On Wed, 1 Dec 2004, Brian C. Fertig wrote:
> But now in this instance it drops them into voice mail. Is there a way
> to have them punch in there phone number so they can keep there space in
> the
> system? Like if they are #20 in queue when they left their # for call
> back
> that when they g
[moved to asterisk-users]
On Wed, 1 Dec 2004, Chris A. Icide wrote:
> currently asterisk requires that you have one D channel per PRI, and that D
> channel must be channel 24.
>
> Is it possible to support one D channel for multiple spans?
>
> It seems that you would need a bonding definition.
Hi,
> -Original Message-
> I still need to configure it, but need to know how I would go
> about assigning my server and certain devices (which will be
> connected via the internet, no local hardware involved) us
> telephone numbers.
>
>
>
> Also, with Asterisk, if I understand corr
If I am not mistaken, I believe the dial command is omitted if you do not
have a sound card configured on your system (loaded module).
-michael
On 12/2/04 1:07 AM, "Matt Hess" <[EMAIL PROTECTED]> wrote:
> Does cvs tag v1-0 not have a dial command? I do not seem to have one..
>> dial
> No such co
Hi folks,
thanks for your help with my last question re: japanese FXO. It doesnt sound
very compatible so I will use a SIP FXO gateway then.
Untill I find one, im just trying to get my 2 cisco SIP phones talking to my
* server. just as a learning experience for now. heres what I have so far:
2
James H. Thompson wrote:
Commpartners (who provides hosting for voip-info.org) is doing a network
upgrade tonight.
AH That would explain it :) Just happens to be when I want to work how
to config * :(
Oh well will just have to wait.
To bad we could not mirror the wiki around the world. I am su
Hey guys, I just installed Asterisk, which I find somewhat
confusing.
I still need to configure it, but need to know how I would
go about assigning my server and certain devices (which will be connected via
the internet, no local hardware involved) us telephone numbers.
Also, with A
Commpartners (who provides hosting for voip-info.org) is doing
a network upgrade tonight.
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Luki
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Wednesday, December 01, 2004 8:53
On Thu, 02 Dec 2004 17:31:44 +1100, David Uzzell
<[EMAIL PROTECTED]> wrote:
> Has the wiki died or is it just my routing to the wiki from Australia?
>
> I have not been able to connect to it for the last hour or more :(
>
> David
> ___
> Asterisk-Users
There have been prior discussion on this on the list -- just google for
"starband site:lists.digium.com" and you should find it. IIRC, there
has been some sporadic success, but overwhelmingly, satellite-based VOIP
connections are not considered feasible.
> -Original Message-
> From: Feder
Does cvs tag v1-0 not have a dial command? I do not seem to have one..
> dial
No such command 'dial' (type 'help' for help)
Henry Devito wrote:
Ok try this
Login into console
Set verbose 15
Dial (extension of VoiceMailMain app)
Dial mailbox number
Dial password
Hangup
Does it still die?
See my exa
I have SBC in Illinois, and dialtone service is only $5-ish. Add
another $6 or so for caller-id delivery. Do you need it for outgoing
calls? If yes, getting DSL on top of one of the lines and then using
VOIP termination is likely cheaper than getting their bundles. Avoid
call-waiting and three-
> Dead for me too.. I am in the US..
Dead here too and I am in LA, next door to it (last hop
commp-2.border17.lax.pnap.net).
Maybe there are doing an upgrade... I recall their DB server was spitting
out "too many connection" errors yesterday...
--Luki
__
Dead for me too.. I am in the US..
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: "David Uzzell" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Thursday, December 02, 2004 12:41 AM
Subject: Re: SV: [
Thorben G. Jensen wrote:
It dead from Denmark too :-(
Well I think yes it is! :(
All I get on traceroute from me!
traceroute to www.voip-info.org (66.151.54.101), 30 hops max, 38 byte
packets
1 192.168.2.1 (192.168.2.1) 0.377 ms 0.366 ms 0.189 ms
2 rns02-kent-syd.comindico.com.au (203.194.3
It dead from Denmark too :-(
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af David Uzzell
Sendt: 2. december 2004 07:32
Til: Asterisk Users Mailing List - Non-Commercial Discussion
Emne: [Asterisk-Users] www.voip-info.org
Has the wiki died or is it jus
Federico Gonzalez wrote:
I have an Asterisk with one local Cisco ATA and one remote Cisco ATA
connected to the Asterisk, the remore connection is a satellite link
with an 900ms delay.
This is the same delay I have here. Never less than 900, sometimes over
1500 ms.
