On 2/14/07, Yuan LIU [EMAIL PROTECTED] wrote:
Well, you'll have to decide how you want to hang up the caller: Do you
want him/her to be ignored, or to be told that you are not available (like
an answering machine)? You also need to tell Asterisk how to determine
if
the next invite comes from
Hi!
i have a problem with the PICKUPMARK of the Pickup() application.
E.g. A calls B. B is ringing. C wants to pickup B.
To make this work with PICKUPMARK I have to add the variable PICKUPMARK
to B. But how can I do this? B is just created inside the Dial()
application.
thanks
klaus
PS:
On Thu, 15 Feb 2007, jameson asterisk wrote:
I'm currently looking to deploy an Asterisk server using an FXO media
gateway to connect to the PSTN and was looking for any user experiences that
may aid in selecting a gateway. Specifically i'm looking for a 4-port model
under 500 dollars.
Within
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On 15 Feb 2007, at 01:39, Leo Ann Boon wrote:
Bruce Reeves wrote:
In my experience having ap's with the same SSID and 3 channels of
separation overlapping worked if the phone could roam.
Recommended is 5 channels of separation.
Ronald,
Just be
- Original Message -
From: Nic Bellamy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 13, 2007 8:40 PM
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Larry Shields wrote:
Jens Vagelpohl wrote:
I have two APs (Apple AirPorts) sending on the _same_ channel.
Handover works perfect with no discernible loss of connectivity or
audio using a Siemens SL75. The handover cannot even be noticed.
as I know, best practice says, that neighboring AP should use _non
Hi all,
Does anyone know of an IAX client for Symbian? I have an e61 and would
like to make calls through my home Asterisk box from places where I have
WiFi access, as NAT is in the way I suspect that it'll be a pain to get
SIP working like that as the NAT router doesn't do SIP connection
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Hash: SHA1
On 15 Feb 2007, at 10:23, Pavel Jezek wrote:
Jens Vagelpohl wrote:
I have two APs (Apple AirPorts) sending on the _same_ channel.
Handover works perfect with no discernible loss of connectivity or
audio using a Siemens SL75. The handover
Pavel Jezek ha scritto:
Jens Vagelpohl wrote:
I have two APs (Apple AirPorts) sending on the _same_ channel.
Handover works perfect with no discernible loss of connectivity or
audio using a Siemens SL75. The handover cannot even be noticed.
as I know, best practice says, that neighboring
From: Il Neofita [EMAIL PROTECTED]
Date: Thu, 15 Feb 2007 03:37:14 -0500
But I tought that hangup was suppose to close the call, however, is not the
case and a really did not catch why.
Now I see where the confusion comes from. Asterisk doesn't really speak
English - or Chinese for that
Hi all,
I make mistakes in my explanation, so I will try to re-explain my
problem...
I want to send fax with FoIP.
Analog Fax PATTON SN4960 Asterisk PATTON M-ATA
Analog Fax 2
In the Patton SN4960 configuration I have :
profile voip FOIP
codec 1 g729 rx-length
Hi,
cud any one help me figuring out the problem... When the agent in a queue is
engaged, it cannot accept anymore calls, below is the script;
-- AGI Script dialparties.agi completed, returning 0
-- Executing Dial(Local/[EMAIL PROTECTED],2, SIP/2063||tr) in new stack
-- Called 2063
Hi James,
The only solution I've managed to find so far is to set the wrap-up time
to 5 seconds and tell the operators that if they need more time, they
need to put themselves on pause. See PauseQueueMember and
UnpauseQueueMember.
If someone has a better solution, I'd be most pleased to
Hi all,
I make mistakes in my explanation, so I will try to re-explain my problem…
I want to send fax with FoIP.
Analog Fax ← PSTN → PATTON SN4960 ←T.38→ Asterisk ←T.38→ PATTON M-ATA ←Analog→
Analog Fax 2
In the Patton SN4960 configuration I have :
profile voip FOIP
codec 1 g729 rx-length
Yes I m using E1 the equivalent of T2 (31 channels)
2007/2/14, Melcon Moraes [EMAIL PROTECTED]:
You should answer questions asked to you. I saw Tzafrir Cohen asking you
if you were using a E1 PRI. Are you?
