alex,
thanks for the note... oh well, fun times ahead! :)
daveC
Alex Balashov wrote:
Dave,
On Fri, 13 Apr 2007, dave cantera said something to this effect:
alex,
I have been considering linux clustering for *... am not ready today
and expect it in about 6-9 months... was wondering if
On Fri, Apr 13, 2007 at 12:32:06AM -0400, dave cantera wrote:
I had a similar problem... I forget exactly how I resolved it because it
happened twice... here is the solution from memory.
the sequence of the zaptel, libpri, and asterisk is important. if you
compile zap before libpri, zap
On Thu, Apr 12, 2007 at 10:17:25PM -0700, Yuan LIU wrote:
From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 09:18:46 +0300
On Wed, Apr 11, 2007 at 08:09:16PM -0700, Yuan LIU wrote:
From: Sanjay Rajdev [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 01:29:51 +0530 (IST)
[good
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Alejandro,
SSH to your box like this:
ssh [EMAIL PROTECTED] -L 8080:127.0.0.1:8080 (in Putty it's something called
portforwarding)
This will bind 127.0.0.1:8080 from your asterisk-box to 127.0.0.1:8080
of your local box. So you can access the
On Thu, 12 Apr 2007, Greg Woods wrote:
Aside from this, I love my new asterisk system, and my wife has almost
gotten used to having to dial 9 to get out of the house :-)
Can't help you with your zaptel ssh issues - I use them both on my
systems without any issues at all.
But why force
Stephen Bosch wrote:
Bob Smither wrote:
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options with different
warning audio.
I have to measure somehow the audio file
Hello all,
i followed for the following step.
1)extract AsteriskWin32-0.60-Setup.exe into cygwin folder
2)extract asterisk-1.2.14 into /usr/src
3)applied the patch like that patch -p0 awin32-0.60.patch
(here it will ask YES or NO.Is it possible to give YES recursively?)
4)make
that time i
Few days back I installed Asterisk 1.4.2 with Zaptel 1.4.0.
All SIP accounts were working fine, today I tried to install
a fxs Sangoma A200 card and got the following error.
I believe you need to download the Sangoma drivers from their site.
___
From: Suity Zsolt [EMAIL PROTECTED]
Date: Fri, 13 Apr 2007 08:43:33 +0200
Stephen Bosch wrote:
Bob Smither wrote:
On Thu, 2007-04-12 at 13:28 +0200, Suity Zsolt wrote:
Hi,
I have to set call length to 3min, but before hangup have to warn
caller. There are many IVRmenu and submenu options
Hi There
I would like to find out if there are any asterisk compatible cards that
support CDMA 2000 1X EV-DO and Ruim cards.
Brian
-
=
The information contained in this electronic message and any
attachments are intended for specific
I've got a voicemailbox with one message store. When I try to read it,
I get the followiing error:
ast_openstream_full: File digits/1F does not exist in any format
Obviously, I can just clear out that mailbox, but is this a bug that I
should be reporting?
/Per Jessen, Zürich
Hello Garth,
* Garth van Sittert [EMAIL PROTECTED] [13-04-07 01:27]:
Has anyone managed to get Asterisk 1.2 faxes working reliably with
spandsp 0.0.3? I am running Asterisk 1.2.17 and spandsp 0.0.3pre28 with
a Digium b410p card. Everything compiled smoothly but only about 70% of
faxes
Hi,
I have created two extensions (156157) with voicemail enabled. When I receive
a call from outside, my IVR is responded. When user press 156, if he (156)
unable to answer the phone, the voice mail will be goes to 156 and 157 email
IDs. I mean, I want to send voice mail to multiple email
Hi,
I have configured the below things:
Extensions
Trunk
Outbound route
Inbound route
IVR
Ring group
If anybody call to my DID number, my IVR is responded. After that, if he press
1, then Ring group will be activated. All are working fine.
My Problem:
I want to
Hi Chandra,
One option is to set up a group on your mail server containing multiple
addresses and use the group name as the email address within Asterisk.
rgds,
Phil.
Crazy Boy
I am recording a conference created using MeetMe.
Is it possible to record different users on different channels in the
file? (like a stereo registration, with sx channel for one user and dx
for another)
--
/*/
nik600
https://sourceforge.net/projects/ccmanager
I love this thread, especially when it came to the chicken boner
part of the discussion - brings back NANAE with a smile - and I'm glad
no one found it off-topic, I think it's well worth talking about (the
suit, not the chicken boners) as this may have an effect on some of
what we do.
