As said by Moises the "reason" in the orginate events is not working in version
1.2.12.1. Does anyobdy know in what version it is working, preferably one later
than 1.2.12.1 and not prior to 1.2.5? Is it working correctly in asterisk 1.4?
___
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As said by Moises the "reason" in the orginate events is not working in version
1.2.12.1. Does anyobdy know in what version it is working, preferably one later
than 1.2.12.1 and not prior to 1.2.5? Is it working correctly in asterisk 1.4?
___
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Hi all,
I have a little question I was not able to find answered on the Net::
How can I measure ASR (Answer-Seize Ratio) ?
Which is its definition ?
In other terms, it is said to be a ratio between "answered calls" and
"total attempted calls".
(cfr:http://en.wikipedia.org/wiki/Least_cost_routing)
Tom Vile ha scritto:
I am trying to use the Gotoiftime CMD to not allow calls to be placed
between the hours of 12am-5am, except if you know the PIN number to
dial out and if the call is for 911.
What is the best way to implement this solutions?
I have the gotoiftime like so:
exten => s,1,G
On Wed, Nov 22, 2006 at 05:49:07PM +0200, Eugeniy Khvastunov wrote:
> Hi All!!
>
> Prompt how to record all calls passing through certain span?
Send them to a diferent context, in which you mark them as "need
recording" using a channel variable.
--
Tzafrir Cohen
icq#168497
On Wed, Nov 22, 2006 at 05:02:42PM -0300, Ard wrote:
> Hi,
>Can somebody in the list tell me why sometimes when I do the TOP
> command I see more than one asterisk process ?
>
> Sometimes it appears and desappears again...
Which kernel do you use? 2.4 by any chance? If so: are all of them wi
On Wed, Nov 22, 2006 at 04:51:24PM -0300, Lincoln Zuljewic Silva wrote:
> Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four
> X100P - FXS). Both boards are recognized by the operating system as showed
> above:
>
> :08:00.0 Communication controller: Tiger Jet Network
On Thu, Nov 23, 2006 at 12:03:15AM +0100, Timothy Parez wrote:
> Hi,
>
> I've been able to
> make
> make install
> the Zaptel drivers (1.2).
>
> I'm using a b410p so I executed the following command
> make b410p. I tried this on multiple machines, but it always failes:
First off, the b410p drive
Hello Users.
I'm Now doing on Asterisk Manager for My knowledge Growth, Can anybody
explan me on Asterisk Manager settings...
in manager.conf
[general]
enabled =yes
port = 5038
bindaddr = 192.168.2.75
displayconnects = yes
[hyperion]
secret = hyperion
permit=192.168.2.76/255.255
Hi
where can i buy that Book
Ram
On 11/22/06, Patrick <[EMAIL PROTECTED]> wrote:
On Wed, 2006-11-22 at 15:45 +0530, ram wrote:
[snip]
> Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring
> switchtype
> Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring
> s
48VDC is a long time telco standard - and has become the Power over Ethernet
standard.
Keep in mind that 'electricity' isn't the measure - it's power. Power is
not synonymous with voltage.
The formula V=IR (voltage equals current time resistance) points to a higher
voltage allowing lower curre
I am trying to use the Gotoiftime CMD to not allow calls to be placed
between the hours of 12am-5am, except if you know the PIN number to dial out
and if the call is for 911.
What is the best way to implement this solutions?
I have the gotoiftime like so:
exten => s,1,GotoIfTime(5:00-11:59|mon-
I'm using my own lookup script, published on http://muware.com/asterisk
It does use various web-services in attempting to find a name -- this
obviously only works in the US environment. I maintain the incoming
numbers using a very crude php-script, but phpMyAdmin works wonders.
Vincent Delpo
I am a newbie. Just got my Asterisk working and I love it.
I want to do the following, believe it should be possible, but can't work
out how:
When I get an incoming call, I want to answer and just send ringing to the
calling channel.