Check
http://lists.digium.com/pi
Has the wiki died or is it just my routing to the wiki from Australia?
I have not been able to connect to it for the last hour or more :(
David
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To UN
Hi all,
Is it possible to download Danish, German and French audio
files for Asterisk somewhere, or does everybody just record them?
Thank you in advance
Thorben
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> I am having problems with Broadvoice incomming calls.
> Did anybody who use broadvoice as a provider
> experienced and problems today?
Bart,
here's me saying two days ago no problems... but... Broadvoice has been
flakey today, indeed. Seems like their LAX server has been going up and
down.
I
dear list,
I am using asterisk version 0.7.1,and asterisk-oh323 version
0.5.10.Both are installed successfully,and I can see the configuration
file in asterisk directory.But when I changed the file h323.conf
,asterisk doesnot start..it gives the error
booting.asterisk:relocation error:
/usr/lib
How can I even tell if there's been a compilation problem?
The last line in any make-based build will tell you of an error if one
occurred.
At this point, type "asterisk" and then "asterisk -r" at a command
line. The first one starts Asterisk and detaches it as a daemon
(background process)
Alan Ingleby wrote:
Hi all. I'm a very computer literate person, but an a bit of a noob
when it comes to linux etc.
I've a got a brand new PC.
I've stuck in a TDM400P, with one FXO port.
I've installed Fedora Core 3.
Everything good so far. Now for Asterisk.
GO to a shell then...
md src
cd src
I
Hi all. I'm a very computer literate person, but an a bit of a noob
when it comes to linux etc.
I've a got a brand new PC.
I've stuck in a TDM400P, with one FXO port.
I've installed Fedora Core 3.
Everything good so far. Now for Asterisk.
GO to a shell then...
md src
cd src
I'm now at /home/a
Tim,
You may see description of new 1.3.4 firmware at polycom.com (check -
http://www.polycom.com/common/pw_cmp_updateDocKeywords/0,1687,3641,00.pdf
) released in October.
Though, it was proven over time that troubles with a SIP phone like "not
hearing" one side or the other is NAT related prob
On Wed, 1 Dec 2004 22:34:54 -0600, Brent Clements wrote:
>So how does Asterisk fit in to the scheme of things when connecting to a
>IAX/Sip termination provider?
>
>Does a IAX/Sip termination provider just provide incoming call termination
>or does it do inbound and outbound?
>
>-Brent
It depends
Steven Critchfield wrote:
On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote:
Steven Critchfield wrote:
On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote:
So, isn't the issue he/I are chasing after essentially 'why is cpu consumption
jumping 30% (or 100%) every ten seconds when zaptel is
So how does Asterisk fit in to the scheme of things when connecting to a
IAX/Sip termination provider?
Does a IAX/Sip termination provider just provide incoming call termination
or does it do inbound and outbound?
-Brent
- Original Message -
From: "Sean Cook" <[EMAIL PROTECTED]>
To: "As
Simplest explanation:
IAX and SIP are protocols that allow devices to talk VOIP. When you
connect to a proxy device via either protocol, the proxy is said to
terminate those protocols.
Brent Clements wrote:
I think I have idea what IAX and SIP termination means, but can
someone explain this to
I think I have idea what IAX and SIP termination
means, but can someone explain this to me?
For instance, how and why would I use someone like
iax.cc?
Thanks,
Brent
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Robert Barnes wrote:
On Wed, 1 Dec 2004 20:13:16 +1000, Robert Barnes
<[EMAIL PROTECTED]> wrote:
This has happenned to me now too - so I doubt that your hardware is faulty...
Oops - wcfxs was renamed to wctdm some time ago... Working again now.
RAB
I still use wcfxs (comes with my debian
Do you use more than one satellite?
Brandon Patterson
If you have a one-way dealy of 900ms you have a *very* bad satellite system.
We run many IP phones across out satellite links. Our *round-trip* delay is
<600ms, one-way including VoIP component latencies is no more than 350ms.
It also looks
LA seems to be down. Switch to DCA or MIA and you'll probably be OK.
Bruce Komito
High Sierra Networks, Inc.
www.servers-r-us.com
(775) 236-5815
On Wed, 1 Dec 2004, Bartosz Wegrzyn - asterisk wrote:
> Hi,
>
> I am having problems with Broadvoice incomming calls.
> Did anybody who use broadvoic
If you have a one-way dealy of 900ms you have a *very* bad satellite system.
