Does anyone have an end-to-end summary of how they have successfully set up the buddy feature including all the relevant Asterisk and Polycom config snippets. All I have been able to do so far is scrounge up bits and peices from the list and Wiki - nothing that covers the entire process... I think
Title: Asterisk Advice
Hi
Looking at using Asterisk for a new remote office with about 200 extensions. We have an existing EPABX in our central office with about 120 POTS lines.
The central office and the new office will be connected over a 4Mbps leased line.
We want to be able to connect
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
I use SellVOIP and Voxee which both seem to allow that.
Hi Ira.
Thank you for this information!
--
Tomislav Parčina
Lama Computers Split
Stinice 12, 21000 Split
Tel.: +385(21)495148
Mob.: +385(91)1212148
SIP: [EMAIL PROTECTED]
e-mail:
K Y Iyer wrote:
Hi
Looking at using Asterisk for a new remote office with about 200
extensions. We have an existing EPABX in our central office with about
120 POTS lines.
The central office and the new office will be connected over a 4Mbps
leased line.
We want to be able to connect our
You can try VoipJet (http://www.voipjet.com)
A simple configuration in you extensions.conf as below will solve your
problem.
exten = _X.,1,SetCIDNum(1341212)
exten = _X.,n,Dial(IAX2/[EMAIL PROTECTED]/${EXTEN})
Thank you!
--
Tomislav Parčina
Lama Computers Split
Stinice 12,
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Tomislav,
RPID is short for Remote-Party-ID. Basically, Remote-Party-ID is a
way, using a header (Remote-Party-ID) to completely separate caller id
presentation from authentication information with SIP. I should point
out
Hi,It looks like Voice quality is more and more frequently embedded in phones themselves, mostly with RTCP statistics.I've understood RFC3550 is a first step and RFC3611 goes one step further with MOS and R-factor mesurement.
I suppose that best practices should include a way to monitor those
(Subject changed from 'Re: [asterisk-users] Polycom Buddy Watch Broken with
2.0.1 Firmware?' as it was a bit off topic).
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Bishop
Sent: 03 October 2006 07:34
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
On 3 Oct 2006, at 08:31, K Y Iyer wrote:
Hi
Looking at using Asterisk for a new remote office with about 200
extensions. We have an existing EPABX in our central office with
about 120 POTS lines.
The central office and the new office will be connected over a
4Mbps leased line.
We
Hi,If I may follow on this thread, would you try to :1. offer users a single extension for both voice and fax calls with (automatic or human handled) fax detection2. or would you build a centralised fax server gathering fax extensions with which a staff member would read fax headers to forward
I'm looking for a simple and cheap way to ensure QoS when the local
network usage is high. Has anyone used the ZyXEL ES-105A Desktop
Ethernet Switch (http://shurl.org/ulXPP) with Asterisk? AIX in
particular?
It looks like it's typically less than $40. Which is dirt cheap
compared to real QoS
Hi there,
I have two active channels (say SIP/3000-xyzk and Zap/1-1) on a wait
context and would like to bridge them together.
I found a temporary solution using the MeetMe application, but I was
looking for a more native one, not involving conference rooms at all.
Thanks in advance for
If you really want _07. to be tested afterall the above patternmatches, you must define it in other context and add it as an include for the current context.Asterisk first will look for your patternmatches in the current context and oonly after this will lookup your include context. This way you
Hi,
if I setup a call from the pbx (+35220404244) to another sip client
(2040) registered on the same asterisk box, the success of the call
setup depends on in which context the call from the pbx falls in.
With the configuration below, sip call signalisation works fine but as soon
as the sip
Title: Asterisk Advice
HiLooking at using Asterisk for a new remote office with about 200 extensions. We have an existing EPABX in our central office with about 120 POTS lines.The central office and the new office will be connected over a 4Mbps leased line.We want to be able to connect our
Hello!
Why Asterisk tell: Unknown signalling method 'pri_cpe'
Why the asterisk does not know such signaling method?
[chan_zap.so] = (Zapata Telephony)
Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown
signalling method 'pri_cpe'
Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601
Hi ,
I got an error (actually the CLI broken with error message below)
when the realtimeupdate running. I am using the latest version 1.2 of
asterisk and addon. Anyone facing the same problem and anyone can
tell how to solve it.? Thanks!
---
-- Executing
Eugeniy Khvastunov wrote:
Hello!
