John Brown (CV) wrote:
Hi List,
Just a quick note that we have cleared all back logs of Grandstream
product. If you have been awaiting shipment, its shipped. Everyone
should be getting tracking numbers shortly.
We also have NEW STOCK that can ship within 2 to 3 days of order
BT-101
BT-102
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal or another
Implementation wise, it would be more frustrating to kick the already
registered user off, and make it more likely it'd be noticed if there
where two registered people.
On Thu, Dec 11, 2003 at 10:15:10PM -0800, Chandra wrote:
last time i was experimenting IAXClient as a true client from dial up
SW wrote:
Hi folks,
I want to fix hole in my asterisk set up.
I use Vocal as my sip proxy and * for voice mail and the g/w to PSTN,
Iconnect, fwd etc. So from Vocal I redirect sip requests which needs to go
'other' places. This senario works fine.
Now the issue is someone else running a vocal
On Tue, 2003-12-09 at 05:10, listas iPfone wrote:
Hi
The version 1.260 of chan_sip.c already have that patch?:
http://bugs.digium.com/file_download.php?file_id=430type=bug
That link didn't work for me, but the NAT patch has not been put into
CVS yet. It needs to be TESTED more, so if you
Hi
Have you got the context set-up in the sip.conf to say which extension
context to use for incoming calls fro FWD Iconnect.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson
Sent: 12 December 2003 08:01
To: [EMAIL PROTECTED]
Hi all
Disregard my last post I replied to the wrong e-mail, I should have replied
to an off list e-mail.
That will teach me not to put my glasses on.
Dave
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Olle E. Johansson
Sent: 12 December 2003 08:01
To:
Hi,
When I build the spandsp library needed for the FAX application, running
./config I get the following with 'no':
[EMAIL PROTECTED] spandsp-20031021]# ./configure
...
checking for _doprnt... no
checking for pow... no
checking for sqrt... no
checking for rint... no
checking socket.h
Hi everyone ,
I have installed festival , following this guide :
http://www.voip-info.org/wiki-Asterisk+festival+installation
But i didn't get festival working with asterisk yet :-/
I got this message from festival server :
-=-=-=-=-=- EST Error -=-=-=-=-=-
{FND} Feature Token_Method
Hi
I have applied the patch, I can register a Grandstream 100 from another
internet connection but I get no audio and a timeout line drop after 5
seconds.
If I call my SipPhone number 17476691936 I hear my welcome message and again
the line times out and drops after 5 seconds.
I notice that the
On Thu, 11 Dec 2003 23:46:46 -0800, andrewg wrote
Implementation wise, it would be more frustrating to kick
the already registered user off, and make it more likely
it'd be noticed if there where two registered people.
Hmm, frustrating but maybe useful, if you were for example on a
Everythings works great with asterisk exept one feature
with redirect : it doesn't redirect when ringing ...
BTW are their any plans to extend the manager API
??
Michael Devenijn
I have noticed some strange behaviour when using messenger as a sip client.
Messenger appears to stop transmitting RTP like some sort of voice activity
detection, and some applications on asterisk also respond by ceasing/not
starting RTP transmission until they get something from messenger.
Hi,
does anybody has experience with using CLIP for FXS-phones with E100P and
TDM400 installed ?! Any little help would be great !
Thanks,
Johannes
**
Johannes von DrachenfelsTelefon:+49 7231 922380 0
I've tried getting this running but mpg123 won't spawn. It spawns fine for
the files but if I try streaming she doesn't work.
I've tried with just about every stream at somafm.com w/o success. I can
play them locally though.
When I try to play them from the server from the command line I get:
#
Hi,
- Original Message -
From: Dan Fernandez
To: [EMAIL PROTECTED]
Sent: Sunday, November 23, 2003 5:45 PM
Subject: [Asterisk-Users] RxFax
I am also having problems receiving my first fax. I get a 320byte file
(for a
4 page fax). If I look a the tiff generated, is just has some few
Hi,
It would be great if the IAX protocol will be able to tranfer fax data (even
converted in another format) between Asterisk boxes, using low bandwidth
codecs like GSM.
