Guys.
Anyway had problems with G BT 100 or 101 volume? Seems the volume is too
loud and when talking it makes the voice cut off due to saturation.
Anyway to reduce the input voice volume on the phones?
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what about CAS 3 Bit?
does * support it?
thanks,
Paradise Dove
On 4/8/05, Steve Underwood <[EMAIL PROTECTED]> wrote:
> David Hajek wrote:
>
> > Hi,
> >
> > is it possible to use Asterisk with T110P and CAS (channel associated
> > signalling)?
>
> There are hundreds of CAS protocols. Quite a few
Digium have told us that a problem that we are having (with accuracy of
zap interface as measured using zttest) may be due to the fact that we
have a Xeon processor with hyperthreading and have suggested turning H/T
off.
Anyone else experienced a problem like this? No too keen about turning
H
Hello,
I don't think this is an * issue, it was more likely the telecom you
connect to having some kind of issues. If they hacked your * box, they
would not likely call you and tell you. Unless they have some reason to
show you down and they made the call themselves.
l.
In data Thu, 14 Apr 2
also add snom-190 and snom-360 to your list
PolyCom 500 and 600 have the same feature too.
On 4/15/05, Brian Leyton <[EMAIL PROTECTED]> wrote:
> Or Flash Operator Panel. http://www.asternic.org
>
> Brian Leyton
> IT Manager
> Commercial Petroleum Equipment
>
>
> > -Original Message-
>
Hi List,
I've spent hours researching on this topic, found tons of info, so far it
doesn't work yet.
Here's the scenario
Asterisk box connected to a router (DMZ enabled to Asterisk) and trying to
send calls to an outside provider.
My SIP phones (outside * NAT) are able to register with no proble
On Thu, 2005-04-14 at 21:35 -0700, Michael D Schelin wrote:
> If your in Los Angeles Call me I've got 130,000 numbers with caller ID
> from my ss7 network. Trust me there's a whole lot more to it than what
> he just said.
I didnt go into what was required for puc certification as that varies
for
Many of these scripts are based on the from which for the most part on
this list is whoever posts a reply. When you reply it goes to the list
address but the from is infact that of the author of the current
message which causes vacation/spam/.. filters to go crazy.
For example I just got a mail bo
Daniel Bruce Lynes wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 14 April 2005 19:25, Ronald Wiplinger wrote:
you miss the point.
I have setup the white listing system that I get rid of the spam, and it
works very good.
The message says it very clear what to do, if you cannot
Matthew Boehm wrote:
(for us at least, subtract the price of the TE110P cause all our T1's
come to us on DS3s, and we already have DS3 routers in place and paid for.)
I'm confused: why would you terminate a T1 from a provider using a
channel bank, rather than directly into an Asterisk server?
On Thu, Apr 14, 2005 at 10:39:10AM -0800, Daniel Bruce Lynes wrote:
> Also note that ztdummy requires that you have a UHCI root hub for your USB.
> OHCI and EHCI will not suffice.
Unless you use kernel 2.6. In kernel 2.6 ztdummy does not need any
strange USB tricks and simply uses the kernel's
Groups for each trunk and check the dial plan groupcount and cycle
thru the trunks
or keep a list of trunks in a DB and just loop thru that first call
route 1 second route 2 etc.
I'll give it some more thought when I wake up but I think you would
have to track concurrent channels per trunk to bala
Rotate or make sure the line you are dialling out on isn't in use before
you try and use it? Lookup setgroup/checkgroup on the Wiki if it's the
latter -
http://www.voip-info.org/wiki-Asterisk+cmd+SetGroup
Allows you to create logical groups for just about anything, check
whether the groups are
If your in Los Angeles Call me I've got 130,000 numbers with caller ID
from my ss7 network. Trust me there's a whole lot more to it than what
he just said.