Why Asterisk tell: Unknown signalling method 'pri_cpe'
Why the asterisk does not know such signaling method?
[chan_zap.so] = (Zapata Telephony)
Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown
signalling method 'pri_cpe'
Oct 3 13:04:02
I need a good phone for the reception desk. Currrently I am using Grandstream, but parking and over head paging is not easy for the receptionist, as it was on the Nortel phone. Paging is still acceptable, which is programmed on one of the shortkeys, but parking on this phone needs to press #70#
I am getting a couple of messages in the log I don't understand. The
first is:
Unsupported transport 'UDP'
The second is
Oct 3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing...
Oct 3 06:55:54 DEBUG[20372] channel.c: Nobody there, continuing...
Oct 3 06:55:54 DEBUG[20372]
yusuf пишет:
Eugeniy Khvastunov wrote:
Hello!
Why Asterisk tell: Unknown signalling method 'pri_cpe'
Why the asterisk does not know such signaling method?
[chan_zap.so] = (Zapata Telephony)
Oct 3 13:04:02 ERROR[5823]: chan_zap.c:10601 setup_zap: Unknown
signalling method 'pri_cpe'
Oct
Hi all,
I had the same problem before so i tried reinstalling and using all
the defaults i could. But unfortunatelly it's the same, it seems that
mysql defined users can't access to the codec files. In both
situations the phone registers and i can see it with sip show
peers(with
Title: Asterisk Advice
Cory J AndrewsVOIPSupply.com454 Sonwil
DriveBuffalo, NY 14225++voice - 800.398.VoIP
X3402email - [EMAIL PROTECTED]AIM - B2CORY
- Original Message -
From:
K Y Iyer
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Yes, I know it, and if there is no solution I will use it, but I just prefer
ChanSpy because I can spy the channel always, even it is not in a meetme
conference.
El Martes, 3 de Octubre de 2006 02:49, Nicolás Gudiño escribió:
I'm using the ChanSpy command for monitor a conversation of a
Dear all,Do you know any tool that can administrate Asterisk
remotely? I only need basic functionalities like adding new extensions,
queus and basic configuration. The problem is that I can't install that
in the same machine as Asterisk (since it is running in open wrt).
Can anyone help me
www.freepbx.org
Bill
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Voipers Portugal
Sent: Tuesday, October 03, 2006
8:22 AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] Asterisk
manager
Dear all,
Do you know any
But that cannot be done remotly, can it? I can't load any more programs in my Asterisk machine. Does it support that?On 10/3/06, Bill Gibbs
[EMAIL PROTECTED] wrote:
www.freepbx.org
Bill
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of
Voipers
Hi All,
I have been looking
at the sources in order to find out how to add support for a new codec but
haven't had much break-through. It would be very helpful if someone can provide
me a few pointers as to where to look. I am very new to Asterisk PBX so please
provide the information in
No you cant install freepbx on a openwrt
The only choice you have is to ssh in
using putty or similar to add extensions and make conf mods.
Cheers,
Dean
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Voipers Portugal
Sent: Tuesday, 3 October
I want to pop up a web page when a queue member phone
rings but, instead of displaying the clid, I want to
display the DID number the call came in. Any ideas how
to best implement this?
Greg
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the
I know that, that is why I asked if there was any tool that would do something like that, but by acessing the Manager API?Anyone?On 10/3/06, Dean Collins
[EMAIL PROTECTED] wrote:
No you cant install freepbx on a openwrt
The only choice you have is to ssh in
using putty or
Hi guys, i just
installed the flortz patch with bristuff-0.2.0-RC8r but when i load module
zaphfc i get this warning message:
Warning: ignoring
syns_slave=0, no such parameter in this moduleWarning: ignoring
timer_card=1, no such parameter in this moduleModule zaphfc loaded, with
I have been trying to get automon working on Asterisk 1.2.12.1 and I am having
some problems. I have searched the list archives and have not found my answer
either. This system is setup for SIP to SIP calls with G.729 codecs. I
believe that I have the config files setup (*1 enabled in
Hi,
I understand that 700 is the default extension to initiate a Call Park.
Does anyone know of a way to configure Asterisk such that it has
more than one park extension for e.g. parkexten = 700,800,900
regards,
Kwang Mien
___
--Bandwidth and
Hello
I would think that using the manager interface, would be the easiest way of
implementation.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af Greg Delgado
Sendt: 3. oktober 2006 14:44
Til: asterisk-users@lists.digium.com
Emne:
p == picciuX [EMAIL PROTECTED] writes:
p Why not? Use group categories... you can assign two groups to the
p same channel if they are different categories...