[]'s
MM
-Original Message-
From: younss azzayani [EMAIL PROTECTED]
To:
ok thank you Cohen thank you very much
2007/2/14, Tzafrir Cohen [EMAIL PROTECTED]:
On Wed, Feb 14, 2007 at 03:44:25PM +, younss azzayani wrote:
Thank You Cohen
What card do you have?
*
Digium TE110P TDM400P, think the problem is with TE110P (configured
as span 2)
Hi,
Even a default Slackware 11.0 with 2.4.33 kernel source failed to look for
page-flags.h and I do
not have a card that your are referring to because this is a development
machine on a laptop. It
used to work before but the current source tree which i get into a week ago
started to break out
This is a PA-1688 chip phone.
Give a look at http://www.aredfox.com/. It has what you need.
Look for Pamtool.
Isamar
On Wed, 14 Feb 2007, Alcides Cremonezi wrote:
Hi! Everyone,
This IP phone came configured for to be used with Soyo VoIP service.
I would like to set it up to work with my
Wireless wrote:
- Original Message -
From: Nic Bellamy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 13, 2007 8:40 PM
Subject: Re: [asterisk-users] The High Performance Echo Canceller (HPEC)
Hello all,
Ive found another issue with the queue application. Assuming Ive
configured a queue with a long periodic announcement and have two queue
members assigned. Both queue members are busy at a time, while another
caller is joining the queue. After a while the periodic announcement is
Hi All,
I'm using asterisk 1.2.15 and call forwarding doesnt work for me.
from my extensions.conf:
; Unconditional Call Forward
exten = _*21*X.,1,NoCDR
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten = _*21*X.,3,Playback(vm-saved)
exten = _*21*X.,4,Hangup
exten =
I am not aware of one.Why would you want your queue announcement
interupted? When we had our Nortel, I found that feature annoying because
people would be transfered to the agent half way through a message.
Confusing. I configured it to not break out of an annoucement.
On 2/15/07, [EMAIL
Am 15.02.2007 um 14:06 schrieb Dominik Zalewski:
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
Just use ${CALLERID(num)} and not ${CALLERID(NUM)}.
Stefan
--
amooma GmbH - Bachstr. 126 - 56566 Neuwied - http://www.amooma.de
Let's use IT to solve problems and not to create
On Thursday 15 February 2007 6:51 am, Steve Underwood wrote:
It looks like octasic have started supplying their echo canceller as
host software for zaptel now. I expect either canceller would work with
the Sangoma cards, as they currently sit in the zaptel framework too.
Out of curiosity, why
Hi Demuel,
Look, i think (im not very sure yet) that the *page-flags.h* file
belongs to kernel = 2.5.x, not to the 2.4.x,
Im using Slackware 10.2 , I have not upgraded yet to the 11.0 but i
think that it comes with the 2.6.x kernel like a native kernel (not in
/test/ directory anymore),
On Thursday 15 February 2007 03:22:58 pm Stefan Wintermeyer wrote:
Am 15.02.2007 um 14:06 schrieb Dominik Zalewski:
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
Just use ${CALLERID(num)} and not ${CALLERID(NUM)}.
Stefan
it didnt help :( Is there is other way to
On 2/15/07, Dominik Zalewski [EMAIL PROTECTED] wrote:
Hi All,
I'm using asterisk 1.2.15 and call forwarding doesnt work for me.
from my extensions.conf:
; Unconditional Call Forward
exten = _*21*X.,1,NoCDR
exten = _*21*X.,2,Set(DB(CFIM/${CALLERID(NUM)})=${EXTEN:4})
exten =
Hi,
I observed that too. I already got that 2.6.x kernel and it is there actually.