I hope to
You might try doing a database lookup, but you'll still have to enter
all 200 caller ids by hand. I think the database lookup would
probably be better than adding 200+ extra lines to your dial plan.
There's probably something in AEL that you could write that might be
more effecient.
On
Hi Dawson,
Thank you for your response. I hope this is the good solution as said by you.
Regards,
Chandra.
[EMAIL PROTECTED] wrote: Hi Chandra,
One option is to set up a group on your mail server containing multiple
addresses and use the group name as the email address within Asterisk.
You should be able to get the latest version from sourceforge.net too.
I'm pretty sure it's a sourceforge project.
Tim.
On 12 Apr 2007, at 20:34, Moises Silva wrote:
Hum, I know Stefan, he is an asterisk-java dev, but he is not online
right now, I will let him know ASAP. Thanks!
On 4/12/07,
OK Yuan,
What I wanted to know is if the extension I've created is right.
exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED])
Will my asterisk bridge a SIP phone that dialed 101 to the SIP user:
[EMAIL PROTECTED] Do I need some think more in order for it to work? Do
you have or know any
Ok I had a chance to test web-meetme 3.0.1 and I have few comments here
- the Makefile for CBmysql lacks procedure that verifies existence of
/var/lib/asterisk/sounds/conf-recordings directory where the conference
records should reside. I had to go through .php files to find out where
they are
C F wrote:
J, Sorry didn't see this email when I wrote the other one (gmail sorts
them on a LIFO order). I can agree with you on everything even with
the terrible pain of getting Polycoms up and running, but once it is
up dont you have less problems with them then with other phones? Isn't
the
Hi Folks,
I know the PAP2T-NA has a jitterbuffer however, it seems to be adaptive,
which is fine for most situations... however, is there some way I can
either:
A) Specify how long it waits before it starts to shrink?
B) Specifiy a fixed sized jitterbuffer?
Hi List...
Lets say I have been asked to do an Asterisk/Trixbox install
for an environment where 2 companies 'live' in the same...
building...
Both companies have 2 incoming Analogue BT Lines
and then have good old BT Phones plugged into them..
I have been asked to setup a VoIP system for one
The short answer is sure, one * box can handle multiple companies, each with
their individual personalities This will take some dial-plan coding,
which may be difficult in the trixbox environment. There are GUIs which
proclaim to specifically support this type of environment -- Thirdlane being
I will add one thing. Parking might be a little problematic out of
the box. If you don't have problems using a patch that is not in the
main branch, there is a valet parking patch that would handle this
without any problems.
On the other hand, if the companies do not have to be 100% separate,
The weird thing is that the phone actually works for now, but I want to
proactively fix anything that may go wrong (this phone _has_ to work until
Saturday)
A SIP debug gives me this:
---
Scheduling destruction of call '[EMAIL PROTECTED]'
in 15000 ms
hd-t3143cl*CLI sip
-- SIP read from
Nah, nothing of the sort. It's actually a phone using Dynamic IP (so I
didn't chose his IP) and that weird address seems like the NAT device's
address (since I have two phones at the same location.)
Now, how the NAT device over there ended up with this address is
problematic.
Mike
On Fri, 2007-04-13 at 07:26 -0500, Lacy Moore - Aspendora wrote:
I will add one thing. Parking might be a little problematic out of
the box. If you don't have problems using a patch that is not in the
main branch, there is a valet parking patch that would handle this
without any problems.
On Fri, 13 Apr 2007, Gavin Spurgeon wrote:
Hi List...
Lets say I have been asked to do an Asterisk/Trixbox install for an
environment where 2 companies 'live' in the same... building...
Both companies have 2 incoming Analogue BT Lines and then have good old
BT Phones plugged into them..
Here is what I had to change on the phone1.cfg file:
I had this value in my 1.6.7 file, put in there following suggestions from
the Wiki
(http://www.voip-info.org/wiki/index.php?page=Polycom+Soundpoint+IP+501) :
reg.1.server.1.expires=30
Now, this worked flawlessly with 1.6.7. But with 2.x,
On Thu, 2007-04-12 at 14:47 -0400, J. Oquendo wrote:
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud
You haven't even mentioned Linksys SPAs. Have you tested them?
ciao
Luca
In my asterisk, I have calls coming in on a Zap channel and going out SIP.
