)
(TX)
Rx: 0 (0) Tx: 0 (0)
Ronald Hartmann
Director Technical Services
VerCom Systems, Inc.
410 Fame Rd, Dayton, OH 45449
Voice:866.VerCom.4 Fax: 866.422.6486
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Anyone know if it is possible to control how aggressively the
Aggressive mode behaves.
Meaning, is it possible to dial back the aggressive mode to have a happy
medium between
Regular and the Aggressive defaults.
I have a situation where Normal echo cancellation is not quite enough,
however when
Anyone have any information regarding the ectoolkit on svn?
~ron
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Anyone who would be so kind to send me a copy of app_changrab.c would
absolutemy make my day.
Thanks in advanced, as I have googled and googled and no go.
~ron
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Asterisk-Users mailing list
To
Good Day list,
I have read wiki pages I have googled to death and am getting no
closer to understanding the methodology of onhold music.
Maybe I am trying to do something that is just not possible:
Here is my desire.
1) Call comes in to the asterisk box via Zap
Good Day list,
I am having a bit of an issue as it relates to the musiconhold
settings in the Version 1.09 of Asterisk
Problem I am unable to set different music classes for different
extensions.
1) (default) I would like to be able to set generic music on hold for
the company,
I have been getting quite a bit of PRI Resets using my Quad PRI Digium
card.
Prior to the resets I am getting similar notices to the following
chan_zap.c:7482 pri_dchannel: PRI got event: HDLC Bad FCS (8) on Primary
D-channel of span 3
Telco claims the PRI's are fine on their end and that it
Are there any switchvox/fonality type Asterisk based PBXs where I can
buy just the software? I don't want to buy their 'bundles' that come
with junky PC hardware. I just want their software/GUI to run on my
hardware.
Have a look at the AMP project
http://sourceforge.net/projects/amportal
Anyone know if the INTEL/Dialogic announcement will become available to
us who do not use the asterisk BE?
Just curious as I have used those cards in the past and they are VERY
STABLE and VERY dependable with excellent quality.
~ron
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Good Day all,
I have a box connected to a netgear switch which allows me to
set priority based upon DSCP Values. This switch has listings from
value 1 - 63. And can be set to normal, high, etc. Does anyone know
what or how to translate TOS= line in the sip.conf file to in order to
have
Good Day list,
Does anyone know if it is possible to setup asterisk such that
it passes DTMF Tones through from One channel to the next transparently.
I have a situation where asterisk is answering the phone on
Channel 1 (first channel of a PRI) and then bridges this call to
Good Day list,
Does
anyone know if it is possible to setup asterisk such that it passes DTMF Tones
through from One channel to the next transparently.
I
have a situation where asterisk is answering the phone on Channel 1 (first
channel of a PRI) and then bridges this call to
Good Day All,
I am experiencing some weirdness using the EM channel and hope
you can offer a little assistance with the problem I am having.
1) call comes into channel 25 (Second Span first channel of a Digium
Quad PRI from SBC-PRI)
2) Call is sent to channel 1 (First Span first channel
Good day,
I am hoping that someone can assist me with a work around.
I have an IVR system that I am attempting to connect asterisk
to, however the IVR was written
Some time ago, and requires tones to be approximately 2 seconds
in length. I am using SIP
Phones,
Good day list,
I have been fighting echo problems on my PRI card.
Everything is working great outbound, however inbound calls have
echo.
I have found the issue but need help in fixing it.
During outbound calling zap show channel 18 shows that the
call is
Stable Version 1.0.7
-Original Message-
From: Andrew Kohlsmith [mailto:[EMAIL PROTECTED]
Sent: Friday, May 27, 2005 11:36 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] PRI Actual-HookState not showing offhook
oninbound
On May 27, 2005 10:57 am, Ronald Hartmann
experiences into wiki over the holiday weekend.