Though Patrick has
put a 2.6.x kernel in /extra, I am still thinking if I will upgrade my existing
kernel in this
laptop. The maintainer of slackware did not made the 2.6.x as the default
kernel for some other
On Feb 15, 2007, at 1:12 AM, jameson asterisk wrote:
Can anyone provide a recommendation based on user experience?
Feel free to suggest an alternative gateway if one stands out.
I've been working with the Grandstream GXW-4108 (the 8 port version
of the 4108), and it was a rough start, but I
Hi,
my Asterisk 1.2.9.1 suddenly freezed (stop now did not work!!) . I
found the following errors inside /var/log/asterisk/message:
Feb 14 14:55:41 ERROR[11273] rtp.c: Unable to allocate socket: Too many
open files
Feb 14 14:55:41 WARNING[11273] chan_sip.c: Unable to create RTP audio
Is there any calendar client that can point at OX for calendar data,
which client can display multiple calendars simultaneously as
*overlapping layers* in the GUI? With UI to de/select calendars from
view, one by one. That is, a single grid of days displayed, with the
events in each day
you just post only call forward activation part of dialplan,
but you must also make dialplan part, that reflect, how is set this
callforward mark,
ie. if callforward is set, dial that number, if not, dial peer...
Dominik Zalewski wrote:
Hi All,
I'm using asterisk 1.2.15 and call forwarding
On Thu, Feb 15, 2007 at 02:57:59PM +0100, Giorgio Incantalupo wrote:
Hi,
my Asterisk 1.2.9.1 suddenly freezed (stop now did not work!!)
Could you kill the asterisk process directly?
--
Tzafrir Cohen
icq#16849755jabber:[EMAIL PROTECTED]
we have this problem. In our case it was due to the voice mail app; it
was failing to unlink files in memory when creating mp3s. Not sure what
your specific problem might be
Giorgio Incantalupo wrote:
Hi,
my Asterisk 1.2.9.1 suddenly freezed (stop now did not work!!) . I
found the following
On Thursday 15 February 2007 04:00:52 pm Pavel Jezek wrote:
you just post only call forward activation part of dialplan,
but you must also make dialplan part, that reflect, how is set this
callforward mark,
ie. if callforward is set, dial that number, if not, dial peer...
Do you have any
Hello,
I'm trying to figure out how to do something that I hope is pretty easy.
I have a remote phone system (Definity ProLogix) connected to my
Asterisk system via a T1 cable (all onsite). I'd like to get some of
these users on a queue hosted on the Asterisk. I've got it setup so
that it seems
Any news about this ?
Mohamed Farid ,,
Telecommunication Security Section Head ,,
Mediterranean Smart Cards Company ,,
92 Tahreer Street. Dokki / Cairo / Egypt
Website: www.mscc.com.eg http://www.mscc.com.eg/
Email : [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]
Phone : +2 02
Hi,
I use two hint-extensions to monitor my two ISDN-Lines:
exten = 10,hint,Zap/10
exten = 11,hint,Zap/11
My Snom subscribed to the hints and then one line gets busy i have a LED
assigned to the line, that flashes til the call is up and then stay on
til the call is over. So far so good.
If a
Apologies in advance as this is not directly Asterisk related, however I
thought I might be able to leverage the experience of particiapants on
this listserv for some advice.
I have a client who is utilizing Talkswith PBX appliances, which have no
native call monitoring/call recording
Ok thank you a lot!!!
On 2/15/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Il Neofita [EMAIL PROTECTED]
Date: Thu, 15 Feb 2007 03:37:14 -0500
But I tought that hangup was suppose to close the call, however, is not
the
case and a really did not catch why.
Now I see where the confusion comes
Brian Capouch wrote:
Stephen Bosch wrote:
And use a different Wiki engine! Augh! (Mediawiki, anyone?)
Who runs voip-info.org?
I'll bet if you volunteered to take it over, the folks who run it would
gladly let you have it
And I'd further bet they'd gladly let you run whichever Wiki
Check out CallRex, they list Talkswitch as a supported product (also
Asterisk):
http://www.telrex.com/callrex.htm
I've seen it being used with Cisco phones on a hosted Covad environment
and it is pretty neat.