Mike
trixter http://www.0xdecafbad.com wrote:
On Thu, 2005-04-14 at 19:37 -0700, snacktime wrote:
I knew xo/level3 were
On 4/14/05, jltaylor <[EMAIL PROTECTED]> wrote:
> Any ideas on how to rotate (evenly distribute) outbound calls over a number
> of 'trunks' or contexts?
Funny I was just looking at the following thread, looks like it might work.
http://lists.digium.com/pipermail/asterisk-users/2003-May/011484.htm
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Thursday 14 April 2005 19:25, Ronald Wiplinger wrote:
> you miss the point.
> I have setup the white listing system that I get rid of the spam, and it
> works very good.
> The message says it very clear what to do, if you cannot / do not want
> to
On Thu, Apr 14, 2005 at 03:28:08PM +0100, Julien Levi wrote:
> I currently run an asterisk server with cvs from May 2004. I'm planning
> to upgrade to the latest stable version but want to segregate a test
> version first. I know I can do this by editing the install_prefix field
> in the makefil
Any ideas on how to rotate (evenly distribute) outbound calls over a number
of 'trunks' or contexts?
James Taylor
MetroTel
3505 Summerhill Road
Suite 11
Texarkana, Tx 75503
903-793-1956
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ht
Michael Crozier wrote:
On Monday 11 April 2005 12:04 pm, Michael Crozier wrote:
The zaptel drivers are proving quite unstable with this combination. If
I attempt to rmmod the zap drivers, the machine hangs and is unresponsive
to keyboard input, ping, or sysreq. Additionally, I attempted to bypass
Adam Robins wrote:
When an outside callers hits my system, I play them a welcome message
and ask that they enter an extension. If the extension is invalid, it
tells them so, and asks them to try again. The relevant logic for this
is:
[extensions]
exten => _2XXX,Dial(SIP/${EXTEN})
;
exten => i,1,P
Kanuri, Seshu (Company IT) wrote:
Does anyone know how Polycom 500s will be able to update their time.
My setup for a time sync with Public domain Time servers is not
successful.
We set the NTP server and timezone using ISC DHCPd.
option ntp-servers 172.16.7.1;
option time-offset
On Thu, 2005-04-14 at 19:37 -0700, snacktime wrote:
> I knew xo/level3 were clecs, and that the numbers came from nanpa, but
> I didn't know the requirements for getting numbers. So theoretically
> anyone with some type of switch can go to nanpa, get a CIC and some
> numbers, and then get someone
So, any way i can resolve this problem?
At 10:55 AM 4/15/2005, you wrote:
On 4/14/05, Kong <[EMAIL PROTECTED]> wrote:
> Hi,
> i found a case here, i really don't know is it a bug or something else.
>
> i have like 200 ip phones connected to my * server, (ATA's and softphones).
> and i had it regist
I do understand how Dial works, but Zap/4 hungup immediately before
Zap/3 is answered. Zap/4 doesn't even rings.
Sorry I didn't mention about this earlier,
206 & 221 are extensions connected to a Panasonic KX-TD1232 pbx.
I have two extensions 211 & 212 connected to my TDM400p FXO ports.
If I want to drop inbound calls from certain regions (like calls to
our 800 number from areas where we don't do business), is getting the
area code list from nanpa and querying against that the best way?
Chris
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On 4/14/05, Kong <[EMAIL PROTECTED]> wrote:
> Hi,
> i found a case here, i really don't know is it a bug or something else.
>
> i have like 200 ip phones connected to my * server, (ATA's and softphones).
> and i had it register to SIP service (FWD), so, when my internet connection
> is down, * is
On Thu, Apr 14, 2005 at 11:01:33AM -0400, Chris wrote:
> Christopher Dittrich,
>
> There is a new voicemail in mailbox 202:
>
> From: "SMITH KENT D"
>
> Length: 0:48 seconds
>
> Date: Wednesday, April 13, 2005 at 12:54:33 PM
>
> Dial *98 to access your voicemail by phone
Mike:
I know this sounds patronizing, but do you have the SIP image files?
If so, what version? Per the Asterisk wiki page on the 7960/7940s, you
may need to upgrade incrementally.