I know that this is possible only with the G.711 now (passing faxes using
the audio stream), but maybe in the
Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer fax
data (even converted in another format) between Asterisk boxes, using
low bandwidth codecs like GSM. I know that this is possible only with
the G.711 now (passing faxes using the audio stream), but maybe
Thanks! I was hoping you wouldn't say that... ;) It is always such a
joy getting the carrier to cooperate.
mattf wrote:
You have to get the local calling information from the carrier that the
lines go through. We have 6 local T1s in our office and they are in 3
different groups of local
Hi,
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 3:42 PM
Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer
Hi,
faxing works great with IAX/IAX2. Even over a 100ms ADSL link.
It does not really depend on the protocol, only on the codec.
regards
kapejod
Am Fr, 2003-12-12 um 14.42 schrieb Senad Jordanovic:
Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer fax
data
- Original Message -
From: Klaus-Peter Junghanns [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 9:02 AM
Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
Hi,
faxing works great with IAX/IAX2. Even over a 100ms ADSL link.
It does not
unsubscribe
-
This mail sent through IMP: http://horde.org/imp/
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Dan wrote:
Hi,
- Original Message -
From: Senad Jordanovic [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 3:42 PM
Subject: RE: [Asterisk-Users] FAX, IAX and *Maybe I'm
dreaming...:-)
Dan wrote:
Hi,
It would be great if the IAX protocol will
Could this possibly be submitted to the bugtracker, if it hasn't
already? Please submit changes in diff -u format to allow quicker
integration.
JT
At 10:06 PM -0500 12/9/03, Adam Rothschild wrote:
On 2003-12-09-20:20:12, Andrew Kohlsmith
[EMAIL PROTECTED] wrote:
I have never been able to
Hi all,
I searched through the archives, but found nothing...
Is there a possibilty, to take over a call ??
I have for example two extensions..
102 and 103
if 102 is ringing, but noone one the desk, I want, that 103 can answer
this call on his phone, by just typing some digits...
has
Hi,
It would be great if the IAX protocol will be able to tranfer fax data
(even
converted in another format) between Asterisk boxes, using low bandwidth
codecs like GSM.
I know that this is possible only with the G.711 now (passing faxes
using
the audio stream), but maybe in the
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 8:25 AM
Subject: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
It would be great if the IAX protocol will be able to tranfer fax data (even
converted in
On 12/12/03 13:56, Dan wrote:
This is because the fax is transmitted using the audio stream.
It is not related to the signaling protocol (SIP/IAX etc.) but to the audio
codec used.
Fax uses FSK modulation to transmit the data. If you compress this in a
lossy way (GSM, MP3, whatever) then the
Thanks - it worked.
Paulo,
That might be a bug in iaxclient -- it should only advertise itself as
supporting GSM, since that's all it currently supports. We'd need to
investigate that a bit.
Anyway, if you just disallow=all and then allow=gsm, it will work for
you.
Hi,
- Original Message -
From: Jim Flagg [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 4:54 PM
Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
- Original Message -
From: Dan [EMAIL PROTECTED]
To: Asterisk Users [EMAIL
Hi,
- Original Message -
From: Alastair Maw [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 4:58 PM
Subject: Re: [Asterisk-Users] FAX, IAX and *Maybe I'm dreaming...:-)
On 12/12/03 13:56, Dan wrote:
This is because the fax is transmitted using the audio
Your order was picked up on THursday by UPS.
All HT-286 orders have been filled.
On Fri, Dec 12, 2003 at 01:57:02AM -0500, Brian Capouch wrote:
John Brown (CV) wrote:
Hi List,
Just a quick note that we have cleared all back logs of Grandstream
product. If you have been awaiting
Hi,
I've compiled the speex codec and installed as per
docs, speex works perfectly between Asterisk gateways using IAX
protocol.
My problem is between X-Lite softphone and a
Asterisk Gateway users cant hear anything when the speex codec gets
selected.
Has anyone else experienced this
Is there a simple way to enable/disable on-the-fly a pri channel?
I want to control the number of incoming lines based on the number of agents I
have.
I took a look at libpri and zaptel and found some hints about a SERVICE
message within the pri protocol, but coudn't find out much more about
jerk face wrote:
The following is from zapata.conf.sample:
; Ring groups (a.k.a. call groups) and pickup groups.