(I have no affiliation with them whatsoever).
Cory Andrews wrote:
Apologies
Hi again,
[EMAIL PROTECTED] wrote:
Hi,
I observed that too. I already got that 2.6.x kernel and it is there actually.
Though Patrick has
put a 2.6.x kernel in /extra, I am still thinking if I will upgrade my existing
kernel in this
laptop. The maintainer of slackware did not made the 2.6.x
Hi Tzafrir,
it was the only solution. I had to kill Asterisk and restart it. I've
got many PBX installed but this is the first time it happened. I've
searched for some opened file limit in linux but found nothing and
ulimit says unlimited.
Giorgio Incantalupo
Tzafrir Cohen wrote:
On Thu,
Hi,
I have the following situation: At a branch , there is a Cisco Call Manager with users all having
Cisco phones. Now I put down a Asterisk 1.2.12 box at the branch, which talks H323 via chan_oh323
to the CCM. So calls go from the CCM, go H323 to the local Asterisk box, then I take it via
Sorry, I sent that message to the wrong list. Tho if you know the
answer, please don't let that stop you from emailing it to me :).
On Thu, 2007-02-15 at 08:21 -0700,
[EMAIL PROTECTED] wrote:
Date: Thu, 15 Feb 2007 08:54:43 -0500
From: Matthew Rubenstein [EMAIL PROTECTED]
Subject:
if you can't use asterisk for recording ;-)
you can try zoom-int callrec, this works listening on switch span port
to record calls...
but it's not free app
Cory Andrews wrote:
Apologies in advance as this is not directly Asterisk related, however I
thought I might be able to leverage the
Hello all .I had one question that, Is it possible to pause a audio
file with out passing any escape digits.
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To UNSUBSCRIBE or update options visit:
Shouldn't you be putting your information in the music-on-hold, rather
than the queue announcement?
Matt wrote:
I am not aware of one.Why would you want your queue announcement
interupted? When we had our Nortel, I found that feature annoying
because people would be transfered to the
We also have not managed to find a solution. Personally, I dunno why the
agents want to stop wrap. I could see what administratively you might want
them to. But for some reason our agents actually wanted to.Anyway, I
created a button that says Wrap Cancel. It does nothing but play a
Can anyone share their experience on the maximum number of calls a given
asterisk box/asterisk software can handle?
I see the asterisk business edition can handle up to 240 simultaneously
with appropriate licensing, but that doesn't seem to be many at all.
For now, I plan to use the stable
So, after reading this, I wonder if anyone has 1.4 and MySQL working...
Is there a non-standard version I can download?
more /usr/src/asterisk-1.4.0/doc/mysql.txt
MYSQL LICENSING UPDATE
==
We were recently contacted by MySQL and informed that the MySQL client
libraries are
On Tuesday 13 February 2007 11:30 am, James Fromm wrote:
Does anyone have a solution to allow an agent to selectively end his
wrap-up time? We define a wrap-up time of 60 seconds to allow our
agents to finish their notes from a call. In some cases, the full 60
seconds is not needed and our
John C. Wolosuk Jr. wrote:
in theory, a gigabit interface can move 1048576Kbit/sec - now if i
generously allocate 96Kbit/sec for every G.711 call, the network
transport can handle, again in theory, 10922 simultaneous calls. would
it be wrong to expect performance near this mark for the
in theory, a gigabit interface can move 1048576Kbit/sec - now if i
generously allocate 96Kbit/sec for every G.711 call, the network
transport can handle, again in theory, 10922 simultaneous calls. would
it be wrong to expect performance near this mark for the asterisk software?
10922 on any
On Thu, 15 Feb 2007, Giorgio Incantalupo wrote:
Hi Tzafrir,
it was the only solution. I had to kill Asterisk and restart it. I've got
many PBX installed but this is the first time it happened. I've searched for
some opened file limit in linux but found nothing and ulimit says
unlimited.
