How can you store pauses in speed dials for the GXP-2000? I used
something like 8005551212,,,1,7890 to dial the toll free number, wait
6 seconds (I'm used to the commas being a 2 second delay), pressing
1, waiting 2 more seconds and then entering 7890. However, when I
press the speeddial
Your problem could be DTMF-related. Make sure that both your sip peer
has dtmfmode=rfc2833 and the GXP-2000 is configured for RFC2833 as well.
- Waldo
On Apr 28, 2006, at 10:20 AM, Johnny Stork wrote:
I seem to be having a problem with my GXP-2000. No matter how
carefully I type in the
Make sure that sip.conf has externip and localnet are properly
configured. I have many GXP-2000 on different nets as my * box with
no problem.
- Waldo
On Apr 28, 2006, at 11:12 AM, Mimmus wrote:
Hi,
I have a lot of GXP-2000 phones not registering with Asterisk server.
After two days of
Make sure dtmf-mode is set to rfc2833 in both sip.conf as well as in
the GXP-2000.
- Waldo
On Apr 27, 2006, at 12:28 PM, dataman wrote:
We are having trouble getting the GrandStream GXP-2000 (1.0.2.13)
to work with the Asterisk (1.2.6) voice mail prompts. We access
voice mail but
I have a bunch of UIP-200 phones working in different locations.
However, in one particular location the conversations sound very
choppy and my client is not tolerating it. Looking through the TFTP
configuration file, I see there are a bunch of parameters that could
adjust the jitter and
H,
One thing is what you type in extensions.conf and another is how
Asterisk sees the dial plan. In the CLI, do a show dialplan and look
for your entries. Asterisk may re-order them differently.
In theory, your regexp should NOT match your 7 digit number. It could
be a bug. Try
I seem to be having the same problems. Is anyone from trxtel reading
this? I guess you get what you pay for :)
- Waldo
On Apr 13, 2006, at 6:36 AM, Gustavo Hernandez wrote:
Hi !
Anybody know if 1800 free termination services from trxtel are in
troubles?
I can´t reach it, and don´t know
Hey Henri,
Long time no talk. How far have you been able to scale oreka up to?
How many simultaneous calls have you been able to record and under
what hardware config?
Thanks,
Waldo
On Apr 12, 2006, at 11:12 AM, Henri Herscher wrote:
Another solution would be to use a dedicated
Can anyone provide any further info on External IVR application? It
seems interesting. I currently have a heavily used AGI script that I
use for a custom IVR. It is written in Perl. I wonder if it would be
more efficient to migrate it to this External IVR. Will it be more
efficient? Will
AFAIK, it doesn't make much of a difference if all you are going to
be mainly using is the TE card. From what I've heard and seen, a
single P4 3GHz machine will handle a fully loaded TE4XX board with no
problem.
- Waldo
On Apr 11, 2006, at 10:30 PM, Tim Connolly wrote:
I was offered
Hi,
I have a few GXP-2000 working fine with Asterisk. The one thing I
have not been able to do is to program the MSG button to dial the
Voicemail extension. How can I program that button? I normally use
extension for voicemail. Can anyone shed any light?
Thanks,
Waldo
Right, but it's asking for a user id not a number to dial. So, how
would I set it to dial extension ?
Thanks,
Waldo
On Apr 9, 2006, at 12:21 PM, Harald Holzer wrote:
Look at the Account Settings for Voice Mail UserID.
Hi,
I have a few GXP-2000 working fine with Asterisk. The one
= 100,4,Macro(hangupcall)
so the user doesn't need to put in a password when they press the
MSG button
Waldo Rubinstein wrote:
Right, but it's asking for a user id not a number to dial. So, how
would I set it to dial extension ?
Thanks,
Waldo
On Apr 9, 2006, at 12:21 PM, Harald
AFAIK, you can use database lookups to fetch the internal caller id
and external caller id depending on the channel that is placing the
call. Then, simply set the corresponding caller id before placing the
call. Alternatively, which is what I currently do, since I don't use
account codes,
Is there any way to define call parking parameters for different
contexts?