Then I want to call the destination channel, send a message as
In a recent interview someone from Digum indicated that the G722 wideband codec
was being worked into Asterisk. This will make Asterisk compatible with
Polycom's new HDVoice products
like the IP650 phone. This is very interesting, potentially exciting, but it
brings up certain questions.
Who w
The amount of electricity used is constant. When run from 48v DC power they
draw less current (mA)...power (w) is constant. Devices run from 5 V DC likely
draw more current (mA). Power =
Voltage x Current.
Michael
--Original Message Text---
From: Zeeshan Zakaria
Date: Wed, 22 Nov 2006 21:29:46
Check with your telco on the "Glare" setting. They probably have Glare set
to "CO Yields" which tells their switch to 'yield' to your switch/pbx when
negotiating which channel to use. We ran into this problem with our PRI and
an older phone system that did not give nearly the amount of insight (
Does it effect the performance/voice quality? Does this also mean that 48VDC
is using less electricity in an office than 5VDC IP Phones?
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Problem: SPA3000 phone does not ring for incoming PSTN call although I
can dial out.
I set up my Sipura with the Voxilla Wizard which is pretty good but
leaves out some important details.
The Voxilla Wizard for Supura SPA3000 gave me a setting for PSTN Tab ->
Dial Plans ->
Dial Plan 8 ()
S
Marcus Franke wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael Welter wrote:
Has anyone tried recording to a ramdisk? To an NFS mount? Was there a
benefit?
RAM disk? Interesting idea, but what to do in case of a server crash
loosing these recorded files?
Or use som
We are looking for a site running Queumetrics in Sydney, Australia.
We have been contacted by a company in Sydney, as a few staff members of a
company that are currently running Queuemetrics would like to see a fully
running installation for training and decision making purposes. Their trial
l
Pavel Jezek wrote:
is possible to control ci$co gateway from asterisk via mgcp? i.e.
asterisk as mgcp call agent?
PJ
I've tested the old Cisco ATA-186 MGCP (firmware 2.16) with Asterisk
<1.2. Works pretty well.
Leo
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FYI, the interval at which the device is checked is 60seconds when OK,
and 10s when not OK.
It can be changed in channels/chan_sip.c. Look for this lines:
#define DEFAULT_FREQ_OK 60 * 1000 /* How often to check
for the host to be up */
#define DEFAULT_FREQ_NOTOK 10 * 1000
Is there in option for a compressed codec? Might be worth a try...
PaulH
On Thu, 2006-11-16 at 13:30 +1100, Lucas Barbuto wrote:
> Hi all,
>
> Originally tried to post this without being subscribed, apologies if
> the list gets this twice.
>
> One of my users has a problem with many of his c
A follow up on my message about my SNOM 200 phones now powering from
my 802.3af Netgear FS108p PoE box.
To follow up for those finding this thread on searches...
I purchased some PowerDSine 6001 units (very cheap on ebay) and they
power the SNOM 200 fine. Some Buffalo units also did this.
So
Currently at our office, if I want someone else to pick up a call, I have
to transfer the call to them. So I'm looking into call parking, which is
ALMOST perfect.
The missing piece of the puzzle: I'm extension 203. I want any call I park
to get parked at extension 2203. I want a call my boss
Hi,
I've been able to
make
make install
the Zaptel drivers (1.2).
I'm using a b410p so I executed the following command
make b410p. I tried this on multiple machines, but it always failes:
[EMAIL PROTECTED]:/usr/src/zaptel-1.2.11# make b410p
[ -f misdn-b410p.tar.bz ] || wget
ftp://ftp.digium.c
Most of our customers have generic names like "Hospital", so I need to
rewrite their caller ID name by looking up the number in a database on the
Asterisk server, and rewriting the name such as "Reading Hospital" so that
we know who's calling.