John C. Wolosuk Jr. wrote:
Can anyone share their experience on the maximum number of calls a given
asterisk box/asterisk software can handle?
I see the asterisk business edition can handle up to 240 simultaneously
with appropriate licensing, but that doesn't seem to be many at all.
For now,
I would like to connect a Legacy PBX (Avaya IP Office 406) to an
Asterisk Server.
The Avaya has 3 CO Ports available. I thought buying a TDM30B card
with 3 FXS ports to connect the * to the Avaya CO Ports.
Is this the right approach? Does any one have experience with such a
configuration?
On Thu, 2007-02-15 at 03:13 +0100, Vincent Delporte wrote:
Hello
If someone had the opportunity of trying those two analog cards, how do
they compare? Digium's sells for $150 while OpenVox's sells for $95.
OpenVox makes cheap knockoffs but they are virtually identical to the
Hello,
Am Donnerstag, den 15.02.2007, 10:55 +0800 schrieb Leo Ann Boon:
1. The smallest mini-ITX case I found that accepts a PCI card is the
Travla C138: If you used a mini-ITX with a Digium TDM400P, do you know
if it fits? I didn't find its width, and apparently, the C138 will not
On Thu, 2007-02-15 at 16:11 +0100, Tobias Wolf wrote:
Hi,
I use two hint-extensions to monitor my two ISDN-Lines:
exten = 10,hint,Zap/10
exten = 11,hint,Zap/11
My Snom subscribed to the hints and then one line gets busy i have a LED
assigned to the line, that flashes til the call is up
If the system is running away then I'd suggest looking deeper into it -
is it opening a file and never closing it again, etc. Hard to track down
unless you have a good knowlege of what's running, etc.
If you think it might be asterisk itself, then check which files it has open.
lsof -p `ps h
If the system is running away then I'd suggest looking deeper into it
- is it opening a file and never closing it again, etc. Hard to track
down unless you have a good knowlege of what's running, etc.
lsof -p `ps h -C asterisk -o pid | head -1` | grep -Fc '/dev/zap/timer'
120
You have to open
Daniel Kocher wrote:
I would like to connect a Legacy PBX (Avaya IP Office 406) to an
Asterisk Server.
I would hardly consider the IP office a legacy PBX
Unless, that is, you consider anything other than Asterisk legacy
IP office is current competition for Asterisk, as is Call Manager
You
On 2/15/07, Angel Heart [EMAIL PROTECTED] wrote:
cud any one help me figuring out the problem... When the agent in a queue is
engaged, it cannot accept anymore calls, below is the script;
Angel,
Check your queues.conf, specifically the joinempty parameter.
See below the relevant part in
On Thu, Feb 15, 2007 at 04:47:56PM +0100, Giorgio Incantalupo wrote:
Hi Tzafrir,
it was the only solution. I had to kill Asterisk and restart it. I've
got many PBX installed but this is the first time it happened. I've
searched for some opened file limit in linux but found nothing and
I would like to connect a Legacy PBX (Avaya IP Office 406) to an
Asterisk Server.
The Avaya has 3 CO Ports available. I thought buying a TDM30B card
with 3 FXS ports to connect the * to the Avaya CO Ports.
Is this the right approach? Does any one have experience with such a
configuration?
From: Pavel Jezek [EMAIL PROTECTED]
Date: Thu, 15 Feb 2007 16:51:24 +0100
if you can't use asterisk for recording ;-)
Cory didn't say that:-) Theoretically you can set up Asterisk in between
Talkswitch and end points, map Talkswitch agents with Asterisk agents, then
use Asterisk to
Daniel Kocher wrote:
I would like to connect a Legacy PBX (Avaya IP Office 406) to an
Asterisk Server.
The Avaya has 3 CO Ports available. I thought buying a TDM30B card
with 3 FXS ports to connect the * to the Avaya CO Ports.