For example, if I have a client in context 100 and another client in
context 200, can they both define parking positions, say, from
701-710, where 701 in context 100 is different from 701 in context 200?
Or even
)
SuperValetParking - Latest from BKW 26/11/2004:
http://www.asterlink.com/svp/
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Waldo Rubinstein
Sent: Thursday, April 06, 2006 9:41 AM
To: Non-Commercial Discussion Asterisk
Subject: [Asterisk-Users] Call Parking
Any opensource solution?Thanks,WaldoOn Mar 29, 2006, at 7:25 PM, Alyed Tzompa wrote: I use Portaone's PortaSIP for everything related to LCRAlyed Return-Path: [EMAIL PROTECTED] Wed Mar 29 16:48:54 2006Received: from digium-69-16-138-164.phx1.puregig.net [69.16.138.164] by
I'm wondering how you guys handle least cost routing within Asterisk.
Basically, we have a few providers with, obviously, different rates
per route. Additionally, we have a number of clients who have DIDs
assigned to them (either pointing to a single SIP peer or to more
complex dialing
Pardon the question, but what I understand of FreePBX is that it's
basically Asterisk with a web interface and some additional modules.
Is that correct? Can you install FreePBX on a system which ALREADY
has asterisk up and running or does it require ITS version of asterisk?
Thanks,
Waldo
Is anyone using this company? Can anyone comment on them? I started
testing their service yesterday, but today, they seem to be totally
out of service.
Thanks,
Waldo
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Asterisk-Users
Can anyone recommend a company that does professional Asterisk
recordings for things like IVR, greetings, MOH, announcements, etc?
Thanks,
Waldo
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To UNSUBSCRIBE or
Do you know when it's coming out? What will the price be?- WaldoOn Feb 22, 2006, at 1:18 AM, Cory Andrews wrote: Clint - Looks like your wish has been granted, and your love affair with Snom can continue. They are soon releasing the new Snom 300, which has most of the features your are fond of in
Ever since I migrated to 1.2.X, I've noticed a problem with the MOH
for any/all my queues.
When there is only one caller in the queue, the caller does NOT hear
the MOH. However, the moment more callers call into the queue, all
the OTHER callers will start hearing the MOH, but the first one
I'm running * 1.2.1 on Slackware.
I have several queues configured to record incoming calls once
answered (without joining the in and out files). Yesterday, I showed
my agents how to transfer a call received from a queue to another agent.
What I realized today is that when listening to
I have read contradicting information regarding whether or not to
have ACPI turned on when running Asterisk on kernel 2.6 with a zaptel
interface. Can anyone confirm what should be the correct setting for
ACPI for properly running this?
Thanks,
Waldo
I'm looking for a reliable 2 FXO-port gateway to connect a Panasonic
PBX to Asterisk. Can anyone recommend a stable and reliable one?
Thanks,
Waldo
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To
!
Thanks
Bjorn
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Waldo
Rubinstein
Sent: Tuesday, December 20, 2005 8:18 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Asterisk FXO Panasonic PBX
I'm looking for a reliable
I've found that I have to disable quality on the UIP200 when I
switched to Asterisk 1.2.X. It worked find with 1.0.9 and under.
Which version of Asterisk are you using?
- Waldo
On Dec 19, 2005, at 10:34 PM, Steven Job wrote:
Having the strangest time getting the uip200 to work with
I meant qualify not quality :)
- Waldo
On Dec 19, 2005, at 11:02 PM, Waldo Rubinstein wrote:
I've found that I have to disable quality on the UIP200 when I
switched to Asterisk 1.2.X. It worked find with 1.0.9 and under.
Which version of Asterisk are you using?
- Waldo
On Dec 19, 2005
the extension of the phone.