Additionally, make sure you have the correct files in the root
directory of your tftp server (for linux, this is pro
On 4/14/05, trixter http://www.0xdecafbad.com <[EMAIL PROTECTED]> wrote:
> On Thu, 2005-04-14 at 19:18 -0700, snacktime wrote:
> > I'm curious about how a company goes about getting nationwide (US)
> > DID's for resale. No I'm not wanting to be a reseller, just curious.
> >
> > For example compani
C F wrote:
Why do ppl do this?
and no I will *not* follow the link.
Good,
can we come to the subject, please?
How can I set it up?
I guess more people would like to know how to get Setgroup / Checkgroup
to work. Obviously it is not doing as I expected it.
bye
Ronald
-- Forwarded messa
On Thu, 2005-04-14 at 19:18 -0700, snacktime wrote:
> I'm curious about how a company goes about getting nationwide (US)
> DID's for resale. No I'm not wanting to be a reseller, just curious.
>
> For example companies like XO or Level3. Do they go out and cut deals
> and/or exchange traffic with
Jon Gabrielson wrote:
And the stupid thing is that it is trivial to set up a script
to autorespond to these things. So assuming it is a
valid MX (which is easy to check for without harrassing
anyone), a spammer has an easier time responding
than a nonspammer.
Jon.
Jon,
you miss the point.
I
I'm Andrew.
On April 14, 2005 10:01 pm, Qiao Yuansong wrote:
> My asterisk box and sip phone are not behind a nat, the sip phone and
> asterisk box are connected by LAN, so the delay is not caused by network
> congestion, and furthermore, there is no delay from sip to pstn.
>
> [sip phone]--LA
Hi,
i found a case here, i really don't know is it a bug or something else.
i have like 200 ip phones connected to my * server, (ATA's and softphones).
and i had it register to SIP service (FWD), so, when my internet connection
is down, * is not able to register itself to FWD, never mind that, bu
Done all that, still doesn’t work.
I do have outgoing and incoming, just can’t
get the incoming to come through the livevoip context.
Thanks
Chris Mason
US Number: (646)722-0001 US
Fax (815)301-9759
Skype: netconcepts
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On
I'm curious about how a company goes about getting nationwide (US)
DID's for resale. No I'm not wanting to be a reseller, just curious.
For example companies like XO or Level3. Do they go out and cut deals
and/or exchange traffic with CLEC's/RBOC's in every region where they
1) need DID's, and 2
Thanks for your reply :).
My asterisk box and sip phone are not behind a nat, the sip phone and asterisk box are connected by LAN, so the delay is not caused by network congestion, and furthermore, there is no delay from sip to pstn.
[sip phone]--LAN--[Asterisk with X100P]--[
A softphone such as SJPhone or X-Lite.
-Kerry
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kumara
Jayaweera
Sent: Thursday, April 14, 2005 6:47 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] How do I make a call thru *PBX
Hi all,
I
Hi all,
I am new to*. but I have a working *PBX in my Linux box. Please, tell me how
can I make a call only with sound card. ( without any other hardware)
Kumara
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I can't get the 7960 to reconfigure and work. I am a newbie to voip. I
went through the list and read some other comments about the 7960 and
unlocking it. It is a used 7960 that came with CallManager. I need to
have SIP. I first reset the phone to factory defaults then I changed
the TFTP server
On April 21st, at 7:30 PM, Mark Spencer and John "Maddog" Hall[1] will
be joining the Toronto Asterisk Users' Group[2], the Toronto Linux
Users' Group[3] and the Ontario Asterisk and VoIP Enthusiasts Group[4]
for an informal chat about "Asterisk and The Open Source Telephony
Revolution".
If you ar
And the stupid thing is that it is trivial to set up a script
to autorespond to these things. So assuming it is a
valid MX (which is easy to check for without harrassing
anyone), a spammer has an easier time responding
than a nonspammer.
Jon.