We run many IP phones across out satellite links. Our *round-trip* delay is
<600ms, one-way including VoIP component latencies is no more than 350ms.
It also looks like the remote is not completely registering with *
Hi,
I am having problems with Broadvoice incomming calls.
Did anybody who use broadvoice as a provider experienced and problems today?
I want to make sure if this is my equipment or the service.
Thanks
Bart,
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I have this same problem with Cisco Phones. Someone will try and call an
extension and asterisk will say "Can't create SIP channel" and I
immediately
do "sip show peers" and the phone in question still has an IP listed.
If I know for fact that my 10 phones will never change IP addresses, how
can
Hello Asterisk users,
Well standard modem will not work. You need to get a card designed to work
with a PBX check out
http://www.voip-info.org/wiki-Asterisk+Hardware For a list of compatible
asterisk hardware. Just for testing you can get X100p compatible cards for
around $30...The reas
Hello Asterisk users,
Well standard modem will not work. You need to get a card designed to work
with a PBX check out
http://www.voip-info.org/wiki-Asterisk+Hardware For a list of compatible
asterisk hardware. Just for testing you can get X100p compatible cards for
around $30...The reas
I saw them too and they looked pretty good. I assume you can buy the minutes
and use them for whatever you want.
Only issue I have with them at the moment is that their ping times don't
seem great from where I will be setting up our initial server.
I may setup an account with them for testing p
Good point ;)
--
Start Your Own ISP!
http://www.YourOwnISP.com
- Original Message -
From: "Linus Surguy" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Wednesday, December 01, 2004 7:43 AM
Subject: Re: [Asterisk-Users] Experienc
Not yet. It's under development.
Greg
Alex Brecher wrote:
Is there anything open source out there that has the same or better feature
set than Asterisk PBX Manager ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter
Sent: Tuesday, November 30, 20
Not sure.
The only issue I was having was the devices themselves rebooting at very
predictable times. Not just one device either, all 4 that I have on my
network would reboot all at once.
Only one way to find out though. Disable lisa and see if it helps?
On Wed, 2004-12-01 at 18:35 -0600, Brian
Ok try this
Login into console
Set verbose 15
Dial (extension of VoiceMailMain app)
Dial mailbox number
Dial password
Hangup
Does it still die?
See my example below
asterisk*CLI> dial 777
-- Executing VoiceMailMain("OSS/dsp", "") in new stack
<< Console call has been answered >>
-- P
Rodney Acosta Coya wrote:
[113]
type=friend
context=test
username=113
fromuser=113
callerid=113
usecallerid=yes
hidecallerid=no
host=172.16.4.226
Dec 1 13:25:49 NOTICE[1112980400]: chan_sip.c:7519 handle_request:
Registration from '' failed for '172.16.4.226'
Mick San,
I wish you luck. I am in Japan also and like others in the list, what
you have to worry less is about the configs.. get some samples in the wiki
site and go for it. If you need some help, feel free to ask.
But, be prepared for some headache to make it working well with NTT FXO
lines
On Tue, Nov 30, 2004 at 07:43:11PM -0500, Andy Reinke arranged a set of bits
into the following:
>I have some Cisco 7960's and want to use them with SCCP - I have gotten it
>working with a few different firmware versions but all seem a little
>flakey. I know that SCCP is not as solid
I have 15 ea Cisco 7905 phones. I have them working with my Asterisk
system, but I cannot get the blind transfer button on the phone to work. I
can use the transfer button to work, but I don't know how to release the
call after I transfer it.
When I try and blind transfer, it just hangs up the p
Any idea if 1.34 fixes the problems with the phones being up for long
periods of time and weird call problems (I cannot hear remote caller,
but they can hear me) ?
-Tim
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul Hales
Sent: Wednesday, December 0
On Wed, 01 Dec 2004 16:01:16 -0800, Mike Benoit <[EMAIL PROTECTED]> wrote:
> Don't run "LISa" on the same network as any SPA-2000 or SPA-3000. (maybe
> even any Sipura device?)
I have a problem with mine locking up, but not while talking. When it
sits idle for a period of time I come back to it an
I am trying to get transparent call routing working, but this
extensions.conf not working correctly:
exten => 200,1,Dial(zap/3/200);infinite loop!!!
old config:
telco t1 pbx t1 win-ivr
new config:
telco ==2t1== asterisk t1 pbx t1 win-ivr
my dream:
on port 1,2
Hi folks,
Im totally new to * but I went ahead and told my boss that it was the way
to go for our new telephone system :) now I have a test box and two cisco
phones and a brand new modem card.