My problem is that when I spy on the SIP channel, I hear the calling parting
breaking in and out, and the called party sounds just fine (SIP). If I spy
on the Zap channel , I hear both sides just fine. I am spying from my
Whats the diffrence between sending and recieving. We have a box in our
providers building (a few racks over). They send the calls via g711u and it
works great. If he is in a good dc it can work for him.
- Original Message -
From: Lee Howard [EMAIL PROTECTED]
To: Asterisk Users
I am having the same issue with 1.2.17. Only certain toll free numbers do we
have issues with.
- Original Message -
From: ismir saljic
To: [EMAIL PROTECTED]
Sent: Thursday, April 12, 2007 5:42 PM
Subject: [asterisk-users] DTMF problem with inbound calls on Toll-Free number
Yes. But dont use the h6315.com. Its not a public mirror. Its my box and I
put it up there cause I was installing asterisk several times a day.
- Original Message -
From: Lee Jenkins [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
On 4/13/07, Mike [EMAIL PROTECTED] wrote:
reg.1.server.1.expires=30
My understanding is that the above causes the phone to re-register
with its server (asterisk) every 30 seconds. I would expect a
registration to be a heavy operation, so that does explain the high
CPU load, possibly leading to
Jose,
Look for chan_zap.so into channel folder in your asterisk installation
dir. If you only see chan_zap.c your module wasn't compiled. Try
recompile asterisk.
Run lsmod and see if zapata module is lodade, if not try modprobe zapata.
In asterisk cli try to load module direct: module load
Fax over G711u works.. but it's touchy... when our customers ask us if we
support it we say yes and no.. and then explain that they can use it.. and
fax away all they want, but if they have issues we don't support it.I've
personally dialed-up to our modem bank from home using G711unice
It would be exten = 101,1,Dial(SIP/[EMAIL PROTECTED])
If Asterisk does not find a [host.domain.com] entry in sip.conf it will
dial by hostname/IP address. Only settings in sip.conf [general]
section will be used. People do this with Asterisk all the time. You
are not trying to do anything
If your chan_zap module was compiled, try to load zaptel modules
using: modprobe zaptel.
In my case, i received an error and i need to use kernel in i686 and
kernel-devel in 586 and recompile zaptel modules.
--
Frederico Madeira
[EMAIL PROTECTED]
www.madeira.eng.br
2007/4/11, Sanjay Rajdev
True, but that being said 1.6.7 did re-register every 30 seconds with no
issues.
Did the phone loss in performance after the upgrade?
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kai-Uwe Jensen
Sent: Friday, April 13, 2007 10:01
To: Asterisk
The point about the politcal side of this is extremely valid. But it also
becomes a business opportunity to provide the hosted PBX as a service to
BOTH companies! Beyond that, some sort of written agreement between the two
companies, the one owning the box and the other as a client may be
I've encountered a similar problem with Cisco equipment. The Cisco proxy
was not replying to Asterisk with an ACK after * sent an OK.
Since version 1.2.14, * was changed so that not receiving an ACK to an OK is
considered a FATAL error.
The specific change that causes this problem is in
On Fri, 13 Apr 2007, Matt said something to this effect:
Fax over G711u works.. but it's touchy... when our customers ask us if we
support it we say yes and no.. and then explain that they can use it..
and fax away all they want, but if they have issues we don't support it.
I've personally
Luca Corti wrote:
On Thu, 2007-04-12 at 14:47 -0400, J. Oquendo wrote:
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud
You haven't even mentioned Linksys SPAs. Have you tested them?
ciao
Luca
We have
Hi all,
I am trying to install Vicidial in an existent FreePBX installation
(I'm using Xorcom packages for Debian Etch), but I didn't find any
documentation, I found only this guide [0], but is for trixbox only,
do you think it will work on FreePBX on Etch?
[0]
Very much so... we actually have a fax machine up here in our NOC running on
g711u attached to an ATA, works fine.
Yes! This is exactly my experience with it working for a telephony
service provider that provides wholesale VoIP platform and CPE for
customers and works primarily over the
Linksys SPAs work well with Asterisk
On 4/13/07, Luca Corti [EMAIL PROTECTED] wrote:
On Thu, 2007-04-12 at 14:47 -0400, J. Oquendo wrote:
1) Snom
2) none! (they're all pretty much the same to me)
3) none! (they all have their pros and cons)
4) Cisco
5) ASStra
6) Polycrud
You haven't
On Fri, 2007-04-13 at 17:46 +0200, map wrote:
Linksys SPAs work well with Asterisk
I know, I use them and besides some initial nasty bugs and occasional
quirks they are quite nice. I also think they are not so ugly.
ciao
Luca
___
--Bandwidth and
On Fri, 2007-04-13 at 07:46 +0200, Jose Limeres wrote:
when I try to make a call through the ZAP
channel get an error message about NO ZAP CHANNEL AVAILABLE. Ztcfg and
zttool show the card correctly installed.