On Thursday 14 April 2005 06:14 pm, C F wrote:
>
you can use a BSTU in a 424 also. It's kind of hit and miss with any of
the RSTU cards.
- Original Message -
From: "Brian Leyton" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Thursday, April 14, 2005 6:38 PM
Subject: RE: [Asterisk-Users] Tos
Kanuri, Seshu (Company IT) wrote:
Does anyone know how Polycom 500s will be able to update their time. My
setup for a time sync with Public domain Time servers is not successful.
Seshu
We had a user with a Sonic Wall Firewall who needed to set the snpt
server to the IP address of his firewal
If you still have problems after trying his config details, we can
record a copy of the sound (you'll probably have it in the end of
voicemail), and analyse for frequency/cadence using WaveLab.
Feel free to call me any time:
(03) 4555770 x 1
or just add a line:
exten => 818,1,Dial(IAX2/[EMAIL PR
Title: Wall Mount PC Case
http://www.spinserver.com/
We have mounted the smallest ones all sorts of places
- even under boardroom tables.
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Wiley
SilerSent: Friday, 15 April 2005 2:35 AMTo: Asterisk Users
Mailing List - Non-Com
Hello,
I have an AGI script that runs a menu at two levels of a tree.
If I call the extension from a voip phone with g711, the menu works fine and
accepts DTMF no probs.
Then, when I Call from a DID, it sends call using SIP and g729 to¨* box.
The IVR also starts running, but no DTMF is deteced.
Hello,
I have a Sipura SPA1001 which for some reason cannot respond to
Asterisk's prompts, for example the voicemail password or "Enter an
Extension". Asterisk seems to not recieve the tones. I can dial my
office PBX and answer prompts through the SPA1001, and Asterisk responds
when I use my
Henry Devito wrote:
> You don't want to use RSTU2's unless you want echo. RSTU3's
> are a little better but BSTU's are what you need.
Will I have the same echo problem with RSTU2 on a DK-424?
I don't think I have another choice other than T1. I've been testing with
an x100p connected to a sta
In your zapata.conf set
usecallerid=no callwaitingcallerid=no and immediate=yes. Remove the
Wait(0) and start your first priority with answer.
- Original Message -
From:
Scott
Wolfe
To: Asterisk-Users@lists.digium.com
Sent: Thursday, April 14, 2005 5:25
PM
Why do ppl do this?
and no I will *not* follow the link.
-- Forwarded message --
From: Ronald Wiplinger <[EMAIL PROTECTED]>
Date: Apr 14, 2005 7:05 PM
Subject: Please confirm your message
To: [EMAIL PROTECTED]
This message was created automatically by mail delivery software (TMDA
There are so many fax information available, so that I am getting confused.
What I hope I can get to work:
Any extension should be able to receive fax, whereby via faxdetect the
fax should be sent to the email address as mentioned in voicemail.conf
Which packages should I install?
How would be th
oh yeah, by the way, They do make a 2 port analog card for the CTX 100 if
you only need a couple lines.
- Original Message -
From: "Brian Leyton" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Thursday, April 14, 2005 4:45 PM
Subject: RE: [Aste
Bruno:
Are you getting any errors or warnings at the CLI?
-Andy
On 4/14/05, Bruno Quintas <[EMAIL PROTECTED]> wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi all, i installed [EMAIL PROTECTED] v0.8, very clean install (great
> piece of software!).
> I have successfully configure
You don't want to use RSTU2's unless you want echo. RSTU3's are a little
better but BSTU's are what you need.
- Original Message -
From: "Brian Leyton" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
Sent: Thursday, April 14, 2005 4:45 PM
Subject: RE
I am currently working on the coding to provide D tone disconnect. There is
a work around I am using right now at a few customer sites. I have done
this several times, interconnecting Toshiba to Toshiba PBX's and Toshiba to
other pbx's.
- Original Message -
From: "Brian Leyton" <[EMA
the feature you are talking about is still not commited to stable. at
the moment it is only availabe in CVS HEAD. You can try to download
the patch and apply it, however I did not succeed in applying it to
1.0.7 so I had to use HEAD.