Any idea if this can be done with Asterisk, and how t
Vincent Delporte wrote:
Hi
Most of our customers have generic names like "Hospital", so I need to
rewrite their caller ID name by looking up the number in a database on
the Asterisk server, and rewriting the name such as "Reading Hospital"
so that we know who's calling.
Any idea if this can
My mission is to get one * box to dial another * box' extensions. I
have set this up previously without any issues by making a simple IAX
trunk/extension pair on the two boxes and create a dial plan with a
prefix like 9|XXX to select an extension on the other box.
My problem is that I now have to
We narrowed this down to when the 'New Call' softkey was used to
initiate the call. When this key was used, the corresponding 'Placed
Calls' entry wouldn't work. Any other method of placing the call does
work.
An upgrade to 1.6.7 fixes the issue.
CP
On Nov 16, 2006, at 4:34 AM, John Marvin w
We had the same problem with WRT54G with no Linksys Linux firmware. At
that time the problem was WRT54G modified the devices IP address, i.e.
Asterisk received the WRT54G IP address instead of device address.
Solution was selecting NAT=yes.
Hope this help
Jorge
tommaso.carrara wrote:
Hi, I'v
I've installed on 6.1 it from ports with ztdummy without an issue. I've
never used zaptel hardware on it though. Had some issues with meetme
and ztdummy but all worked out.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of J. Oquen
On Wed, 2006-11-22 at 12:01 -0700,
[EMAIL PROTECTED] wrote:
> On 11/22/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
> Hi,
>
> Why Aastra phones use more electricity, i.e. 48VDC whereas
> other phones use much less, e.g. Grandstream and Linksys both
> use o
Michael Welter wrote:
When I record to a .wav file, I get gsm encoding. Is there a way to
record using u-law encoding?
The extension for ulaw is .ul
Kevin
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To UN
[EMAIL PROTECTED] wrote:
Hi,
Running Asterisk 1.2.12.1 when dialing a known disconnected number the calls
just rings and rings. We never get the "The number you are trying to reach...".
If we dial the same number from an Asterisk 1.0.11 server again over PRI, we get
the message on the 1st ring
I'm having problems getting my aastra 480i to register with the asterisk
server. I can inititate calls from the phone, but >sip show peers does not
show any IP address registered for this phone. I am probably missing
something stupidly simple. Anyone have an example config to share or
corrections
Hey i said that as per his requirement as an example :) . His requirement is
just around 20 calls . For a moderate server i think sata raid should be
fine ..Heres some result posted by someone for recording calls on ram disk
. http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/118497
Power is nothing to do with voltage (well it is, but not alone), you
need the current too i.e. V * A.
Pylon electricity lines run at very high voltage (several hundred
thousand volts) or the current going down the lines would heat the
cables and you'd lose a lot of power.
48V is just a telco sta
I'm having an issue with call park on my new Linksys 922. It has
soft menu keys for doing call transfer (which I always think is a good
idea because it's amazing how every phone has a different xfer interface
and people always get confused).
However, I can't get a good call park working on it.
On Tue, 21 Nov 2006, Gregory Duchatelet wrote:
> Hi all,
>
>
>
> I have a Diva Server V-BRI-2 card, which support, as written in reference
> guide:
>
> Extended tone processing (human talker detection, generation and detection
> of country-specific tones)
>
>
>
> I would like to detect hu
I would suggest the following
- remove the drivers
- load them manually (zaptel, wcte11xp, wctdm)
Run:
Zttools -> should show unconfigured cards.
Take:
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=1-15,17-31
dchan=16
fxsks=32-35
loadzone = us
defaultzone=us
run:
ztcfg -vv
See what it is
Hi,
Can somebody in the list tell me why sometimes when I do the TOP
command I see more than one asterisk process ?
Sometimes it appears and desappears again...
Thanks,
Ard.