If a phone is ringing
; and it is a member of a group which is one of your
pickup groups, then
; you can answer it by picking up and dialing *8#.
For simple offices, just
; make
Hello,
Anybody has it working with asterisk? Could you share your experience (
good/bad)
Thank you
--
Anton Yurchenko[EMAIL PROTECTED]
Digital Generation
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Hi,
Citeren Steve Kann [EMAIL PROTECTED]:
That might be a bug in iaxclient -- it should only advertise itself as
supporting GSM, since that's all it currently supports. We'd need to
investigate that a bit.
Anyway, if you just disallow=all and then allow=gsm, it will work for
Any proposals for how to overcome these issues? Any volunteers? :-)
There shouldn't be any great need to HTML-ify the CVS commits, although an
RSS feed would kick ass and be simple to do (I think) -- I might give a
crack at HTMLifying the commits to make them prettier though... What do
Hi,
Citeren Steve Kann [EMAIL PROTECTED]:
It would be great if the IAX protocol will be able to tranfer fax data
(even
converted in another format) between Asterisk boxes, using low bandwidth
codecs like GSM.
I know that this is possible only with the G.711 now (passing faxes
using
the
Zope Corporation is seeking bids from qualified service providers for a
prototype Asterisk deployment. The RFP is available online at:
http://www.zope.com/AsteriskRFP
Regards,
Rob
--
Rob PageV: 540.361.1710
Zope CorporationF: 703.995.0412
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 11:16 AM
Subject: Re: [Asterisk-Users] Newbie introduction /* New subject */
Any proposals for how to overcome these issues? Any volunteers? :-)
There shouldn't be
wasim: yes i am! actually i always was ;-)
ok, here is my setup
Fax machine - ISDN pbx - (chan_capi) - Asterisk 1 - LAN(chan_iax2)
- Asterisk 2 - Internet 100ms (chan_iax2) - Asterisk 3 - (chan_capi)
- ISDN pbx - Fax machine
i use plain iax2, no trunking.
regards
kapejod
Am Fr, 2003-12-12 um
I have the MGCP-only version, the DG-104S
Works great for me.
mgcp.conf:
;
; MGCP Configuration for Asterisk
;
[general]
port = 2427
bindaddr = 192.168.0.254
[DG104S]
host = 192.168.0.130
threewaycalling = yes
transfer = yes
callwaiting = yes
context = from-internal
callgroup = 1
pickupgroup =
Hi there,
two questions concerning ISDN BRI, Fritz! passive and chan_capi:
1. I noticed that I run into trouble if I do a capiinit stop and
start while Asterisk is running. Is that normal, or do I need to twist
my configuration somehow?
Background: ISDN callers normally get a voice menu,
Hi Johannes,
i havent tried it myself, but in .de phones use the same FSK-style
callerid like in the US. I have seen .de CLIP phones working with
an ATA186, so the digium FXS cards should do that too.
best regards
kapejod
Am Fr, 2003-12-12 um 12.51 schrieb Johannes von Drachenfels:
Hi,
On Fri, Dec 12, 2003 at 01:57:02AM -0500, Brian Capouch wrote:
John Brown (CV) wrote:
Hi List,
Just a quick note that we have cleared all back logs of Grandstream
product. If you have been awaiting shipment, its shipped. Everyone
should be getting tracking numbers shortly.
We
I have a few dg104s not sh they seem to work fine after firmware upgrade I
some trouble with NAT
I have not used any other equipment so I can't say how they compare to
anything else
James Schenck
Egraph Design Inc.
Arkansas Online Internet Services
(870) 857-3287
IAXTEL (700) 857-3287
[EMAIL
Hi!
I searched through the archives, but found nothing...
Looks like you didn't invest too much in that search...
Is there a possibilty, to take over a call ??
That's a basic feature called pick-up. Try *8 or *8# and look at the
config options pickupgroup and callgroup. Next to that you
Hi list,
Does anyone use the .deb package of asterisk? Is it stable? woks fine?
thanks
Eduardo
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Alastair Maw wrote:
Fax uses FSK modulation to transmit the data. If you compress this in
a lossy way (GSM, MP3, whatever) then the integrity of the data is
affected (more or less seriously depending on the codec used). Fax
machines are generally quite picky, so compressing faxes is unlikely
Hi!