-Original Message-
From: Ronald Hartmann [mailto:[EMAIL PROTECTED]
Sent: Friday, May 27, 2005 3:03 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] PRI Actual-HookState not showing
Good Day all,
I have a Fractional PRI connected to my Asterisk Box via a T100P
card.
When I initiate a call out to phone number 123- the call
sounds great no echo what so ever.
If the person at 123- hangs up and calls me right back (same
handset on both sides)
M (Default) (Slaves: 19)
Channel 20: E M (Default) (Slaves: 20)
Channel 21: E M (Default) (Slaves: 21)
Channel 22: E M (Default) (Slaves: 22)
Channel 23: E M (Default) (Slaves: 23)
Channel 24: E M (Default) (Slaves: 24)
24 channels configured.
Ronald Hartmann
Director Technical
Anyone using this device?
I would love to see some Zapata/zaptel/extensions.conf configs that you
have used to get it to work with asterisk.
Also Any input for setting up the 624 would be great.
Thanks
~ron
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Good Day,
I am curious if anyone knows if there is a way to park a call
into a specific parking spot with asterisk 1.0X.
I think (based on what I read) that app_valetparking.c could
perform this function, however I am unclear as to
If this app is supported in the 1.0X
Good Day list,
It appears that the CDR is inaccurate, (or I am inaccurate when
reading it) when an attended transfer is conducted with a phones
transfer button
Example
+-+++---
Looking for advice on the following feasibility
--PRI (Goes to span 1)--Asterisk (4 span PRI
Card)-- Sip Phones
(Receive ANI and DNIS)
|
|
|
T1 Robbed Bit
Span 2
|
Good Day list,
My head is pounding from google overload.
Does anyone know if there is any way to park a call and specify
the (context, Extension, and Priority) of where to call back to if
the call times out and is parked too long?
ParkAndAnnounce does this feature,
Good Day list,
Need assistance determining the best place to read up on whether
Asterisk can help me out.
I have a situation where I need to do the following
PRI from Telco ---
Analog Channel BankProprietary Box
Good Day List,
I am looking to see if anyone is willing to share their working
configs with me.
I would be happy to add to wiki and document steps to get it to
work with asterisk.
I am looking for both Welgate configs as well as sip.conf and
extension.conf snippets.
Good Day list,
I am having
some issues with my card in that it wants to share IRQs
with everything else in my box.
I am
running WhiteBox Linux 3.0
Is there a
way to tell linux to assign a specific IRQ to a card. (unfortunately my MB does not
have feature of assigning IRQ to
Please do share I am very interested in your mods and the web interface.
-Original Message-
From: Dan Austin [mailto:[EMAIL PROTECTED]
Sent: Monday, February 28, 2005 7:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Advanced Conferencing
Any chance you can share your presentation slides, or handouts etc.
thanks
-Original Message-
From: David Uzzell [mailto:[EMAIL PROTECTED]
Sent: Friday, February 25, 2005 6:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] VoIP/Asterisk
presentation
On Sat, February 26, 2005 1:36, Ronald Hartmann said:
Any chance you can share your presentation slides, or handouts etc.
Sure, but was only slides, no hand outs...
http://www.asterisk.net.au/voip%20in%203%20beers.pdf
http://www.asterisk.net.au/voip%20in%203%20beers.sxi
http
I am having a problem with the Call Queue Feature.
If I let a user into the Queue prior to an agent being available for
them to take the call, they experience the following:
1) they hear that they are the first in line
2) when the agent finally logs in (the caller on hold in the
Good Day list,
Does anyone know if there is a way to get GotoIfTime to accept
individual weekdays instead of a range?
Example Dr. Office is closed on Thursday and Sunday.
Ron
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Good day list,
I am feeling extra stupid this Monday morning and am hoping
someone can come to the rescue.
I am trying to use the ztmonitor utility on my wildfire 4 FXO
card. and have read the following from the wiki.
*Wiki start
If you set this to yes, use
Sorry issue solved.
I had to RTFM better I just needed to increase the gain
higher my magic number ended up being 15.5
Sorry to bug 8000 ppl.