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asterisk-users
This is the scenarios:
1 -
###
/etc/zaptel.conf
span=1,1,0,ccs,hdb3,crc4
bchan=5-19,21-35
dchan=20
fxsks=1-4
loadzone = us
defaultzone=us
###
modprobe wcte11xp
ZT_CHANCONFIG failed on channel 32: No such device or address (6)
FATAL: Error running install command for wcte11xp
2 -
###
/etc/zaptel
On Wed, Nov 22, 2006 at 06:58:13PM +0100, Huib van Wees wrote:
>On 11/22/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
> Why Aastra phones use more electricity, i.e. 48VDC whereas other
> phones use much less, e.g. Grandstream and Linksys both use only
> 5VDC. I first thought i
I think that you are loading the drivers in the wrong order. You can change
the order of loading are first define the E1 followed by the TDM400
Hope this helps,
Henk
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Lincoln
Zuljewic Silva
Sent: woensdag 22 nove
This happens when a call is offered to asterisk on a B-Channel that's
already marked as used, I had the problem with one of my PRI provider,
not hanging up calls but instead giving network congestion when users
hung up...
Trouble was solved at their side...
Regards,
Tristan
Paul Hales a éc
Dumpolid Exeplish wrote:
Hi,
Has anyone installed Asterisk on FreeBSD? i need help/steps on this task
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asterisk-users ma
On Wed, 22 Nov 2006, Dumpolid Exeplish wrote:
Has anyone installed Asterisk on FreeBSD? i need help/steps on this task
Im running Asterisk on FreeBSD, use the port in /usr/ports/net/asterisk
/e
--
http://hostname.nu/~emil
___
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yep. email me offlist. I can help you.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dumpolid
Exeplish
Sent: Wednesday, November 22, 2006 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk On FreeBSD
Hi,
Has anyone
Hello all. I have here a TE110P (configured as E1) and a TDM400P (with four
X100P - FXS). Both boards are recognized by the operating system as showed
above:
:08:00.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device b1d9:0003
Hi,
Has anyone installed Asterisk on FreeBSD? i need help/steps on this task
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Here you go:
[Custom-CLID]
exten => s,1,Answer
exten => s,2,Authenticate(12345)
exten => s,15,Playback(after-the-tone)
exten => s,16,Playback(pls-entr-num-uwish2-call)
exten => s,18,SetVar(CALLFILENAME=${EXTEN:1}-${TIMESTAMP})
exten => s,19,Monitor(wav,${CALLFILENAME},m)
exten => s,20,DISA(no
I'm using a 2811 to transfer 4 digits to another Cisco gateway that
connects to a NEC pbx. Working great when calls are originating from the
Asterisk. When I try to call the Asterisk it is answering the calls, but
not transferring them to the appropriate extension.
Sam
Little Me Childrenswear
http://bugs.digium.com/view.php?id=8405
On Nov 22, 2006, at 9:11 AM, Anthony Rodgers wrote:
Thanks, John - this confirms what we are seeing. 'show hints' output
isn't changing, so it looks like a bug. I'll open one and see what
happens.
A.
On Nov 21, 2006, at 5:44 PM, John Lange wrote:
> Hin
Gregory Duchatelet wrote:
Hi all,
Another question for today, hope an answer for this one…
I have a program talking with asterisk via the AMI. I receive events,
and I would like to insert some events in the dialplan, which could be
catch by my program.
Any idea how to do this ?
Greg
thi
On 11/22/06, Zeeshan Zakaria <[EMAIL PROTECTED]> wrote:
Hi,
Why Aastra phones use more electricity, i.e. 48VDC whereas other phones
use much less, e.g. Grandstream and Linksys both use only 5VDC. I first
thought it was because of PoE, but the ones with 5VDC also run fine on PoE.
What is the dif
What are these?
Nov 22 09:35:23 WARNING[7127]: channel.c:787 channel_find_locked:
Avoided deadlock for '0xf6c06778', 10 retries!
Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_find_locked:
Avoided deadlock for '0xf6c06778', 10 retries!