On 4/14/05, Shaun Tierney <[EMAIL PROTECTED]> wrote:
> Hello all
On April 14, 2005 06:34 pm, chawki hammoud wrote:
> I previously posted about the huge latency introduced
> by iax2. It is a problem introduced by the codec. in
> iax2.conf, i disllowed=all and allow=gsm and the RTT
> is the same as I do ping shell command. When i change
> from gsm to ulaw or alaw,
Another option (which I think is just as good) is to use the patches
available for chan_capi and set it up to receive faxes.
Just search the list for chan_capi and fax.
Yours,
Andrew
On 15/04/2005, at 5:11 AM, Michiel van Baak wrote:
On 10:39, Thu 14 Apr 05, Kib Eki wrote:
Hi,
I found the
I want to signal BUSY condition to a bristuffed HFC-S ISDN line.
However:
"exten => s,1,Busy" has no effect,
"exten => s,1,Playtones(Busy)" is not audable over unanswered line (I
live in the Netherlands...)
So I currently do:
+ exten => s,1,Answer
+ exten => s,2,Playtones(Busy)
+ exten => s,3
Since you are not setting up an actual PBX in the true sense of the
term, the hunting has to be done by the telco if you want to take more
than one call at a time on the same number. Cincinnati Bell here locally
calls their standard SMB phone service "Centrex"; Sprint would probably
call theirs
Hello all! I posted a message a while back about a problem I was having
in December. I was unable to send arguments to the macro in the dial
command. I was told back then to use ^ as the delimiter between the macro
name and the arguments and that I had to upgrade to a newer version of
Asterisk.
Matt, can I assume from your silence that you concurr with my thinking that
realtime is in fact broken, or is it that I am using it incorrectly?
- Original Message -
From: "Rod Bacon" <[EMAIL PROTECTED]>
To: "Matthew Boehm" <[EMAIL PROTECTED]>; "Asterisk Users Mailing List -
Non-Commerc
Hi:
I previously posted about the huge latency introduced
by iax2. It is a problem introduced by the codec. in
iax2.conf, i disllowed=all and allow=gsm and the RTT
is the same as I do ping shell command. When i change
from gsm to ulaw or alaw, then i have the huge RTT and
high jitter and evntually
Is there a way to have an FXS
port not ring but just pick up? Here is what I am doing.
I have Mitel 200SX plugged into
one of FXS ports on my TDM400 so that my Mitel users can make calls out via
VoIP. Currently when I dial that Mitel extension from the Mitel, it rings the
port on TDM400 a
Hello,
I've been working a lot with asterisk lately. I've
had a LOT of positive experience with various SIP
clients (grandstream hardware phones & ATAs, X-Lite,
SJPhone, etc...), and I've had no trouble getting
asterisk behind a NAT to talk SIP to clients across
the internet behind another NAT usi
On April 14, 2005 05:48 pm, Michael Di Martino wrote:
> Call come in over the pots lines however Outbound goes out thru the VOIP
> provider.
> However looking at the configs I cannot figure out what controls how
> call are sent out.
> In other words where in the config files does it determine that
Just guessing but look for something like this.
This is from an old config of mine...
[trunkint] ; ; International long distance through
trunk ; exten => _9011.,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
exten => _9011.,2,Congestion
[trunkld] ; ; Long distance context accessed
through t
Or Flash Operator Panel. http://www.asternic.org
Brian Leyton
IT Manager
Commercial Petroleum Equipment
> -Original Message-
> From: Henry Devito [mailto:[EMAIL PROTECTED]
> Sent: Thursday, April 14, 2005 11:57 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject:
Henry Devito wrote:
>
> I have done this with CTX's you need analog cards in your
> ctx. and fxo cards in the * servers. Email me off list I am
> a Toshiba Dealer.
Good point - I missed the fact that he doesn't have analog station ports.