Im having plenty of trouble with learning all the config stuff but ill leave
that for another day. ie:
AMP but you already knew that.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Alex
Brecher
Sent: Wednesday, December 01, 2004 6:28 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] Asterisk PBX Manager
Is the
Don't run "LISa" on the same network as any SPA-2000 or SPA-3000. (maybe
even any Sipura device?)
On Wed, 2004-12-01 at 14:57 -0700, Bryan Mannos wrote:
> What was the fix?
> ___
> Asterisk-Users mailing list
> [EMAIL PROTECTED]
> http://lists.digium.com
Any idea if 1.34 makes Daylight Savings work for us people in Australia?
PaulH
-Original Message-
From: Andrei (MPI) [mailto:[EMAIL PROTECTED]
Sent: Thursday, 2 December 2004 9:30 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Polycom IP500
Hey All,
I use cvsup to update my Asterisk servers. I have the following in the sup
file...
*default host=cvs.digium.com
*default base=/usr/src
*default release=cvs tag=v1-0
*default delete use-rel-suffix
asterisk
;libpri
;zaptel
asterisk-addons
asterisk-sounds
After it downloads the files, I d
Hello,
I am trying a setup that is the following:
SIP Phone (Zultys) --> Asterisk ---> H.323 GK (Cisco) > PSTN
Any calls from H.323 GW through GK goes to PSTN, no problem.
SIP Phone registers to Asterisk, and calling to Voice Mail, No Problem.
SIP Phone to PSTN, rings normally, on the PSTN
Hi all,
We've got a number of users connected in a configuration which is basically:
(a)
SIP Phones -> Asterisk -> IAX -> Our Asterisk -> Cisco AS 5xxx (SIP) -> PSTN
We also have users in a configuration:
(b)
SIP Phones -> Asterisk -> IAX -> Our Asterisk -> Digium E1 -> PSTN
The second server in bo
Hi Chris,
First of all, you need to configure ftp or tftp and watch syslog
closely - what the phone is looking for at boot time. You would need to
put config files into (t)ftp directory, named according to MAC address
of you phone. XML and Web is really weird - they do not even share same
conf
Is there anything open source out there that has the same or better feature
set than Asterisk PBX Manager ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Clive Carter
Sent: Tuesday, November 30, 2004 4:33 PM
To: [EMAIL PROTECTED]
Subject: [Asterisk-User
I have a working Asterix system in a very simple configuration, just
using a VOIPTalk account to make and receive calls with two extensions.
Everything works OK but
I have got a strange problem when making a call to a PSTN number form
Asterisk. When I dial out from a SIP phone it rings no p
I will be in Paris this December and would like to know if anyone would
like to meet to have an informal Asterisk get together. Please e-mail me
directly off-list your availability Dec 19, 20, 21 and 22 and we will
select whichever day is most convenient for the most people. Please try
to let
[EMAIL PROTECTED] is believed to have said:
>Telnet uses TCP, SIP listens on UDP, use netstat instead.
>
>
>B
Bob,
thanks for the hint! I should have imagined that SIP could not use a tcp
protocol...
Aldo
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Two * servers: *a and *b.
Outside call comes in *b, and is automatically routed to *a. Someone on
a sip phone connected to *a then decides to transfer the call to someone
on a sip phone connected to *b. The transfer works.
At this point, is *a still in the converstation? Or is * smart enough
On Wed, 2004-12-01 at 14:51 -0800, Ed Rubright wrote:
> On Wed, 2004-12-01 at 05:45, Michael Graves wrote:
>
>
>
> > Compound this with the fact that small FXO adapters tend to suck. I've
> > tried X100p clones and the Sipura SPA-3000. Neither were acceptable. I
> > recently installed a TDM11. A
With Fedora Core 2, I notice that my Digium cards are on irq 17 and 19.
Are these irqs cascaded through irq 2? Will these devices have
interrupt conflicts at irq 2?
Thanks,
--
Michael Welter
Introspect Telephony Corp.
Denver, Colorado US
+1.303.674.2575
[EMAIL PROTECTED]
www.introspect.com
___
On Wed, 2004-12-01 at 05:45, Michael Graves wrote:
> Compound this with the fact that small FXO adapters tend to suck. I've
> tried X100p clones and the Sipura SPA-3000. Neither were acceptable. I
> recently installed a TDM11. After only a few days I'm still happy with
> it. The relative volume
> But now in this instance it drops them into voice mail.
No. In "MY" instance, they goto voicemail. YOUR instance can do whatever you
want.