Is this the right approach? Does any one have experience with such
Carlos Chavez wrote:
On Thu, 2007-02-15 at 03:13 +0100, Vincent Delporte wrote:
Hello
If someone had the opportunity of trying those two analog cards, how do they
compare? Digium's sells for $150 while OpenVox's sells for $95.
OpenVox makes cheap knockoffs but they are
I have a handful of 7912's connected to my asterisk 1.2.14 server. (6 to
be exact).
I get the X on the display sometimes for loosing registration.
I have the config file for the 7912's
SipRegInterval: 60
and asterisk is the default.
; maxexpirey=3600
;defaultexpirey=120
I've not changed
I would like to connect a Legacy PBX (Avaya IP Office 406) to an
Asterisk Server.
The Avaya has 3 CO Ports available. I thought buying a TDM30B card
with 3 FXS ports to connect the * to the Avaya CO Ports.
Is this the right approach? Does any one have experience with such a
configuration?
Is something like this possible?
Answer
Playback (what extension to pause)
Get input --- how do I do that?
PauseQueueMember (input from user)
Playback (agent paused)
Hangup
Eventually I found it:
The Read Application
http://www.asteriskguru.com/tutorials/read.html
Or
I'm looking at setting up an asterisk box dedicated to SIP-H323 conversion
(a 3rd party is currently converting the protocols for us).
1. Is it worthwhile to split this functionality onto a second server? Or
should we let the ast pbx handle the conversion? (we have a couple hundred
active
On Feb 15, 2007, at 3:17 AM, Wireless wrote:
- Original Message -
From: Nic Bellamy [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, February 13, 2007 8:40 PM
Subject: Re: [asterisk-users] The High Performance
We've mostly gotten our Asterisk install working, but there are a couple
glitches I haven't been able to fix.
I'm looking for someone who knows Asterisk, can do some consulting work,
and is in Southern California. Los Angeles or Orange County are ok, but
I'd prefer someone in the Inland
On 2/15/07, Cory Andrews [EMAIL PROTECTED] wrote:
Apologies in advance as this is not directly Asterisk related, however I
thought I might be able to leverage the experience of particiapants on
this listserv for some advice.
I have a client who is utilizing Talkswith PBX appliances, which have
I have had a lot of complaints about the time it takes to setup a call.
I have timed it and it is almost five seconds before it even starts
ringing. The SIP device sends the request almost instantly but the
channel is taking a long time to pickup and dial. It wouldn't be so bad
but they hear
From: Jordan Novak [EMAIL PROTECTED]
Date: Thu, 15 Feb 2007 13:45:39 -0600
I have had a lot of complaints about the time it takes to setup a call.
I have timed it and it is almost five seconds before it even starts
ringing. The SIP device sends the request almost instantly but the
channel is
do your sip phones dial after a timeout? If the timeout is set to
around 5 seconds you may have a dialplan issue on your sip hardware.
Shane
On 2/15/07, Jordan Novak [EMAIL PROTECTED] wrote:
I have had a lot of complaints about the time it takes to setup a call. I
have timed it and it is
I only say this because nobody in our office knew how to use the
checkmark on snom phones to initiate a call, they always just waited
for the phone to initiate the call for them :)
On 2/15/07, Shane Spencer [EMAIL PROTECTED] wrote:
do your sip phones dial after a timeout? If the timeout is set
We use Linksys/Sipura phones, and do mass provisioning via tftp and
http.
There is no need for a compiler for the SPA-841, 941, 942, 3000, or
2000 phones at least; I don't have direct experience with others. We
feed a raw XML configuration file to the phone via a cgi-bin script
which receives the
Daniel Kocher wrote:
I would like to connect a Legacy PBX (Avaya IP Office 406) to an
Asterisk Server.
The Avaya has 3 CO Ports available. I thought buying a TDM30B card
with 3 FXS ports to connect the * to the Avaya CO Ports.