When I tried to use the debug command ZAP SHOW, it was not present in
the CLI.
We are looking at using Asterisk as a call recording server for an Avaya
VoIP S8700 system in a multi-site VoIP Call Center. All calls will be
coming in to one location and sent out via VoIP to other call centers.
What kind of specs should we be looking at purchasing for our Asterisk
server
On 4/13/07, Diego Quintana Cruz [EMAIL PROTECTED] wrote:
Hi all,
I am trying to install Vicidial in an existent FreePBX installation
(I'm using Xorcom packages for Debian Etch), but I didn't find any
documentation, I found only this guide [0], but is for trixbox only,
do you think it will work
Matt wrote:
Hi Folks,
I know the PAP2T-NA has a jitterbuffer however, it seems to be
adaptive, which is fine for most situations... however, is there some
way I can either:
A) Specify how long it waits before it starts to shrink?
I'm not sure about the PAP2 but the Linksys version has
This is correct. We have several machines hooked up with way.
Alex Balashov wrote:
On Thu, 12 Apr 2007, Mike Lynchfield said something to this effect:
No you are being misled.. SER can NOT DO IAX, SER = SIP only
No, you are not being misled. In my most recent response, I was
referring
hi!
First of all i want to tell i have a dedicated server on layeredtech
with direct internet connection and i currently dont use iptables, so
this is not about network configuration =).
well so, i install asterisk-1.4.2 on my server, and next install
asterisk-gui from the digium repository.
On 11 de abr de 2007, at 21:07, James Harper wrote:
A dialplan of '(S0:s)' will get your phone to jump straight into
the
's' extension in asterisk as soon as someone picks it up. From
there you
can do something like:
It worked perfectly! Thanks!
Just remember that having Asterisk supply
Hi again, i enabled SIP debug on the server. and appears this
[Apr 13 11:49:57]
--- Transmitting (no NAT) to 192.168.0.100:19934 ---
192.168.0.100 is my local ip address, there is the topology of the network:
72.232.33.66 - asterisk server, 1gb/s direct internet access
232.32.76.11 - my home
On 4/13/07, J. Oquendo [EMAIL PROTECTED] wrote:
C F wrote:
J, Sorry didn't see this email when I wrote the other one (gmail sorts
them on a LIFO order). I can agree with you on everything even with
the terrible pain of getting Polycoms up and running, but once it is
up dont you have less
On Fri, 2007-04-13 at 09:36 +0200, Per Jessen wrote:
I've got a voicemailbox with one message store. When I try to read it,
I get the followiing error:
ast_openstream_full: File digits/1F does not exist in any format
Obviously, I can just clear out that mailbox, but is this a bug that I
On 4/13/07, Patrick [EMAIL PROTECTED] wrote:
Do you know where this patch can be found? My googling came up empty.
http://www.freeswitch.org/asterisk_stuff/
app_valetparking.c works on 1.4. You have to add it to the menuselect
file. There's also a version for 1.2. I'm using it with 1.4
Well, if the Linksys version has it.. PAP2 should as well. I'll have to
re-look at the options.
On 4/13/07, Andres [EMAIL PROTECTED] wrote:
Matt wrote:
Hi Folks,
I know the PAP2T-NA has a jitterbuffer however, it seems to be
adaptive, which is fine for most situations... however, is
Hi Manolet,
Can you provide your sip.conf?
Thanks!
-- Alex
--
Alex Balashov [EMAIL PROTECTED]
___
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Hello,
Are HPEC (ie zaptel 1.2.X) and bristuff compatible ?
And the results ?
Regards
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Which version of Zaptel and Asterisk are you using.
If you have compiled Asterisk 1.4.2 with Zaptel 1.4.0 or a lesser version of
Zaptel, you may face this problem.
Regards,
Sanjay Rajdev
- Original Message -
From: Greg Woods [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
I think that the best choice is the snom family...
We use all snom in ower office. We tried the Polycom but the support is
not so good.
Bruno.
C F wrote:
On 4/13/07, J. Oquendo [EMAIL PROTECTED] wrote:
C F wrote:
J, Sorry didn't see this email when I wrote the other one (gmail sorts
them
C F wrote:
I must say I have never run into a situation where I had low
bandwidth, I always make sure there is at least 768k up, with a less
than 150ms latency (not always have been able to meet the later, but
never more than 250ms), so can't realy comment on this one.