> Could this by chance be done with a AGI of some sort?
You bet. Since the queue.conf drops them into a context, you can have
that context do whatever
Test completed successfully..
test dialplan:
exten => 555,1,Answer
exten => 555,2,Wait(2)
exten => 555,3,Playback(digits/0)
exten => 555,4,Playback(digits/1)
exten => 555,5,Playback(digits/2)
exten => 555,6,Playback(digits/3)
exten => 555,7,Playback(digits/4)
exten => 555,8,Playback(digits/5)
exten
i am not sure about the timer but normally you would want a delay that
is less than 500ms for good audio quality.
smadi
Federico Gonzalez wrote:
Hello,
I have an Asterisk with one local Cisco ATA and one remote Cisco ATA
connected to the Asterisk, the remore connection is a satellite link
with an
Hello,
I have an Asterisk with one local Cisco ATA and one remote Cisco ATA
connected to the Asterisk, the remore connection is a satellite link
with an 900ms delay. I can make calls from the remote site to the
local site, but when try to call from local to remote it doesn't work.
The Asterisk tim
Try to play a number sound file by using the Playback application, I think
the voicemail uses the same app to play the digits. See if that works.
exten => 500,1,Playback(digits/3)
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Mat
hi;
if i have a conference session with two user, can we ring a third user
and make him/her join the conference session?
moe smadi
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Some more info would be nice
What software version? Did you setup the Micronet as peer, proxy or as
gateway? Did you setup the user and password?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Federico
Gonzalez
Sent: Wednesday, December 01, 2004 7:03 P
Steve Kennedy wrote:
On Wed, Dec 01, 2004 at 02:53:50PM -0500, Kanuri, Seshu (Company IT) wrote:
Tell me which one can get me access to the LinkSys Linux using SSH? Does
Satori has this feature? I am not so concerned with Voice Shaping and
QOS at this time, but more interested in converting this i
On Wed, 1 Dec 2004 12:37:13 -0800 (PST), Ben Kirkpatrick wrote:
> Do you find it difficult to manage four LD providers?
> Can you show me part of your LD Macro and how it's used?
>
> I'm toying with two LD providers now, but don't have failover setup.
>Just using each one for what they are b
Hey All,
First Time Writing.
I'm trying to set up my IP500 phones to register SIP with *. I input all
the (I assume) correct data in to the fields on the Web Interface. And I
get no notification that the phone is even attempting to register, no
failed messages etc. I have read that the Web interf
Hello,
I connected a Micronet SP5014 2FXS + 2FXO gateway to the asterisk, the
problem is i can make call but can't receive calls. If i make a "sip
show peers" it shows the micronet is not connected to the asterisk.
Does anybody knows how to configure the micronet and asterisk to solve
this proble
But now in this instance it drops them into voice mail. Is there a way
to have them punch in there phone number so they can keep there space in
the
system? Like if they are #20 in queue when they left their # for call
back
that when they get to number 4 or 5 that they would be called back and
p
What was the fix?
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On Wed, 2004-12-01 at 14:47 -0600, Rich Adamson wrote:
> >Looking at the Changlog for 2.6.9, it would appear a fair amount of
> work has been down in the pci stuff and the interrupt support areas.
> Since that seems to be an issue that keeps rearing its head with the
> digium analog cards, maybe th
yup.. that's something I thought of as well.. and it's all there..
funny thing is.. I can start asterisk.. login just fine to voice mail..
I try again right away and I get that error that I had sent earlier and
get cutoff..
Henry Devito wrote:
-Original Message-
From: [EMAIL PROTECTED
I'm setting up an asterisk server, used as a gateway to regular phone
lines. I've got a TDM400P card with FXO modules, but I'm only using one
to test.
When I make outgoing calls, occassionally it seems like the incoming
audio is switched off. It will work fine for several calls, and then
for
Maybe try something like:
exten => s,1,SetCIDName(${EXTEN}: ${CALLERIDNAME})
exten => s,2,Dial(${ARG2},20,rt)
exten => s,3,Goto(s-${DIALSTATUS},1)
or am i mixed up...?
On Wed, 1 Dec 2004 14:35:28 -0700, Tim Thompson <[EMAIL PROTECTED]> wrote:
>
> I have Aastra pt480e phones and would like to pr
You can setup recording by default. This is how I have mine setup. I
don't believe the way app_queue is now you can have the agent press
something to have it start recording.
.o---o.
Brian Fertig
Network Engineer
Planet Telecom, Inc.