-Steve
- Original Message - From: Waldo Rubinstein
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, December 19, 2005 11:04 PM
Subject: Re: [Asterisk-Users] Asterisk with Uniden uip200
Sorry I can't help you further. Maybe someone else can chip in.
- Waldo
On Dec 20, 2005, at 1:27 AM, Steven Job wrote:
I have:
1) nat=route
2) dtmfmode=inband
Tried that and no luck. :-(
Yes, I have local and remote (behind NAT) UIP200.
You also need to make sure to specify in the
Is there a way to optionally keep asterisk in the media path in order
to record calls using the Monitor command? For example, if I have a
SIP peer that is defined with canreinvite=yes, this means that if
possible, Asterisk will not be in the media path. However, what
happens if the user
If that's the case, is it possible to override the canreinvite
attribute of a SIP peer in extensions.conf before a call is made or
answered by that peer?
- Waldo
On Dec 8, 2005, at 9:15 AM, Steve Totaro wrote:
Is there a way to optionally keep asterisk in the media path in order
to
I understand. But because the majority of calls are not to be
recorded, I don't have a need to keep Asterisk in the media path all
the time. That's why I'm wondering if you could dynamically keep it
in the media path or not.
- Waldo
On Dec 8, 2005, at 10:26 AM, Steve Totaro wrote:
Well,
This and Time Bandit's comment makes sense. I didn't realize that
these options in the Dial string will force Asterisk to stay in the
media path even if canreinvite=yes.
I'll give it a try.
Thanks,
Waldo
On Dec 8, 2005, at 11:18 AM, Steve Totaro wrote:
There may be a better way but off
Has anyone confirmed this? It sounds like an interesting theory.
- Waldo
On Dec 8, 2005, at 12:46 PM, Philipp von Klitzing wrote:
Hi!
This and Time Bandit's comment makes sense. I didn't realize that
these options in the Dial string will force Asterisk to stay in the
media path even if
SEPARATOR ***On 12/6/2005 at 9:22 PM Alvaro Parres wrote: Why using SIP instead of IAX2 ??? Only a question becouse i always use IAX On 12/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Well... not so perfectly.What I'm experiencing is that during certain call
Buy the O'Reilly Asterisk book. It describes them in one of the
apendixes.
- Waldo
On Dec 7, 2005, at 5:58 AM, Eugene Prokopiev wrote:
Hi,
Where can I find Asterisk modules description? For example, I need
to know what is app_zapateller, app_zapbarge, app_zapscan and chan_zap
--
I'm looking for a provider that can offer VoIP origination and
termination in the domestic US and Puerto Rico. To be more exact,
Toll Free numbers origination is a must. I'm looking for a block of
100 domestic toll free numbers and 100 local DIDs. Estimated traffic
is about 100K
on authentication for INVITE to
'5095551212 sip:[EMAIL PROTECTED];tag=as3e387d65'
and the caller gets busy signal. However, other callers calling the
same number get thru with no problems. Why is this?
Thanks,
Waldo
On Dec 5, 2005, at 10:30 AM, Waldo Rubinstein wrote:
This worked perfectly.
Thanks
t 10:22 PM, Alvaro Parres wrote:Why using SIP instead of IAX2 ??? Only a question becouse i always use IAX On 12/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote: Well... not so perfectly.What I'm experiencing is that during certain call volumes, many calls go thru from box1 to box
})
__
2nd Machine sip.conf
[box1]
username=box2
type=friend
host=10.0.0.1
secret=*
in extensions.conf
exten = _X,1,Dial(SIP/box1/${EXTEN})
--xce
*** REPLY SEPARATOR ***
On 12/5/2005 at 12:11 AM Waldo Rubinstein wrote:
I have 2 Asterisk servers running 1.2.0
username= did it.
Thanks,
Waldo
On Dec 5, 2005, at 2:14 AM, Luki wrote:
Any ideas on how to correctly set this up?