*Ducks the items thrown
I have some customers that come in via SIP on a T1 and then we drop
their calls off on a PRI. They can fax with about a 98% success rate
using an ATA-SIP over a T1-Ethernet- Asterisk - PRI-PSTN. As long
as the fax machine can negotiate 9600 baud it will work. We have a
customer doing this and
From: Ronaldo Zacarias Afonso [EMAIL PROTECTED]
Date: Fri, 13 Apr 2007 08:06:04 -0300
OK Yuan,
What I wanted to know is if the extension I've created is right.
exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED])
OK, the syntax is a bit off.
exten = 101,1,Dial(SIP/[EMAIL PROTECTED])
will send the
On Fri, Apr 13, 2007 at 07:10:42PM +0200, Olivier wrote:
Hello,
Are HPEC (ie zaptel 1.2.X) and bristuff compatible ?
And the results ?
Latest bristuff (0.3.0-PRE-1y-[de] and 0.4.0-test1): probably yes, as
they were built with a recent enough zaptel version, and the bristuff
zaptel patch does
What wrong with this:
[get-dnisinfo]
; sub-routine to get owner's password
exten = s,1,Verbose( == )
exten = s,n,MYSQL(Connect connid localhost root password dax)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\
dnislookup\ WHERE\ dnis=\'${IVR-Exten}\')
exten =
On 2/4/07, 李君 [EMAIL PROTECTED] wrote:
http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO ;
(...)
== Channel 'SIP/111-086497c8' jumping out of macro 'nway-start'
Feb 2 16:53:10 WARNING[4218]: res_features.c:1385 ast_bridge_call: Bridge
failed on channels SIP/112-08641920
of course, download it from here:
http://contelecltda.com/sip.conf
but i dont edit the sip.conf, is the default make samples sip.conf
file. i just use the asterisk gui interface to add the user...
2007/4/13, Alex Balashov [EMAIL PROTECTED]:
Hi Manolet,
Can you provide your sip.conf?
On Fri, 2007-04-13 at 22:42 +0530, Sanjay Rajdev wrote:
Which version of Zaptel and Asterisk are you using.
If you have compiled Asterisk 1.4.2 with Zaptel 1.4.0 or a lesser version of
Zaptel, you may face this problem.
It happened to me with asterisk 1.4.1 and Zaptel 1.4.0
--Greg
To follow up on the don't need to dial 9 to get out topic, in some
places, there are so few phone prefixes, you can simply match them
exactly. Here's for where I live:
exten = _747,1,Dial
exten = _966,1,Dial
exten = _738,1,Dial
exten = _752,1,Dial
exten = _1NXXNXX,1,Dial
Barton Fisher wrote:
What wrong with this:
[get-dnisinfo]
; sub-routine to get owner's password
exten = s,1,Verbose( == )
exten = s,n,MYSQL(Connect connid localhost root password dax)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ password\ FROM\
dnislookup\ WHERE\ dnis=\'${IVR-Exten}\')
On Fri, 13 Apr 2007, Barton Fisher said something to this effect:
What wrong with this:
Well... what is wrong with it? :-)
I'm not trying to be funny, but, what are the symptoms that it doesn't
work? Error output on Asterisk console? Logs? Anything you can provide
would be helpful.
I am getting the following compile error on centos 5. Any suggestions?
Jerry
CC [M] /usr/src/digium/zaptel-1.4.1/xpp/xbus-core.o
/usr/src/digium/zaptel-1.4.1/xpp/xbus-core.c: In function ‘debugfs_open’:
/usr/src/digium/zaptel-1.4.1/xpp/xbus-core.c:171: error: ‘struct inode’
has no member named
Hello.
I have a Digium TE110P in my server asterisk, have connected a PRI of the
PSTN. the incoming calls work correctly, but when attempt to make calls
outwards does not work and it leaves an error to me like the following one:
*-- Channel 0/1, span 1 got hangup request
Apr 11 22:43:45
On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:
of course, download it from here:
http://contelecltda.com/sip.conf
but i dont edit the sip.conf, is the default make samples sip.conf file.
i just use the asterisk gui interface to add the user...
Well, then my conjecture
I am looking to allow some users to login to a website and change where
their ext is forwarded to. any ideas? It can be very simple or I can
install a full package and then allow certain users certain access.