~ron
-Original Message-
From: Ronald Hartmann [mailto:[EMAIL PROTECTED]
Sent: Monday, February 14, 2005 11:18 AM
To: asterisk
Good Day list,
I have worn out my google toolbar today looking for a way to
determine which group an incoming call belong to, but have not been very
fruitfull in my endeavors.
I am trying to figure a way to determine which group the
incoming call that I just answered is part of
Good Day List,
I have several different SIP Phones running on my asterisk box.
I have recently purchased a snom 190 and a snom 220.
They work great, EXCEPT I have to have the volume turned all the
way up in order to hear the conversation on the other end.
All the
Help!
I have 2 computers and one works fine I can play the messages..
The second however is unable to play. it somehow has realplayer marked
as the associated program to play through and I have no idea how to
change this. Any help is appreciated on how to fix the one computer so
it uses
Does Anyone have paging working on a Zultys phone? If So any urls you
may be able to pass my way so I can attempt to get it working.
Thanks
ron
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Erros on boot. Running Release 1 Latest Version.
Feb 2 13:32:20 localhost kernel: Zapata Telephony Interface Registered
on major 196
Feb 2 13:32:20 localhost kernel: Freshmaker version: 71
Feb 2 13:32:20 localhost kernel: Freshmaker passed register test
Feb 2 13:32:29 localhost kernel: Module
Looking to see if anyone has a WElltech 3804 Config that they would
be willing to share.
I will update/place them on the wiki if I get one that
works.
thanks
ron
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So I have read and read and read... google is my friend and the wiki is
by brother...
However, I am still unclear on what the preferred method of using the
pound sign is.
If the Pound sign is set aside for Transfer.. Then when I make an
outbound call to my bank I can not Enter my PIN followed by
Good Day List,
I have my asterisk box setup to be an ntp server, and my zultys
4X4 phone is able to get the time, however
I must first select the TimeZone Offset and then it will use the
time setting from my server.
This is a hassle because every time the phone reboots
only seems to
control the value that you are prompted with when the phone boots.
If I get a solution, I'll let you know.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Mon, 17 Jan 2005, Ronald Hartmann wrote:
Good Day List,
I have my asterisk box setup
, 17 Jan 2005, Ronald Hartmann wrote:
I have been reading the RFC http://www.faqs.org/rfcs/rfc2132.html on
this and I think the issue may be related to the setting of the Time
Offset
3.4. Time Offset
The time offset field specifies the offset of the client's subnet
in
seconds from
Has anyone had any luck getting the PARK button on the Zultys
4X4 to work with * ?
It would be very nice to get that to work
I have tried putting in #70# to have the phone send the Pound
to initiate transfer then 70 (my parking lot) the Pound to finalize
transfer.
Good Day List,
I am looking for some feed back on a new Asterisk Box.
I am looking to spend my money wisely on a new dev box.
I have been given the go ahead on the following expenditure and
seek your advice.
SuperMicro X5DPA-GG MB
Key Features
Good Day List,
I am finalizing my research on the ACD Ques and have (what I
hope to be) one last hurdle.
Is there any feasible way to determine if a que has agents
currently available to take a call.
I have looked at the Show Queues, show queue quename, show
agents
Check the signallin gthat you have for the card.
Remmember that FXO ports require FXS Signalling
And FXS ports require FXO Signalling.
Confusing you bet
If this does not help then you will need to forward come config files.
Ron
-Original Message-
From: ismaelg [mailto:[EMAIL
Good Day again list,
Encountered another problem in the ACD queue...
If I use the ADDQueueMember to dynamically add members as
foolows,
exten =
403,1,AddQueueMember(techsupport|SIP/${CALLERIDNUM})
lets assume I called extension 403 from my extension 2204.
Yes, its me again!
I have found an issue with AgentCallBackLogin.
If there are calls in the queue waiting to be transferred to an agent.