Try adding authuser= and/or username= to the configuration. Do a SIP
DEBUG and see what peer asterisk looks for when trying to authenticate
the INVITE. It probably can't find
I have 2 Asterisk servers running 1.2.0. One of them is a PSTN
gateway. Currently they are connected using IAX2. I wanted to play
with SIP.
I setup a sip entry (type=friend) in the PSTN gateway box and a sip
entry (type=user) in the second box in order to send calls using SIP
to the
I have similar problems with call drops.I don't know if "Shadow Ping" is some kind of pinging software. I have run a lot of flood pinging and everything comes back just fine. I don't have Cisco phones, I use Softphones and it's the only application running on the PCs (aside from MS Windows and
Or put everyone in a Meetme room and record the conversation in the
meetme room -- just an idea.
- Waldo
On Dec 1, 2005, at 2:00 PM, Dave Walker wrote:
Use sox to make a quadriphonic (4 channels) audio file. Any more
than 4 in a call would be silly ;-)
Innocent Evil wrote:
What you
Is there a way to monitor zaptel errors with something like Nagios?
I have a TE405P and seldomly I see messages like this:
Zaptel: Master changed to TE4/0/1
wct4xxp: Setting yellow alarm on span 4
wct4xxp: Clearing yellow alarm on span 4
which means that somehow the T1 went down and came back
Look at http://www.asternic.org/
- Waldo
On Nov 29, 2005, at 10:40 PM, Hiu Yen Onn wrote:
Hi all,
I am one of the client of the SIP/Asterisk, connected via Xlite
client. How should i know the rest of the active SIP users? Are
there any graphical tools giving a list of the active sip
I haven't tried any other one since FOP does what I need.
- Waldo
On Nov 30, 2005, at 12:37 AM, Hiu Yen Onn wrote:
other than asternic.org???
do u have any others alternatives
Waldo Rubinstein wrote:
Look at http://www.asternic.org/
- Waldo
On Nov 29, 2005, at 10:40 PM, Hiu Yen Onn
I apologize for the resend. I haven't received much feedback from this.
I also noticed that what I'm getting is the caller id as the caller
name and the sip peer name as the caller id number.
Does anyone have any ideas/suggestions?
Thanks,
Waldo
On Nov 26, 2005, at 2:52 AM, Waldo
I think I discovered a bug.
I have a dual Xeon machine running * 1.2.0
I have a queue defined to play the default music on hold class, which
simply plays an mp3 file.
When a call comes into the queue (note that there are no agents
logged in, but I have joinempty=yes and leavewhenempty=no
Hi guys,
I'm trying to forward a call from one * server to another using SIP.
Everything works when I use fromuser in the sip entry of the *
forwarding the call. The problem is that when the receiving * sends
the call to the UA, it puts the caller to be the value of fromuser
instead of
I've asked the same question in several occasions in the past and
never received a response. I figured this project was dead and stop
pursuing using it.
- Waldo
On Nov 21, 2005, at 12:17 PM, Lenz wrote:
Well, this is interesting - is anybody actually using app_icd out
there? :-)
l.
I have followed the recommendations. After some further tweaking, this is the most I've been able to get:[EMAIL PROTECTED] zaptel-1.2.0]# ./zttest -vOpened pseudo zap interface, measuring accuracy...8192 samples in 8190 sample intervals 99.975586%8192 samples in 8191 sample intervals
I had the exact same dilemma and switched to using AddQueueMember/
RemoveQueueMember instead of using agents. This solved my problem.
- Waldo
On Nov 17, 2005, at 7:13 PM, snacktime wrote:
I'd like some feedback on my solution so far for using queues in a
multi tenant configuration. For
Hello guys,
I've been having a a recurring problem with people complaining about
calls being dropped.
I have 3 asterisk servers:
Gateway: running Asterisk 1.2rc2 with TE410P connected to 4 T1s of
the PSTN
Server 1: running Asterisk 1.2rc2 with ztdummy using Gateway to
access the PSTN
watched the output of zttest to make sure your interrupts
are firing adequately?