Is this the right approach? Does any one have experience with such a
Andrew Kohlsmith wrote:
On Thursday 15 February 2007 6:51 am, Steve Underwood wrote:
It looks like octasic have started supplying their echo canceller as
host software for zaptel now. I expect either canceller would work with
the Sangoma cards, as they currently sit in the zaptel framework
Wireless wrote:
Does anyone know if the HPEC will work on a Sangoma A200 / 2 port FXO card?
(I'm assuming so as it still uses Zapel) I've 2 PSTN lines one of which I
cannot get rid of the echo, I've tried a 2GHz machine as apposed to my
normal P3 650MHz and this made no difference. Would the
Jordan said the SIP device sends the request almost instantly so
it's not the SIP phone's fault. The channel bank probably takes 1-2
seconds to pick up and wait/check for dial tone, 1-2 second dialing,
and the telco takes 1-2 second to ring. So the complete PDD is ~5
seconds.
You could try
Hello,
In our investigation of the AddQueueMember vs.
AgentCallbackLogin situation, the major loss with using the
published AddQueueMember replacement is that it assumes each agent
is always using the same phone.
We were not implementing agents this way at all. In fact the _only_
thing we really
I would like to use the * as VoIP Gateway.
Something like that:
A user takes off a phone on a Avaya extension and dials for example 8
to reach the CO Port. Then Asterisk answers and sends a dial tone. The
user dails a numer and Asterisk is doing the rest! (Sending the call
to an SIP or IAX
Help!
I'm (still) having issues with Asterisk Queues.
I want to implement a queue so that callers get the 'all our staff are
busy at the moment, your call has been placed in a queue and will be
answered by the first available operator. You may press 1 at any time
to leave a voicemail'
I remember seeing some way to allow unknown clients to register in Asterisk,
but can no longer find any reference to such. Pointers?
Yuan Liu
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Hi all,
I am trying to implement native format (ulaw) voice prompts and music on
hold. Different documentation has different file extensions. Does Asterisk
recognise them all? So far I have .ulaw .ul .pcm . Which should I use so
Asterisk recognises them as native uLaw files
Audiocodes blatently violates the GPL... dont use their gear.
On 11 Feb 2007 19:11:51 -, [EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
I have 2 identical AudioCodes MP-112s. They have the same config except for
the SIP usernames/passwords and the device IP. The configs in extension.conf
and
I tried that. It didn't work :(
On 2/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
On Tuesday 13 February 2007 11:30 am, James Fromm wrote:
Does anyone have a solution to allow an agent to selectively end his
wrap-up time? We define a wrap-up time of 60 seconds to allow our
agents to
The conference bridge runs Ulaw codec by default. If you let people connect
with GSM or other codecs, Asterisk will use CPU power to convert audio
between codecs ... What about alaw channels is there any transcoding work
being done there?
___
You seem to start asterisk with safe_asterisk. That script
starts asterisk on a console of its own. Maybe it wa done to
allow the use of colors.
If you want a plain 'asterisk' to run with colors, try the patch in
http://bugs.digium.com/view.php?id=9048
Hi, Tzafrir ... The
With the call forward button on the phone? ;)
PaulH
Stefan
it didnt help :( Is there is other way to implement call forwarding?
___
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All of our SIP phones dial instantly when the users finished dialing.
We can do this because we have no ambiguous extension lengths. i.e. no
_XXX and _ and we don't use the . pattern match.
Shane Spencer wrote:
I only say this because nobody in our office knew how to use the
checkmark on
Daniel Kocher wrote:
Daniel Kocher wrote:
I would like to connect a Legacy PBX (Avaya IP Office 406) to an
Asterisk Server.
The Avaya has 3 CO Ports available. I thought buying a TDM30B card
with 3 FXS ports to connect the * to the Avaya CO Ports.
Is this the right approach? Does any one have
Hello everyone,
Now that astlinux-trunk has been coming along very nicely, I thought
I would try to add support for hard realtime capabilities to AstLinux.
If everything works (and there are no problems with zaptel), with a
little tweaking this should improve the audio quality on systems with
How do you fake echo for testing purposes then?
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+1-917-207-3420 Mb
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Nic
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