Thank you very much. A colleague of mine found the time to test this,
and it works.
Next challenge: figure out how to add a call to a
Hello,
RING 1 26 TIPfirst Zap channel
RING 2 27 TIPsecond Zap channel
RING 3 28 TIPthird Zap channel
etc..
Best Regards,
Francois BERGERET,
France.
-Message d'origine-
De : [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] De la part de C F
Envoyé : mardi 3
Hi group,Can anyone help out in selecting the right codec to download from the digium site.Im using an AMD Sempron 2800+ CPU speed 1.6 GhzThanks in advanceDan
___
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Have you considered using the Flash Operator Panel (www.asternic.org)?
Jon Schøpzinsky wrote:
Hello
I would think that using the manager interface, would be the easiest way of
implementation.
Jon
-Oprindelig meddelelse-
Fra: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] På vegne af
On Tue, 2006-10-03 at 09:20 -0400, Clif Jones wrote:
I have been trying to get automon working on Asterisk 1.2.12.1 and I
am having some problems. I have searched the list archives and have
not found my answer either. This system is setup for SIP to SIP calls
with G.729 codecs.
You do have
Hello,
I thought I'd ask this, just in case I'm wrong. We're trying to set up
'remote' users via asterisk. Basically all there is to this is asterisk
forwarding a DID to a cell phone. My question is this: Is there any
possible way for our local asterisk box to setup the connection and the
On 10/3/06, Greg Delgado [EMAIL PROTECTED] wrote:
I want to pop up a web page when a queue member phone
rings but, instead of displaying the clid, I want to
display the DID number the call came in. Any ideas how
to best implement this?
Checkout Asterisk Desktop Manager at
First all i wrote syns_slave instead sync_slave, anyway also with sync_slave i
got the same error.
This is the modinfo output :
modinfo zaphfc
filename:/lib/modules/2.4.31/misc/zaphfc.o
description: HFC-S PCI A Zaptel Driver
author: Klaus-Peter Junghanns [EMAIL PROTECTED]
license:
Hello,
We are using asterisk with 6 POTS lines and Caller ID is not always
read from the lines properly. Is there a way to make asterisk
wait for the caller id before proceeding with the dial plan or is it
possible a setting is wrong in a conf file somewhere? Any
guidance would be helpful.
Here is an example of what I have:
in extensions.conf:
exten = 2111,hint,SIP/2111
exten = 2111,1,Dial(SIP/2111,60)
my Polycom's all pull config's via TFTP. Due to the nature of our setup, I
have individual configuration files for each phone. I have in my
mac-address-directory.xml file the
Does anyone have a working example of how Presence can be enabled and made
functional with the Cisco 7961's utilizing the SIP image?
Scott Higginbotham
Systems / Network Operations Manager
215.259.2185 or 1.800.835.5710 ext 2185
[EMAIL PROTECTED]
___
On Tue, Oct 03, 2006 at 10:33:38AM -0400, Steve Glaus wrote:
I thought I'd ask this, just in case I'm wrong. We're trying to set up
'remote' users via asterisk. Basically all there is to this is asterisk
forwarding a DID to a cell phone. My question is this: Is there any
possible way for
Sorry but that doesn't answer my question. My question is was: Which
slot on the TDM2400 is Zap channel 1, which according to you will be
Pair 1 (1 and 26) on the 66 block?
On 10/3/06, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
Hello,
RING 1 26 TIPfirst Zap channel
RING 2 27 TIP
Do you have sendcalleridafter=2 in your [channels] section of
/etc/asterisk/zapata.conf? (I had to change it for mine to work)
Nick
--
Nick Ellson
CCDA, CCNP, CCSP, CCAI,
MCSE 2000, Security+, Network+
Network Hobbyist, VFR Private Pilot.
On Tue, 3 Oct 2006, [EMAIL PROTECTED] wrote:
We have two * boxen linked with an IAX trunk they are on different networks.
Calling from any ext to any other remote ext on either box works
fine.Outgoing calls from either box also works fine.
Problem when dialing in from DID (SIP or IAX provider) number, the
caller cant transfer to any
Hi again,
I've found a strange thing. If I use the same account from a
softphone(twinkle) it behaves well.