Thanks in advance
Jason
___
When I try to park a call using a snom 320 phone, the phone disconnects
before I hear the parking spot announced. Is there a way to avoid this?
I have tried the following:
Press button programmed as extension 700
Press Transfer, then button programmed as extension 700
Press Hold, then button
Product selection is not cut and dry. What are your business requirements?
So you need encryption? If so, what kind?
Do they need support for outbound proxies?
Are you going to use the same model for remote deployments?
Do you need WAP capabilities?
Do you need programmable speed dials?
Do you
On 21:12, Fri 13 Apr 07, Tzafrir Cohen wrote:
On Fri, Apr 13, 2007 at 07:10:42PM +0200, Olivier wrote:
Hello,
Are HPEC (ie zaptel 1.2.X) and bristuff compatible ?
And the results ?
Latest bristuff (0.3.0-PRE-1y-[de] and 0.4.0-test1): probably yes, as
they were built with a recent
My wife's name is Nanae... =)
The VoIP patent stuff is something that needs to be talked about more. VoIP
is really going to suffer in the years to come because of patents. Might
make a good topic for a whitepaper at a conference of speaking engagement.
Can anyone tell me what the capacity is of 2 E1's in minutes. Ie how many
minutes can 2 E1's take.
Steve
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I have a CISCO 7912 phone, the LED on the phone does not glow when there is new
voicemail, can we configure Asterisk to have the LED glow on new Voicemail.
Regards,
Sanjay Rajdev
___
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? I think you are confuse E1's are a volume of ports eg 30 channels
So therefore 60 simultaneous calls or 3600 minutes per hour (or x2 for
your question of 7200 minutes per hour).
Now if you are talking about voip it changes again, what codec are you
using which will determine traffic
592 centimeter
Forum wrote:
Can anyone tell me what the capacity is of 2 E1’s in minutes. Ie how
many minutes can 2 E1’s take.
Steve
___
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On Fri, 13 Apr 2007, Forum said something to this effect:
Can anyone tell me what the capacity is of 2 E1's in minutes. Ie how many
minutes can 2 E1's take.
Well, an E1 has 30 usable timeslots, so 60 DS0s * 1440 minutes/day * 30
day-month = 2.592 million minutes.
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Alex Balashov [EMAIL
On 4/13/07, Forum [EMAIL PROTECTED] wrote:
Can anyone tell me what the capacity is of 2 E1's in minutes. Ie how many
minutes can 2 E1's take.
42.
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mmm are you sure that asterisk-gui generate it on the sip.conf file?
cause i see a new file called users.conf, and i can see the sip users
on it. Anybody uses asterisk now and can check it please??
2007/4/13, Alex Balashov [EMAIL PROTECTED]:
On Fri, 13 Apr 2007, Manolet Gmail said something to
On 4/13/07, Sanjay Rajdev [EMAIL PROTECTED] wrote:
I have a CISCO 7912 phone, the LED on the phone does not glow when there is new
voicemail, can we configure Asterisk to have the LED glow on new Voicemail.
I'm not sure if there's anything specific you'll need to do on the
Cisco, but for me
Hi Sanjay,
This is easily fixed.
Check this bug report for how to fix it:
http://bugs.digium.com/view.php?id=8575
Thanks,
MG
On 13/04/07, Sanjay Rajdev [EMAIL PROTECTED] wrote:
I have a CISCO 7912 phone, the LED on the phone does not glow when there is new
voicemail, can we configure
On Fri, Apr 13, 2007 at 02:52:09PM -0500, Jason Walker wrote:
I am looking to allow some users to login to a website and change where
their ext is forwarded to. any ideas? It can be very simple or I can
install a full package and then allow certain users certain access.
Thanks in advance
Hi All,
Customer is requesting 1 incoming toll free #, that dial out to 4
different terminating numbers, not ring all at once but ring #1, then
#2, then #3, then #4, then back to #1 consecutively on inbound calls,
regardless if someone is on #1. So this is not like a hunt group,
more like an
/asterisk-users
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Hi,
We had an install working quite well of SpanDSP on our machine until
recently where it has began spitting out an error stating
unable to translate from unknown to unknown.
Any ideas ?
Regards,
Sahil Gupta
VoiceValley
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On Fri, 13 Apr 2007, Manolet Gmail said something to this effect:
mmm are you sure that asterisk-gui generate it on the sip.conf file?
cause i see a new file called users.conf, and i can see the sip users
on it. Anybody uses asterisk now and can check it please??
Hmm. I use 1.4.x here and
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