Sorry I didnt mention that, what if the line I`m transfering too is busy?
How do I return the call then?
It all depends on what you do in extensions.conf and the +101 priority of
the extension that is busy. If you do nothing then the call is simply
gone and you can't do anything about
Hello..Sorry for posting this herebut I
cant see any other way to get a hold of JOHN BROWN
I placed an order of 4 SPA2000s with Chagres
Technologies over 2 months ago. John what is the status of my
order? I have emailed, faxed and called.but still no reply from
you or your company. If
I have my connection as below,
diax(IAX) --- (IAX) * (IAX) -- IAX(*) --- PSTN
In the middle tier of asterisk, it if not completely forwarding the
stream and it consumes the system bandwidth.
I DONT have settings like notransfer=yes
Can you please help me how can I
something is starting to stink around
here
- Original Message -
From:
Darren Browning
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 12:14
PM
Subject: [Asterisk-Users] Chagres
Technologies _WHERE IS MY ORDER?
Hello
..Sorry for posting this
Anyone here have any good/bad things to say about first hand experience
with the new Wifi SIP phones? I am considering one for my office as an
alternative to FXS+Analog cordless.
Thanks,
Michael Graves
--
Michael Graves [EMAIL PROTECTED]
Sr. Product Specialist
John, did you ever get any feedback from the GS wish list?
I love the BT-102's with 1 exception. The speaker phone.
I have not come up with a combination that makes it acceptable.
If I had a way to cover up that button I would go ahead and deploy the
phone.
But the db level and echo to the far
If anyone on the list has successfully configured and used the GS
speaker phone,
could use please share
Great fix, replaced with Cisco 7960
--
DataCrest, Inc. -- Technically Superior **
Walker Haddock http://www.datacrest.com
Hi,
1) you are doing capiinit stop/start while * is running?
do you think that unloading a device driver while * is still
running is a good idea? ;-)
2) there is no (clean) way to determine if another application
is using a B channel. However you will get a circuit-busy
reason back from
Hi list,
Well I really didn't want to see things get to this point,
but Sherman at Sipura along with their President Jan F.
leave me no other choice.
SIPURA has been provided a letter from our attorney for
Breach of Contract and damages. They have yet to respond.
A quick background.
1.
I have a 7960 phone running V6 SIP. When in a telnet session with the phone,
I can ping other devices on the local network. I can't ping anything outside
the local net. I can't ping anything on the outside of the router. None of
the computers on the subnet have this problem, they can ping outside
Walker Haddock wrote:
If anyone on the list has successfully configured and used the GS
speaker phone,
could use please share
Great fix, replaced with Cisco 7960
Almost the same fix as mine, Snom 200's.
Now I just need to fix the bottom line.
--
Bob Knight
[-w] the work option
These are just claims.
Post a pdf of the contract.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Brown
(CV)
Sent: Friday, December 12, 2003 8:21 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIPURA Breaches Contract
2. SIPURA and
Does this kind of stuff really belong here? Heck for all
we know you guys did something dumb to make them mad at you
and they had good reason for their actions.
Who knows, why should we care??
Someone here said you guys charged him _prior_ to
shipping an item. If anyone charged me
On Fri, Dec 12, 2003 at 10:49:19AM -0800, Paul Mahler wrote:
I have a 7960 phone running V6 SIP. When in a telnet session with the phone,
I can ping other devices on the local network. I can't ping anything outside
the local net. I can't ping anything on the outside of the router. None of
the
Hi!
I am using X-Lite and NetMeeting.