As fas as i've tried, connecting both phones, a linksys phone and
twinkle, at the same time to the same account and calling to the same
extension asterisk works well for the twinkle but
I'm trying to set up the o extension for the voicemail app that will bring the caller out if they go to someone's voicemail. I've set the operator=yes flag in the voicemail.conf file, I then added an o extension into extensions.conf
to point to a macro that I have created. When I try testing this
On 14:08, Tue 03 Oct 06, Eduard Mart?nez wrote:
Yes, I know it, and if there is no solution I will use it, but I just prefer
ChanSpy because I can spy the channel always, even it is not in a meetme
conference.
Simply run every call via a meetme
--
Michiel van Baak
[EMAIL PROTECTED]
Jay R. Ashworth wrote:
On Tue, Oct 03, 2006 at 10:33:38AM -0400, Steve Glaus wrote:
I thought I'd ask this, just in case I'm wrong. We're trying to set up
'remote' users via asterisk. Basically all there is to this is asterisk
forwarding a DID to a cell phone. My question is this: Is there
Hi,
I have asterisk with zaphfc, two ISDN cards, one TE mode another NT mode.
When connecting cards together with isdn crosscable link is ok.
When connecting TE card to the Panasonic KX-TDA100 link is also ok.
When connecting NT mode card to the panasonic, panasonic doesn't see the
link,
Before acting on this e-mail or opening any attachments you are advised to read 20:20 Logistics Ltd,Dextra Solutions Ltd, CaudwellLogistics Ltd, The Mobile Phone Repair Company Ltddisclaimer at the end of this e-mail.
You might play with the ParkAndAnnounce() application which parks a call
and then plays the resultant parking slot number to a channel of your
choosing.
Moj
Zeeshan Zakaria wrote:
I need a good phone for the reception desk. Currrently I am using
Grandstream, but parking and over head paging
Hi,
You haven't applied the florz patch correct. You apply it like this :
1. Go to your bristuff directory
2.zcat [path to florz patch
file]/zaphfc_0.3.0-PRE-1o_florz-12.diff.gz | patch -p1
When it is applied the you will get this output:
[EMAIL PROTECTED] ~]# modinfo zaphfc
filename:
On Tue, Oct 03, 2006 at 05:25:38PM +0100, Paul Philpott wrote:
Confidentiality Notice This e-mail is confidential and intended for the use
of the named recipient only. If you are not the intended recipient please
notify us by telephone immediately on +44(0)1270 412020 or return it to
- Original Message -
From: C F [EMAIL PROTECTED]
Sorry but that doesn't answer my question. My question is was: Which
slot on the TDM2400 is Zap channel 1, which according to you will be
Pair 1 (1 and 26) on the 66 block?
Each card supports 4 lines so the card in Slot 6 would be
Matt wrote:
We have two * boxen linked with an IAX trunk they are on different
networks.
Calling from any ext to any other remote ext on either box works
fine.Outgoing calls from either box also works fine.
Problem when dialing in from DID (SIP or IAX provider) number, the
caller cant
Thank you
On 10/3/06, Mailing List [EMAIL PROTECTED] wrote:
- Original Message -
From: C F [EMAIL PROTECTED]
Sorry but that doesn't answer my question. My question is was: Which
slot on the TDM2400 is Zap channel 1, which according to you will be
Pair 1 (1 and 26) on the 66 block?
Does anyone know if asterisk currently supports the US government's
Communications Assistance for Law Enforcement Act (CALEA) regulations?
If not, does anyone have this item on their To-Do list?
For those that are not familiar with CALEA, it's the governement's way
of intercepting or
Rich Adamson wrote:
For those that are not familiar with CALEA, it's the governement's way
of intercepting or monitoring voice communications (presumably with
a court order) for law enforcement personnel, etc. The broadband /
ITSP compliance due date is May 14, 2007.
For those unfamiliar
On Tue, Oct 03, 2006 at 12:13:27PM -0500, Rich Adamson wrote:
Does anyone know if asterisk currently supports the US government's
Communications Assistance for Law Enforcement Act (CALEA) regulations?
If not, does anyone have this item on their To-Do list?
Why in hell *would* anyone?
I
Hi,
I have several uniden UIP200 phones with asterisk that just hang.
I am running that phones latest software.
I just connected about 3 days ago a UIP200 to the system. I have not called
that extension or used. just connected it. I noticed it's display had hung.
tried to pickup the handset and
Jay R. Ashworth wrote:
On Tue, Oct 03, 2006 at 12:13:27PM -0500, Rich Adamson wrote:
Does anyone know if asterisk currently supports the US government's
Communications Assistance for Law Enforcement Act (CALEA) regulations?
