions, and the DP that drive them
hehe
I just installed * with a T1 span to and Adit600 with 2fxs and 1fxo
The 8 fxo ports were for zone pageing
works great
should work with any fxo device and an existing page system
On Dec 9, 2005, at 11:34 AM, C F wrote:
Overhead paging is totally possible, there are several articles
Robert La Ferla wrote:
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Thanks. This will call/ring multiple extensions but what about waiting
for X rings before going to voicemail? How do I do
Robert La Ferla wrote:
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Thanks. This will call/ring multiple extensions but what about waiting
for X rings before going to voicemail? How do I do
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