And then I log into take calls via the AgentCallBackLogin The call in
the queue
Is transferred to me faster than I can enter in my password, and the
Queues.conf
I understand that by setting the variable context,
I can have asterisk drop a call out of the ACD Queue and send it to the context_location specified by the context variable;
Is it possible to have asterisk only drop out of the queue
if only a certain number is pressed.
: [Asterisk-Users] ACD Queue question.
-Original Message-
From: Ronald Hartmann [mailto:[EMAIL PROTECTED]
Sent: Monday, January 10, 2005 8:46 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] ACD Queue question.
Queues.conf
Is it possible to have asterisk
Good Day list,
Thanks for
taking the time to reading this posting!!!
I have read
a plethora of postings about the TDM WildCard and
analog lines. And have the following
question.
Many people
have recommended the Sipura products, and others have
recommended a channel bank
Tried downloading the newest Zaptel Stable from CVS.
When I try to do a make it errors out.
I have used google and I see lots of posts to the pastebin but
have not found any resolutions.. I am hoping that someone in here can
assist me.
Thanks for any links or words of
Good Day list,
I have a friend who is interested in implementing an asterisk
implementation at his offices.
The configuration would consist of the following
Site A Asterisk Box With 12 incoming lines and 15 phones
Extensions 101-115
Site B
Good Day list,
Anyone know if there is a way to have the AgentCallBackLogin
function play a voice file after the agent picks up the phone?
If this is not an available feature, any ideas on the difficulty
in making this feature?
Example:
Extensions.conf
It has been narrowed down to the CONTEXT the call was originally located
in.
I need to figure a way within the parking application to set the
extension I want the call to ring back to.
Specifically I want to set it to always ring back to the phone that
parked it.
Example, below
..
When the
Ok I have read every line in the wiki for call parking.
HELP PLEASE
I need to perform a simple call parking that upon timeout will ALWAYS
rings back to the extension that actually parked the call.
Am I just crazy or is this not possible.
It has been narrowed down to the CONTEXT the call was
Trouble in Parking Paradise!
Good Day all,
I have a situation that I have tracked as far as I can take it and am
looking for assistance into the matter.
My setup. Asterisk 1.0.1 with the AMP config
environment.
When I have auto attendant answer the phone and I dial my
Good Day,
Well I had the box up and running accepting an incoming pstn line.
Then made a change to the zap conf file and now asterisk will not start
001 Had the box up and running
002 went to change the zapata.conf to include my pstn lines
003 and then rebooted and asterisk will not load.
That darn signaling issue
Gr
Anyways..
I Must recall that FXO ports get FXS signaling
And vice versa.
-Original Message-
From: Ronald Hartmann [mailto:[EMAIL PROTECTED]
Sent: Thursday, November 04, 2004 3:18 PM
To: [EMAIL PROTECTED]
Subject: FW: ZapTel problems
Good
I understand about the soft button and yes this does work, however I am
trying to figure out if the actual Button (the mechanical one on the
phone)
That says transfer is it possible to get this to work.
~ron
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Good Day list,
I have spent better part of the morning reading through the user group
messages and have found some people stating that they are able to get
the Transfer Button to work on the Snom 190/220
However, I am unable to find HOW they pulled this off. When I press the
transfer button it
WookSung TelephoSee 2000 Help needed.
Can not get the phone to register
with asterisk. I am not sure what the problem is at this
point.
I have the setup of the phone as:
Server1 192.168.3.1
Port1: 5060
Display: TelephoSee
URI: blank
Userid: 2205
Password: password
Good day list,
Need
some assistance in setting up the snom190 with asterisk.
My
voicemail server is at extension *98.
I
have successfully been able to leave a message for an extension
The
MWI comes on and I can check messages by dialing *98 on the snom and all is
great.
Problem fixed for the Vmail Soft Key
I had the userfrom in the wrong context.
Anyhelp on the hold button is still appreciated.
Good day list,
Need
some assistance in setting up the snom190 with asterisk.
My
voicemail server is at extension *98.
I
have successfully
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