When I call from netmeeting to X-lite or from X-lite to netmeeting, the call is
stablished correctly. But after some seconds, they hang up. I get some errors
messages:
- 7:38.574 H245:8122c90 H245Read error: Bad file
descriptor
-
I have a 7960 phone running V6 SIP. When in a telnet session with the phone,
I can ping other devices on the local network. I can't ping anything outside
the local net. I can't ping anything on the outside of the router. None of
the computers on the subnet have this problem, they can ping
Hi all, I installed the estara softphone and had no
problem registering it with asterisk. I could make
calls to other hardware SIP phones (Cisco 7960) from
the softphone, but I couldn't call the softphone from
the Cisco 7960s. The asterisk console gave me an error
message saying unable to create
Florian Overkamp wrote:
Hi,
Citeren Steve Kann [EMAIL PROTECTED]:
It would be great if the IAX protocol will be able to tranfer fax data
(even
converted in another format) between Asterisk boxes, using low bandwidth
codecs like GSM.
I know that this is possible only with the G.711 now
On Friday 12 December 2003 07:25, Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer fax
data (even converted in another format) between Asterisk boxes,
using low bandwidth codecs like GSM.
I know that this is possible only with the G.711 now (passing faxes
using
John Breeden wrote:
These are just claims.
Post a pdf of the contract.
It is in NO ONE'S best interest that he do so.
Legal matters belong in courts of law, not email discussion lists.
I daresay most lawyers would have advised against the original posting.
I am not a lawyer, but my layperson's
it's a firmware problem on GS, they are working on that but it seems its
not that simple to make volume higher on the speaker and echo go away,
anyway 4.26 seems stable for now and with many new features!
Miguel
On Fri, 2003-12-12 at 17:55, Bob Knight wrote:
John, did you ever get any feedback
In sip.conf do you have
type=friend
for your softphone?
If not you'll only be able to send or receive calls depending on the option you
selected.
-sb
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Hao Zhong
Sent: Friday, December 12, 2003 2:29 PM
To:
it's a firmware problem on GS, they are working on that but it seems its
not that simple to make volume higher on the speaker and echo go away,
anyway 4.26 seems stable for now and with many new features!
Miguel,
What are the new Features?
Robert
___
On Friday 12 December 2003 07:25, Dan wrote:
Hi,
It would be great if the IAX protocol will be able to tranfer fax
data (even converted in another format) between Asterisk boxes,
using low bandwidth codecs like GSM.
I know that this is possible only with the G.711 now (passing faxes
using the
biggest is DSL can now be connected directly to the GS no need of dhcp
or static ip no more, mute/del button works now the date tap on the
screen and many more cant remember download it at
http://www.grandstream.com/TEMP/FIRMWARE/
or update with 4.3.153.50 tftp
Miguel
On Fri, 2003-12-12 at
On Friday 12 December 2003 02:47 pm, rnc Info Lists wrote:
it's a firmware problem on GS, they are working on that but it seems its
not that simple to make volume higher on the speaker and echo go away,
anyway 4.26 seems stable for now and with many new features!
Miguel,
What are the new
- Original Message -
From: Bob Knight [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 12:49 PM
Subject: Re: [Asterisk-Users] John Brown from Chagres!
Almost the same fix as mine, Snom 200's.
Now I just need to fix the bottom line.
Have you gotten MOH to work
Hi,
This is our test setup: 4 phones, 2 logged into the one queue, the other
2 phones are used to dial into the queue.
If there are 2 calls coming into the queue at the same time, we would
like to have the 2 queue phones ringing at the same time (one for each
call). But as it is, the 2nd phone
Hi Scott, thanks for the reply. Here is how my
sip.conf looks like for the softphone, I tried type
'3Dfriend' and asterisk didn't like it.
[hzhong-desk]
type=friend
username=hzhong-desk
callerid=Hao Zhong Desk 8005
mailbox=8005
secret=cisco
nat=no
host=dynamic
canreinvite=no
qualify=200
Title: Message
Hi
Miklos,
try
starting * with -vvvc and see if there is any
warning
also,
try connecting to your mysql server by issuing mysql asteriskcdrdb then
show tables;
select * from cdr;
best,
PHM
-Original Message-From:
[EMAIL PROTECTED]
Hi!
I just installed the mysql cdr support and my database is not registering
the calls :(
It is necessary to unload the cdr_csv.so? how to do it?
No.
[global]
hostname=localhost
dbname=asteriskcdrdb
password=new_password
user=asteriskcdruser
Make sure you can connect from the box
Title: Message
Were
it me, I would see if MySQL is creating the socket where Asterisk expects
it.