If not, does anyone have this item on their To-Do list?
Why
Thanks for the response. Answers inline..
-Original Message-
From: Michael Neuhauser [EMAIL PROTECTED]
Sent: Oct 3, 2006 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Clif Jones [EMAIL PROTECTED]
Subject: Re: [asterisk-users]
We are running asterisk-1.2.4 with zaptel-1.2.7 on Fedora Core 4-2.6.14-1.1656_FC4smp. It is installed on a Dell PE 2500 with 2x900 MHz processors and 1 Gb RAM and 1 SCSI Disk. The server has a Digium TDM400P card which is connected to 4 POTS lines. The server is also
I want to bundle two ISDN-lines (4 B-channels alltogether) with
asterisk. I need it for remote administration of my customers. Sometimes
the 2 B-channels of the first ISDN lines are busy (speech or data
connection), then asterisk should to take a free channel of my second ISDN
line.
We need no
On Tue, Oct 03, 2006 at 01:59:31PM -0400, J. Oquendo wrote:
Does that mean that they've made it illegal to use Asterisk?
Illegal? It's illegal to question your new government thank you.
I, for one, welcome our new Republican overlords.
Cheers,
-- jr 'not' a
--
Jay R. Ashworth
I, for one, welcome our new Republican overlords.
lol you are just full of pop culture references, aren't you?
___
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asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
Inline...
On Tue, Oct 03, 2006 at 12:13:27PM -0500, Rich Adamson wrote:
Does anyone know if asterisk currently supports the US government's
Communications Assistance for Law Enforcement Act (CALEA)
regulations? If not, does anyone have this item on their To-Do list?
Why in hell
On 3 Oct 2006, at 19:53, Colin Anderson wrote:
I, for one, welcome our new Republican overlords.
lol you are just full of pop culture references, aren't you?
Abortions for some, miniature American flags for others.
Seriously though - is anyone aware of a precis of CALEA? I'm about to
On 10/3/06, Matthew Thompson [EMAIL PROTECTED] wrote:
On 3 Oct 2006, at 19:53, Colin Anderson wrote: I, for one, welcome our new Republican overlords. lol you are just full of pop culture references, aren't you?Abortions for some, miniature American flags for others.
Seriously though - is anyone
Matthew Thompson wrote:
On 3 Oct 2006, at 19:53, Colin Anderson wrote:
I, for one, welcome our new Republican overlords.
lol you are just full of pop culture references, aren't you?
Abortions for some, miniature American flags for others.
Seriously though - is anyone aware of a precis of
For about 20% of the calls to the outside world, the voice on the
other end of an outside line is incredibly choppy. Enough to where
we have to hang up and call on a cell phone. It is always the same
numbers that are choppy. The funny thing is, if I press mute while
talking on a choppy
Rich Adamson wrote:
As the OP of this thread, I'm involved with an itsp operation that
includes asterisk with links to regional/national itsp facilities,
PRI's to local pstn facilities, and broadband sip/iax connections to
residence and business customers. I don't think the legalese will be
Sorry for hijacking this thread but I was about to post the same problem.
I get this in my logs
Oct 3 08:30:03 WARNING[29528] chan_zap.c: Ignoring signalling
Oct 3 08:30:03 WARNING[29528] chan_zap.c: Ignoring sendcalleridafter
Oct 3 08:30:03 WARNING[29528] chan_zap.c: Ignoring signalling
Oct
On Tue, Oct 03, 2006 at 12:53:12PM -0600, Colin Anderson wrote:
I, for one, welcome our new Republican overlords.
lol you are just full of pop culture references, aren't you?
I don't get to use that one much, this not being Slashdot.
Cheers,
-- jra
--
Jay R. Ashworth
On Tue, Oct 03, 2006 at 02:28:01PM -0500, Brandon Galbraith wrote:
end user (IANAL, though). CALEA should be fairly easy to implement
either in Asterisk (with patches) or on the media gateway (when
the call hits the PSTN/GSM network). It's even easier if it's
SIP end to end, as you
On Tue, 2006-10-03 at 14:09 -0500, Rich Adamson wrote:
[snip]
Anyone care to confirm or elaborate on those thoughts / guesses?