Nov 22 09:35:24 WARNING[7127]: channel.c:787 channel_fin
Hi,
You can do it using AstDB, just load the database with callerid names and
numbers and then include a lookup on database in all incoming calls, so you
can override whatever you wanted:)
On 11/22/06, Steven <[EMAIL PROTECTED]> wrote:
There are two I can think of.
Hoodahek and asterdex (or
There are two I can think of.
Hoodahek and asterdex (or asteridex)
We used hoodahek at first, but now use asterdex(sp?)
It has a web interface to enter the new names into.
We use it to fixup, corp. cell phones and used to use it for our leagcy PBX
extensions.
--
--
Steven
http://www.glimaso
Thanks, John - this confirms what we are seeing. 'show hints' output
isn't changing, so it looks like a bug. I'll open one and see what
happens.
A.
On Nov 21, 2006, at 5:44 PM, John Lange wrote:
Hints are not working in 1.4b3 period. Snom 360s do not show any status
updates. However, before
I bought the phone in Germany. Except another wlan phone from Siemens which
was not available any more, I did not find any alternatives to it.
-Original Message-
From: Olivier [mailto:[EMAIL PROTECTED]
Sent: Wednesday, November 22, 2006 8:44 AM
To: Asterisk Users Mailing List - Non-Commerc
I've never used Asterisk MGCP, and I've only used MGCP gateway on Cisco
IOS when controlled from Cisco CallManager (with PRI D-channels
backhauled to CallManager).
In terms of making an invalid number dialed via Asterisk to Cisco...
behavior on Cisco side is entirely subject to how you've programm
Hello Everyone,
I just upgraded to the latest beta version and I am running into one
problem. We purchased g729a licenses from digium and they aren't
loading anymore. If I roll back asterisk to 1.2.10 the codecs work
fine. I've downloaded the new 1.4 version of the codec from their
website and
Yeah, doing more testing shows that the speed keys are broken, but dialing
it works!!! Ugg!!!
can you let me know if you get a new firmware? Im going to try and
downgrade...
Thanks!
On 11/22/06, Alban <[EMAIL PROTECTED]> wrote:
Yes, already.
Waiting now for a new firmware...
Regards,
Alban
Eric "ManxPower" Wieling wrote:
Eugeniy Khvastunov wrote:
Hi All!!
Prompt how to record all calls passing through certain span?
Next time I'll have coffee before hitting Reply.
pbx-1*CLI> show application monitor
pbx-1*CLI>
-= Info about application 'Monitor' =-
[Synopsis]
Monitor a chann
is possible to control ci$co gateway from asterisk via mgcp? i.e.
asterisk as mgcp call agent?
PJ
Bas van der Veen wrote:
Scott,
Thanks for the reply. I am experiencing the following with a 2801:
- user mistypes a phone number, so the number becomes non-existent
- asterisk sends the call t
qualify=xxx in sip means, consider peer as OK if delay reply is bellow
xxx (ms)
qualify checks (POKE) is every 60s (and is not configurable in sip.conf)
qualify setting in iax.conf is working differently, this is how
frequently to check peer (and is not possible to set some POKE delay
threshlo
On Wednesday 22 November 2006 08:43 am, Michael Welter wrote:
> Is there a programmatic to to trim the silence from the beginning and
> end of a recording? From a .wav file? From a .ulaw file?
>
> Thanks,
try "man sox" - look for 'silence'
Brett
___
-
[EMAIL PROTECTED] wrote:
Hi
I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by
outbound SIP.
Now i want to detect DTMF-Tone Code coming from the called party to
trigger a signal.
Can this be done with asterisk? I read that the codec with DTMF
detection are ulaw and alaw. But
Eugeniy Khvastunov wrote:
Hi All!!