Those cards aren't too terribly expensive though. I mi
Good luck! So far an upstream switching issue is really sounding most
probable.
You should grab the CDRs from that time frame to verify no 3-way and
then tell
BT when it happened exactly. Might not tell you squat but worth a try.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[
Should have in iax.conf.
;This registers you to them
register=:@64.34.59.73
;THis context serves to ID incoming, if you ahve a DID
it shoudl come here
[livevoip]
type=user
secret=mySecret
host=64.34.59.73
callerid="Livevoip IAX
User"
context=livevoip-in
;This one is your outgoing...
I have just
inherited a Asterisk box which is configured as follows.
10 internal
Sip Phones
3 Pots
Lines
1 voip
provider (SIP)
Call come in
over the pots lines however Outbound goes out thru the VOIP
provider.
However
looking at the configs I cannot figure out what controls how c
We're running Cisco 7940 / 7960 with only 1 line available. Conferencing
is possible, however the agents would have to a) know how to do that and
b) make a positive action to do so when there is no need to do so. We do
not, and never have, make conference calls with agents.
Julian.
Damon Estep
In data Thu, 14 Apr 2005 10:39:38 -0700, Richard Lyman <[EMAIL PROTECTED]>
ha scritto:
lenz wrote:
Hello,
you have to enter "/var/log-xcast/queue_log_live" as the file and "DPS"
as the queue (select it from the drop-down box) for the demo to find
*snipped
one thing i noticed, i did a copy pa
In data Thu, 14 Apr 2005 10:39:38 -0700, Richard Lyman <[EMAIL PROTECTED]>
ha scritto:
lenz wrote:
Hello,
you have to enter "/var/log-xcast/queue_log_live" as the file and "DPS"
as the queue (select it from the drop-down box) for the demo to find
*snipped
one thing i noticed, i did a copy pa
Wiley Siler wrote:
The call bridge is the onoy thing that seems suspect.
Can an internal user do a 3-way to an external site? If so, this could
In theory, but no-one has ever done so.
explain how someone else could hear a conversation but it would mean
that the agent did something really dumb like
I have a newly provisioned livevoip account which registers
OK but the incoming calls are not being authenticated as livevoip and only work
as the guest context:
[livevoip]
type=user
secret=mySecret
host=64.34.59.73
callerid="Livevoip IAX User"
context=livevoip-in
[guest]
t
Hello I have a Dell Poweredge / Dual 3.2 GHZ XEON / 2GB ram running
asterisk
It's configured using realtime-extensions / sipfriends / iaxfriends (to
a local mysql daemon), 80% of all calls are IAX <-> SIP calls with no
codec transcoding and no jitterbuffering, and 20% of all calls are IAX
<-> IAX
Title: Bizarre - VM just stopped for one user
The other users work fine but this one does not.
Here is from the CLI on calling the user….
-- AGI Script Executing Application: (Dial) Options: (SIP/1000|120|tr)
-- Called 1000
-- SIP/1000-1bc2 is ringing
-- Got SIP respon
I'm using v 1.0.7 of * and was wondering if it's possible to route via
called number using SIP without any patches or agi?
Thanks,
Kevin
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Stephen wrote:
>
> I need to connect two sites with asterisk integrated with Toshiba ctx100.
> Currently both have 4co and 8 extensions.
>
> Site A starts the extension with 8xx and Site B starts the extension with
1xx.
>
> How can I integrate two sites with asterisk that is
> transparent to us
On 4/14/05, Titux <[EMAIL PROTECTED]> wrote:
> Hi,
> I'm having problems with the compile of supertone lib.
> Can anybody giveme a list of reqs. for correctly compile and install
> MFCR2 package?
> I read the softswitch page but didn't have success.
> The problem is, I can compile spandsp without p
I've actually ran in to something similar. Though luckily it hasn't
happened in a while.
I've gotten at least 4 complaints of an in-progress call all of a sudden
being able to hear (but not speak to) another conversation that is in
progress. I've never been able to track it down, but I'm running 1
Hi,
I'm having problems with the compile of supertone lib.