Waldo Rubinstein wrote:
Hello guys,
I've been having a a recurring problem with people complaining
about calls being dropped.
I have 3 asterisk servers:
Gateway: running Asterisk 1.2rc2 with TE410P connected to 4
adequately?
Waldo Rubinstein wrote:
Hello guys,
I've been having a a recurring problem with people complaining
about calls being dropped.
I have 3 asterisk servers:
Gateway: running Asterisk 1.2rc2 with TE410P connected to 4 T1s
of the PSTN
Server 1: running Asterisk 1.2rc2 with ztdummy using Gateway
cable.
Regards,
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Waldo
Rubinstein
Sent: Tuesday, November 15, 2005 1:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Problem with call drops
Hello guys,
I've
Is there a way to have a separate MOH/Media server for playing music
and/or audio prompts/files?
I have an * box where calls come in and sit in a queue until an agent
is available. I noticed that at the end of the day, I end up with a
bunch of zombie mpg123 processes for calls that were
I had the same problem when I upgraded and fixed it by using the
following syntax:
AddQueueMember({queue_name}|{channel})
So, before when I had:
AddQueueMember(Ventas),
Now, I need to have:
AddQueueMember(Ventas|SIP/1234).
Because I don't know of a function that will just give me the
I upgraded one of our gateways connected to the PSTN with a TE410P to
1.2rc1.
What we are noticing is that, with the exact same configuration files
in /etc/zaptel.conf and /etc/asterisk/* from 1.0.9, we are starting
to receive a significant amount of calls where one of the digits in
the
Hi guys,
I have a question about the timing source parameter in zaptel.conf.
I have 4 T1s coming into a TE410P.
One T1 is with one carrier, who provides timing signal.
The other 3 T1s are from a different carrier, all sharing the same
timing signal.
Based on this, I have in
Bockman wrote:
Waldo Rubinstein wrote:
One T1 is with one carrier, who provides timing signal.
The other 3 T1s are from a different carrier, all sharing the
same timing signal.
Based on this, I have in /etc/zaptel.conf something like:
span=1,1,0,esf,b8zs
em=1-24
span=2,1,0,esf,b8zs
em=25-48
span
if source 1 is down, and so on..
Bart
- Original Message - From: Waldo Rubinstein
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 09, 2005 9:12 AM
Subject: [Asterisk-Users] Zaptel T1 Timing Source
Hi
mpg123? Could there be
something else going on? Any advice?
Thanks,
Waldo
On Nov 8, 2005, at 4:18 PM, Waldo Rubinstein wrote:
I don't know if there could be a memory leak or something in 1.2b2,
but I noticed that my box running 1.2b2 eats through memory like
crazy.
I'm running 1.0.9
Thanks. Got it.
- Waldo
On Nov 9, 2005, at 1:13 PM, Don Pobanz wrote:
Waldo Rubinstein wrote:
span=1,1,0,esf,b8zs
em=1-24
span=2,1,0,esf,b8zs
em=25-48
span=3,2,0,esf,b8zs
em=49-72
span=4,2,0,esf,b8zs
em=73-96
You are misunderstanding the span provisioning. There is ONE clock
Furthermore, I noticed that at one point, I have 11 active channels
in * in box B and I have 79 mpg123 processes running. It looks like
after a call is off hold or hung up, the mpg123 process is not
terminated. Does this make sense?
- Waldo
On Nov 9, 2005, at 1:10 PM, Waldo Rubinstein
.
If there is two Telco connected T1, select it as source 2
Now if timing source 1 goes down, timing source 2 will take over.
Bart
- Original Message - From: Waldo Rubinstein
[EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]
On Behalf Of Waldo Rubinstein
Sent: Tuesday, 8 November 2005 11:32
Wasn't aware of it, but if quality is good, it makes sense
since all I'm archiving is speech.
Will evaluate further.
Thanks,
Waldo
On Nov 7, 2005, at 7:14
wrote:
Check out the new app_mixmonitor app with 1.2b2. It produces one file
that is mixed already.