Prompt how to record all calls passing through certain span?
pbx-1*CLI> show application record
pbx-1*CLI>
-= Info about application 'Record' =-
[Synopsis]
Record to a file
[Description]
Record(filename.format|silence[|maxduration][|options])
Records f
Vicky wrote:
I doubt that . I think qualify=500 means asterisk checks every 500 ms if
the
other extension is available or not . Because when qualify=( value in ms )
is set and you do a sip show peers in console asterisk whos how much
latency
is there between extension and asterisk . If i set q
Hi All!!
Prompt how to record all calls passing through certain span?
---
Thanks...
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-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Michael Welter wrote:
> Has anyone tried recording to a ramdisk? To an NFS mount? Was there a
> benefit?
>
RAM disk? Interesting idea, but what to do in case of a server crash
loosing these recorded files?
You will get very angry customers if you
Dont Use Call Progress Instead Use The M Option In App Dial That Asks
The User To Press A Button To Accept The Call
On 11/21/06, shadowym <[EMAIL PROTECTED]> wrote:
Anyone tried this,
I put in an Asterisk/FreePBX phone system to replace one of those el cheapo
Bizfon analog key systems. The Bi
Sorry, asking too quickly, thats what im looking for :
http://www.voip-info.org/wiki/index.php?page=Asterisk+Manager+API+Action+Eve
nts
Greg
_
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de Gregory
Duchatelet
Envoyé : mercredi 22 novembre 2006 15:33
À : asterisk-
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Vicky wrote:
> Yeh even a
> simple UDMA 5 enabled hard drive can handle 30 calls recording easily .
> Sata hard drives are even better .
>
Hehe, UDMA sounds like EIDE drives.. nice to see they are fast enough,
but I do not recommend those as server h
Is there a programmatic to to trim the silence from the beginning and
end of a recording? From a .wav file? From a .ulaw file?
Thanks,
--
Michael Welter
Telecom Matters Corp.
Denver, Colorado US
+1.303.414.4980
[EMAIL PROTECTED]
www.TelecomMatters.net
_
I had a hell of a time getting AMD to work correctly on 1.4. If I didn't
compile asterisk correctly (I'm not sure how I was doing it incorrectly), it
wouldn't work (and would just stop the dialplan execution). Try recompiling
everything (make clean && make install) and see if that helps. (It's work
Hi all,
Another question for today, hope an answer for this one.
I have a program talking with asterisk via the AMI. I receive events, and I
would like to insert some events in the dialplan, which could be catch by my
program.
Any idea how to do this ?
Greg
___
I know in advance maybe I'm overlooking something stupid,
however I'm really lost and cannot find the solution...
Situation:
- asterisk-1.2.13 on a linux box with no iptables active.
- one iax2 peer defined
- one wildiax phone running on my laptop
the soft phone is configured to connect & regis
Yeh even a
simple UDMA 5 enabled hard drive can handle 30 calls recording easily .
Sata hard drives are even better .
On 22/11/06, Marcus Franke <[EMAIL PROTECTED]> wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
>
> Does anyone have experience with recording mult
Has anyone tried recording to a ramdisk? To an NFS mount? Was there a
benefit?
[EMAIL PROTECTED] wrote:
Hi,
We want to build an Asterisk system that needs to be able to record,
when in a peak situation, a maximum of twenty calls simultaneously. I
could not find any reference to performanc
I doubt that . I think qualify=500 means asterisk checks every 500 ms if the
other extension is available or not . Because when qualify=( value in ms )
is set and you do a sip show peers in console asterisk whos how much latency
is there between extension and asterisk . If i set qualify = no then
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tim Panton
Sent: 21 November 2006 19:16
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Hairping calls and Originating CLI
On 21 Nov 2006, at 10:08, Adrian Marsh wro
I am using asterisk to receive call from a DID provider . In configured
everything in freepbx properly and its working . I forwarded incoming calls
from did to a certain extension . Now i tried calling from another sip
provider to this box , when i call from other provider to my DID number
then c
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
[EMAIL PROTECTED] wrote:
>
> Does anyone have experience with recording multiple calls
> simultaneously on a single system with or without performance trouble?