Can anybody giveme a list of reqs. for correctly compile and install
MFCR2 package?
I read the softswitch page but didn't have success.
The problem is, I can compile spandsp without problems.
In fact my [EMAIL PROTECTED] 0.8 comes with span
If the time is off by exactly x hours, check the *timezone* in ipmid.cfg.
On 4/14/05, Kanuri, Seshu (Company IT) <[EMAIL PROTECTED]> wrote:
>
>
> Does anyone know how Polycom 500s will be able to update their time. My
> setup for a time sync with Public domain Time servers is not successful.
Does anyone know of a provider that sells custom or vanity local DIDs?
The only one I know of that comes close to such an offering is
Sunrocket that lets you pick a number from a list of available
numbers. Does anyone else offer anything like this?
___
Greg, the Diva Server cards are around $900 for a single BRI and $2500
for a Quad. The Adran unit can be had for less than $400 used.
Brian
On Wed, 2005-04-13 at 16:30, Gregory Wiktor - ADCom Corp. wrote:
> Brian,
> I am looking into the diva isdn cards for around 200 at this point. I
> had the
PLEASE TRIM YOUR POSTS, it takes less than 30 seconds!
On April 14, 2005 04:27 pm, Damon Estep wrote:
> The user stated that the line is PRI ISDN, not likely to be a physical
> short as that would take the digital line out, not produce crosstalk,
> had to be a switching issues with the telco or *,
I'm able to call park just fine, I can pick up a call just fine. but
if nobody picks up the call and Asterisk tries to send the call back
to te extension that parks it, it fails.
HELP!
001 -- Executing NoOp("SIP/0004f201e463-a-7650", "EXTEN=3599
CONTEXT=toll-access") in new stack
002 -- Exec
Use Gotoif instead of Goto. Check Gotoif usage.
This will give you enough features to fork the calls after the extension
is re-entered
Seshu
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam
Robins
Sent: Thursday, April 14, 2005 3:11 PM
To: Asterisk U
Exactly. Time to check the CDR
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Thursday, April 14, 2005 1:25 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Overheard conversation. Co
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Patrick May
> Sent: Thursday, April 14, 2005 1:59 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Overheard conversation. Comments pleas
Don't use it multiple channels, just do like this
exten => s,1,Dial(channel1,1)
exten => s,2,SetVar(C1S=${DIALSTATUS})
exten => s,3,Dial(channel2,1)
exten => s,4,SetVar(C2S=${DIALSTATUS})
exten => ;keep on going for all your channels
then do like this:
exten => s,1,GotoIf($[${C1S} = BUSY]?500)
exte
What kind of voip phone? Is it possible the user conferenced 3 calls
inadvertently? Easy to do on some multi call appearance phones (snom in
particular)
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of Asterisk
> Sent: Thursday, April
Do you have a firewall turned?___
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Thx
Increased the busycount and seems fine so far.
Moises Silva wrote:
>what is the configuration you have in zapata.conf ???
>try using callprogess=no and busydetect=no, and if you have trouble to
>hangup the calls, then try callprogress=no and busydetect=yes ...
>i had troubles when both p
I recall seeing though, that they may not show up as lit buttons,
meaning you may not neccesarily be able to see status.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Henry
Devito
Sent: Thursday, April 14, 2005 2:57 PM
To: Asterisk Users Mailing L
Of course, you can have the telco put on rj-11 jacks and just run them
to the tdm. Plus consider an analog fxs port for fax, etc.
Greg
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Bruce Lynes
Sent: Thursday, April 14, 2005 2:49 PM
To: 'Asteris
Hello,
Does anyone know of a script that can take a voicemail, and deliver it
to a mobile phone or pbx vm system?
For example, I have a panasonic dbs voicemail, which has vmwi and
telephone based vm navigation. I want to accept vm's on asterisk, then
later forward the vm to the pbx, by for exampl
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