-Original Message-From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of listas
iPfoneSent: Friday, December 12, 2003 1:47 PMTo:
[EMAIL PROTECTED]Subject: [Asterisk-Users]
I had a similar problem with my 7960 phones.
It ended up being a problem with quotes in the SIP.cnf file.
Do a sip show peers from the console to see if the 7960 is registered properly.
For a test set the following values in the cnf file
line1_name: 8005
line1_shortname: 8005
I'm limited in what I can say at this point. However it
would seem that there was a very productive conversation
between the parties and both have strong interest in establishing
a positive relationship.
Chagres remains committed to the SIPURA product and I do personally
believe they have a
Hi,
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
On Friday 12 December 2003 07:25, Dan wrote:
You're not going to get that working because GSM is a lossy codec.
It is able to get extreme savings in size, because it optimizes out
parts of the sound that most humans
Hello everybody,
First time installer and I need the lists advice. My plan is to use
asterisk PBX with some hardware to terminate my calls coming from
several operational gnugk gatekeepers.
Do have RH9 and downloaded the latest asterisk from CVS. Compiled
according instructions and is running
Its going to try one at a time till its answered thats how its designed.
Trying to call more than one person at a time might cause more drama than
its worth. Just tell your agents to answer their phones faster... if they
dont fire them.
bkw
On Thu, 11 Dec 2003, Markus Mayer wrote:
Ok, so let
but, if this is case then how can you run a call center with asterisk?
What if you have 40 simultaneous calls coming into the call center, most
calls would be missed, even if you have 40 available agents. Of course
one call should go to one agent, but if a second call, or a third call
joins the
On Fri, 2003-12-12 at 13:33, Brian West wrote:
Its going to try one at a time till its answered thats how its designed.
Trying to call more than one person at a time might cause more drama than
its worth. Just tell your agents to answer their phones faster... if they
dont fire them.
bkw
Hi Scott, my 7960s can call each other without any
problem. I changed ip.conf as you recommended, and
still didn't work. But from sip show peers, it looks
like my softphone is not talking to asterisk properly.
Asterisk got the softphone's IP address, but its
status is unreachable, I'm trying to
Ok, has already been answered. Thx, folks.
-Markus
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On Friday 12 December 2003 16:07, Christian Hoffmeyer wrote:
On Fri, 2003-12-12 at 13:33, Brian West wrote:
Its going to try one at a time till its answered thats how its
designed. Trying to call more than one person at a time might
cause more drama than its worth. Just tell your agents to
- Original Message -
From: Markus Mayer [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 4:25 PM
Subject: Re: [Asterisk-Users] Simultaneous incoming calls
Ok, has already been answered. Thx, folks.
-Markus
For the sake of Googlers to come, give the resolution
- Original Message -
From: Tilghman Lesher [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, December 12, 2003 4:36 PM
Subject: Re: [Asterisk-Users] Queue only ringing one agent at a time
Pissing on the gurus is also not a recommended course of action.
Drawing turf lines over
Drawing turf lines over useless information doesn't foster a helpful,
collaborative environment no matter who is giving out the useless advice.
FYI: Brian actually has spoken with both myself and Derek about our Asterisk
implementation, and therefore knows us well enough to tell us to fire our
Instead of quacking out useless information, it's more useful to not answer.
Their are alot high places on this planet.. pick one and jump.
bkw
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agents if they're too slow answering the phones. He was kidding. :)
yes I was kidding... And a group of people are working on a new plugable
agents and queues setup. So far from what I have seen/heard its going to
rock. I magine loading a xyz_acd.so for xyz call strategy and it just
becomes
On Fri, 12 Dec 2003, Derek Barber waxed:
but, if this is case then how can you run a call center with asterisk?
What if you have 40 simultaneous calls coming into the call center, most
calls would be missed, even if you have 40 available agents. Of course
one call should go to one agent,
When you look at 'show translation' you see:
Translation times between formats (in milliseconds)
but is this the number of milliseconds required to convert 1 packet of
data, or the amount of time required to translate 1 second of data? I am
assuming it is the time for 1 second.
Thanks,
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