Tampa,
I too get this message just about any time that I hangup when
listening to a prompt. In my case I notice it during Background().
Any one else running into this?
On Wed, 03 Nov 2004 12:07:10 +0100, Joerg Beck
<[EMAIL PROTECTED]> wrote:
> I got a strange error message on the CLI, saying:
>
> War
Check out queues.conf
;
; A context may be specified, in which if the user types a SINGLE
; digit extension while they are in the queue, they will be taken out
; of the queue and sent to that extension in this context.
;
context = cytel-queuewaitnomore
In our case, if a person presses 0 then they
http://www.voip-info.org/wiki-SCCP-HOWTO2
-
Message: 1
Date: Wed, 1 Dec 2004
08:12:18 -0500
From: Rodney Acosta Coya
<[EMAIL PROTECTED]>
Subject: RE:
[Asterisk-Users] cisco 7902g
To: 'Asterisk Users Mailing
List - Non-Commercial Discussion'
<[EMAIL PROTECTED]>
Message-
I have Aastra pt480e phones and would like to present the caller the CID
info about the extension being called.
I have tried the following with no avail:
exten => 311,1,Macro(stdexten,311,${TIM})
exten => s,1,SetCIDName(${ARG2})
exten => s,2,Dial(${ARG2},20,rt)
exten => s,3,Goto(s-${DIALSTATUS}
. Does anyone know how I can build a dial peer
> with
> a destination pattern that will strip off all of the extra stuff and just
> process the 4 digit did?
[*]
Look at the Cisco website and search for digit-strip command and/or the
translation rules. These are part of the dial peer enhancement
We want to set up monitoring of calls going into our queue. We want to
know if there was a way to initiate it, by having the agent who picks up
the call dial a number to initiate the recording.
Ruben T. Santos
Director of Network Operations
Brand X Networks
(866) 487-324
hi
Today I´ve installed, apache 2.0.52, mysql-4.1.7, asterisk-perl-0.08
and ASTCC prepaid card aplication from CVS, so now I have access to
the astcc-admin.cgi from web server
http://asterisk/cgi-bin/astcc-admin/astcc-admin.cgi and I´ve been able
to create the database from "Configure" menu but
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Matt Hess
> Sent: Wednesday, December 01, 2004 11:47 AM
> To: [EMAIL PROTECTED]
> Subject: [Asterisk-Users] voicemail cuts off / hangs up
>
> I'm having a problem with voicemail where
Rich Adamson wrote:
Looking at the Changlog for 2.6.9, it would appear a fair amount of
work has been down in the pci stuff and the interrupt support areas.
Since that seems to be an issue that keeps rearing its head with the
digium analog cards, maybe there is something 'fixed' in that area.
Not b
Is it possible to have customers to be in queue and have a prompt that asks
them if they want to leave a phone number so when there time is
up they will get a call back so they can speak with the CSR?
.o---o.
Brian Fertig
Network Engineer
Pl
It did not fix my spandsp/TxFax problems, however :-(
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At 02:07 AM 11/28/04, you wrote:
I hope this is an appropriate question for the list..
I am looking for a VOIP termination provider who can offer the following:
-Flat Rate DID's in lots of areas
-GOOD customer service/support with quick response times
-Toll Free DID's at a reasonable rate
-Reliable
> On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote:
> > Steven Critchfield wrote:
> > > On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote:
> > >
> > >
> > >>So, isn't the issue he/I are chasing after essentially 'why is cpu
> > >>consumption
> > >>jumping 30% (or 100%) every ten secon
On Wed, 2004-12-01 at 13:03 -0700, Michael Welter wrote:
> Steven Critchfield wrote:
> > On Wed, 2004-12-01 at 13:36 -0600, Rich Adamson wrote:
> >
> >
> >>So, isn't the issue he/I are chasing after essentially 'why is cpu
> >>consumption
> >>jumping 30% (or 100%) every ten seconds when zaptel i
Kido,
I start to compile openh323 1.3.15 as recomend in the
asterisk-oh323-7.0 README but I get an error:
h323ep.cxx: In member function `virtual BOOL H323EndPoint::IsLocalAddress(const
PIPSocket::Address&) const':
h323ep.cxx:2397: no matching function for call to `PIPSocket::Address::
IsR
> Tell me which one can get me access to the LinkSys Linux using SSH?
> Does Satori has this feature? I am not so concerned with Voice Shaping
yah it has ssh, but if all you want is more of a linux box
then suggest you look at the openwrt.org, they have a writeable
file system on the flash & not t
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