> What kind of system do I need?
>
Well, isnt this just a simple calculation?
Do a rec
Ron McCarthy wrote:
Hey list,
Im in this HUGE crisis. Im trying to get a Polycom 601 with two expansion
modules to work. I need the XML config files I guess. Does anyone have
these
I can have? Im trying to get this phone up and running, and haveing MUCHO
problems, can someone help me out!! Im
On Wed, 2006-11-22 at 15:45 +0530, ram wrote:
[snip]
> Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring
> switchtype
> Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring
> signalling
> Nov 22 15:43:23 WARNING[14623]: chan_zap.c:10874 setup_zap: Ignoring
> rxwi
Hi
I have calls comming from a SIP-ATA-Box via Asterisk to PSTN Phones by
outbound SIP.
Now i want to detect DTMF-Tone Code coming from the called party to
trigger a signal.
Can this be done with asterisk? I read that the codec with DTMF
detection are ulaw and alaw. But i couldn't find a command t
Tom Rymes wrote:
>Dan,
>
>If I have followed this thread correctly, your problem
>is that, when you pick up a local analog phone connected
>to asterisk through a zap channel, asterisk generates
>a dialtone, and everything works fine, except that the
>echo is intolerable. Then, you install an
Enrico Pasqualotto wrote:
Enrico Pasqualotto wrote:
hi all, how can I set the interval in second from retrasmit the magic
packets when qualify is set to on?
You have to set qualify=second instead of qualify=yes|no.
This is WRONG. qualify=500 means "consider this device lagged if
responses t
Why would I want to join MSN groups then MS can't get an OS right!
Now MS whats to do get into VOIP that will be a total messup.
The thing is when MS will try to say that they asterisk.
MS has no place anywhere around Asterisk.
You will see what I mean just look at the bottom of MY website.
I j
Hi,
We want to build an Asterisk system that needs to be able to record,
when in a peak situation, a maximum of twenty calls simultaneously. I
could not find any reference to performance and recording. I need to
order a new server but need to know the specs I need.
Does anyone have experience
Yes, already.
Waiting now for a new firmware...
Regards,
Alban
Le Mercredi 22 Novembre 2006 13:50, Steve Davies a écrit :
> On 11/22/06, Alban <[EMAIL PROTECTED]> wrote:
> > I'm having the same problem, pressing a speed dial/extension when 2 calls
> > are on the phone connect the 2 calls together.
Hi
Most of our customers have generic names like "Hospital", so I need to
rewrite their caller ID name by looking up the number in a database on the
Asterisk server, and rewriting the name such as "Reading Hospital" so that
we know who's calling.
Any idea if this can be done with Asterisk, a
Hello Users
I'm Doing working on Both OpenSER and Asterisk ...
9001 and 9003 are registered in OpenSER
in extension.conf
[from-sip]
exten=>115,1,Park()
exten =>115,2.Hungup()
in Feature.conf ( default park no 701)
in sip.conf
[9001]
...
..
[9002]
[9003]
When 9003 dial the 115 ( Parking i
Mark was working on this, I think it was called "sla" and it called
"something line apperance"
On 11/21/06, John Lange <[EMAIL PROTECTED]> wrote:
Hints are not working in 1.4b3 period. Snom 360s do not show any status
updates. However, before you file a bug report you might want to check
to se
You didn't get the memo? I will have alice send you over a copy
Seriously, yes there are laws against what he did but it would be in
the best interest of their sales department to send the software with
every phone.
On 11/21/06, Brian Capouch <[EMAIL PROTECTED]> wrote:
Matt wrote:
>
On 11/22/06, Alban <[EMAIL PROTECTED]> wrote:
I'm having the same problem, pressing a speed dial/extension when 2 calls are
on the phone connect the 2 calls together. Typing the number instead of using
speed dial works.
With older firmware, 6.2.1 or 6.3, it was working... But then other problem
w
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