On 11/8/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Hilton,
AFAIK, you can optionally record in gsm. However, I think * won't do
it natively. It will do -in and -out wav files, soxmix them
protections by
using OGG instead of MP3. As far as compression goes, I've found the
difference between the two of them to be negligible. I've always used
OGG when possible to stay IP safe.
On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm trying to archive out call recordings and would appreciate
.
On 11/8/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm using it for originating calls but the problem I have is that
most of the recordings I have are from automatically recorded from
the Queue command (in queues.conf), so I don't know if you can tell
in queues.conf to use MixMonitor.
Thanks
I'm trying to archive out call recordings and would appreciate some
feedback as to which audio compression is more recommended MP3 or
OGG. In the past, I've use lame to convert to MP3, but I noticed the
audio volume drops significantly. Is it just a setting on the command
line of lame or
I don't know if there could be a memory leak or something in 1.2b2,
but I noticed that my box running 1.2b2 eats through memory like crazy.
I'm running 1.0.9 on a 1.5GB RAM machine. After 8 hours from a clean
reboot, the machine is using about 900MB of RAM.
On a 1.2b2 with 2GB RAM, after
no doubt that if you tune
complexity,
quality and bitrate parameters you will be able to get that filesize
down even further. Can't see any reason at all why you shouldn't be
able
to whack mp3 for filesize.
Cheers,
Mark.
-Original Message-
From: Waldo Rubinstein [mailto:[EMAIL PROTECTED
=ulaw
allow=gsm
canreinvite=no
dtmfmode=rfc2833
language=en
[100074]
type = friend
secret = mysecret
qualify = yes
nat = never
host = dynamic
callerid = Waldo Rubinstein 211
context = test-context
mailbox = [EMAIL PROTECTED]
The phone is at IP 10.0.10.236, so it's within the localnet.
Thanks,
Waldo
That would be great
- Waldo
On Nov 7, 2005, at 7:36 AM, Warren Burstein wrote:
Tim Litwiller wrote:
Well, I'd like them to drop in my voicemail when done recording -
maybe in a separate recordings folder but I'd like to use the
same interface to play them back.
I would like that, too.
: [EMAIL PROTECTED]
VM Extension : asterisk
LastMsgsSent : 0
Call limit : 0
Dynamic : Yes
Callerid : Waldo Rubinstein 211
Expire : 11077
Insecure : no
Nat : No
ACL : No
CanReinvite : No
PromiscRedir : No
User=Phone : No
Trust
Ok. That fixed it, but why? It works just fine in 1.0.9 and 1.2b1.
Very strange.
Anyway, thanks.
- Waldo
On Nov 7, 2005, at 10:57 AM, C F wrote:
The unreachable is the problem. Try adding a qualify=no to that sip
entry.
On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Additionally
that somewhere in your settings you have a qualify on, or that
1.2 has it on by default. Do the following:
cd /etc/asterisk
grep .*qualify.* ./*
and see the output, if the only line that has qualify is that
qualify=no, then this looks like a bug to me. Please report back.
On 11/7/05, Waldo Rubinstein [EMAIL
IP safe.
On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm trying to archive out call recordings and would appreciate some
feedback as to which audio compression is more recommended MP3 or
OGG. In the past, I've use lame to convert to MP3, but I noticed the
audio volume drops
with 1.2b2 or not.
1. Is the UIP200 on the same subnet as asterisk?
2. if not, is the UIP200 or asterisk natted?
In the meantime I will try to see on my 1.0.9 install if it works or
not with UIP200 phones.
Thank You.
On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I do have qualify=yes pretty
safe.
On 11/7/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I'm trying to archive out call recordings and would appreciate some
feedback as to which audio compression is more recommended MP3 or
OGG. In the past, I've use lame to convert to MP3, but I noticed the
audio volume drops significantly
are getting when calling that
extension?
On 11/6/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Nope. It isn't active. I even factory reseted the phone but still the
same. One more piece of information: it works just fine in 1.2b1. I
beginning to think it could be a bug in 1.2b2.
Any other ideas
I'm interested.
Thanks,
Waldo
On Nov 5, 2005, at 2:54 PM, Anthony Rodgers wrote:
And, I couple of times now I have offered to post a BBEdit language
module to the wiki, but have no idea where to put it.
Last chance for anyone who's interested...
Regards,
--
Anthony Rodgers
Business
(Do Not Disturb) button is not active on the
UIP200?
On 11/4/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro
phones.
All phones register fine with * and I can place outbound calls with
no problem.
I can call from the X-Pro to any
I am running * 1.2b2 with some UIP200 phones and a bunch of X-Pro
phones.
All phones register fine with * and I can place outbound calls with
no problem.
I can call from the X-Pro to any other X-Pro. I can call from UIP200
to any other X-Pro. However, the UIP200 cannot receive calls.
I understand. Are there or is there any other queueing application
for Asterisk that is more efficient than the out of the box Queue
application?
Thanks,
Waldo
On Nov 2, 2005, at 8:32 PM, Kevin P. Fleming wrote:
Waldo Rubinstein wrote:
Is this a feature/problem because I use
I suppose the * and SER topic has been discussed way too much, but I
searching through all the archives, I haven't really found an answer
to what I think could be done.
I would like to setup a set of asterisk servers with identical
configuration files so that a SER machine can load balance
I have been trying to find more information on the One Touch Record
feature in 1.2 (features.conf) but have not been very successful.
Basically, I've been trying to get more information as to:
1) Do I need to specify any particular option in the Dial command
2) How can I customize the location
Hi all,
I'm using * 1.2b2, however, what I'm about to describe also happens
to me in 1.0.7 and 1.0.9.
I use AddQueueMember to add stations to my queues and I don't have
any agents defined in agents.conf.
What I noticed is that whenever my queue strategy is something other
than ringall,
Hi guys. Please disregard this. I'm testing connectivity after being
down due to Hurricane Wilma.
Thanks,
Waldo
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Hi guys,
Just a quick question. I need to buy a dual T1 card and I'm debating
between TE210P or the Sangoma A102u. Any recommendations?
Thanks,
Waldo
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and I
hope I've given you enough information to answer your question for
yourself.
MATT---
On 10/23/05, Waldo Rubinstein [EMAIL PROTECTED] wrote:
Hi guys,
Just a quick question. I need to buy a dual T1 card and I'm debating
between TE210P or the Sangoma A102u. Any recommendations?
Thanks
2005 17:41:31 +0200, Waldo Rubinstein
[EMAIL PROTECTED] wrote:
I have played with AddQueueMember and it works great. However, there
is one problem that I have and I hope someone can point me in the
right direction.
My client's agents rotate seats. This means that if I want to track
calls
actually received or placed a call regardless of which extension he/she may be sitting on?Thanks,WaldoOn Oct 10, 2005, at 12:22 PM, Waldo Rubinstein wrote:BJ,Thanks for the prompt response. Both my clients work by using the AgentCallBackLogin so that * can send queued calls to them regardless of which
You mean to say that it will ONLY log if I have an h extension or if
I don't? Shouldn't it be logged no matter what?
- Waldo
On Oct 11, 2005, at 5:31 AM, Simone Cittadini wrote:
Dinesh Nair ha scritto:
On 10/10/05 22:30 Waldo Rubinstein said the following:
1) When are asterisk CDR logs
Thanks.I understand your POV. However, in addition to usage-based billing (which is what you refer to), I need to bill for the account. So, if the user placed two simultaneous calls with the same account, that may be fine because it could have been the call-waiting feature. However, if the user
Hi list,
I have a couple of questions related to asterisk billing and the
generation of cdr logs. I've searched the wiki but have not found my
answers, hopefully you guys can help.
1) When are asterisk CDR logs _normally_ generated? When the call
arrives, when the call hangs up, or both?
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