These guys seem to have a ready made solution or box:
http://www.lawfulinterception.com/
Regards,
Patrick
___
--Bandwidth
Going to the other extreme, what would it take to create an untappable and
untraceable telephone service over the Internet?
Asterisk is a good start, especially because the code can be examined (as
long as G729 is avoided) and any law enforcement back doors removed.
Now instead of trying to
I'm sorry. You seem to have fallen into the sar-chasm.
And I thought the smiley would be enough hint. :-)
never. 2 Smileys: maybe ;)
Yes of course. These notebooks tend to get forgotten in a cupboard
until the day they're needed. And then they're so out of date that
they're more
On 10/3/06, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Tue, Oct 03, 2006 at 02:28:01PM -0500, Brandon Galbraith wrote:end user (IANAL, though). CALEA should be fairly easy to implementeither in Asterisk (with patches) or on the media gateway (when
the call hits the PSTN/GSM network). It's even
On Tue, Oct 03, 2006 at 02:47:46PM -0500, Henry J. Cobb wrote:
Going to the other extreme, what would it take to create an untappable and
untraceable telephone service over the Internet?
Well, define untraceable. Avoiding traffic analysis is *much* harder
than avoiding content divulgement, and
Does anyone have any info on using the call-park feature on Polycom phones? All I can find is that it must be supported by the SIP server. It doesn't appear to have any related configuration settings or other such clues as to how to use it.
Assuming the feature is not supported by Asterisk,
Hello,
Which analog hardware is the best in quality-of-voice terms? In a
production environment, which one gives more reliability and stability??
Thanks in advance,
R.R. Libera
___
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Henry J. Cobb wrote:
Going to the other extreme, what would it take to create an untappable and
untraceable telephone service over the Internet?
Skype using Tor+Privoxy+Knoppix+open_network_anywhere
--
J. Oquendo
Have a number of asterisk servers and want to get some good stats tracking
going (with Asterisk-Stat) -- but this requires cdr logging to mysql, apache
and the stats software running on each server.
Or does it? Of course, I can either run the stats package on the webserver
and direct it to each
On Tue, Oct 03, 2006 at 08:41:47PM +0200, Ekkard Gerlach wrote:
I want to bundle two ISDN-lines (4 B-channels alltogether) with
asterisk. I need it for remote administration of my customers. Sometimes
the 2 B-channels of the first ISDN lines are busy (speech or data
connection), then asterisk
Anyone used Astmanproxy written by David Troy?
I just tested it for the first time. I connected the proxy to three backend
Asterisk servers, and ran a SIPPeers command.
AsteriskManagerOutput
Event Value=PeerEntry/
Channeltype Value=SIP/
ObjectName Value=3254103/
ChanObjectType Value=peer/
Is it possible to set up SER with asterisk so that any INVITE that is
forwarded from SER to asterisk does not need to be authenticated by
asterisk?
Thanks,
Mark
___
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asterisk-users mailing list
To
On Tue, 2006-10-03 at 14:16 -0400, Clif Jones wrote:
Thanks for the response. Answers inline..
-Original Message-
From: Michael Neuhauser [EMAIL PROTECTED]
Sent: Oct 3, 2006 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com, Clif
How do I take out few extensions (vm enabled extensions) from the
default company directory listing?
thanks.
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On Tue, Oct 03, 2006 at 05:26:07PM -0300, R.R. Libera wrote:
Which analog hardware is the best in quality-of-voice terms?
A WeCo 2500. Hands down.
In a
production environment, which one gives more reliability and stability??
Where can I buy Digium Hardware in Tampa? Not Ebay or e-commerce. Thanks
a lot.
R.R. Libera
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Mark Price wrote:
Is it possible to set up SER with asterisk so that any INVITE that is
forwarded from SER to asterisk does not need to be authenticated by
asterisk?
Sure, in your sip.conf entry:
[your ser entry]
type=friend
host=enter ip address
insecure=port,invite
That last entry will
On the entry to hide, add hidefromdir=yes as an option.
On 10/3/06 5:24 PM, asterisk-user [EMAIL PROTECTED] wrote:
How do I take out few extensions (vm enabled extensions) from the
default company directory listing?
thanks.
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Do you have anything special in your sip.conf for the Polycom phones?On 10/4/06, Scott Higginbotham [EMAIL PROTECTED]
wrote:Here is an example of what I have:in extensions.conf:exten = 2111,hint,SIP/2111
exten = 2111,1,Dial(SIP/2111,60)my Polycom's all pull config's via TFTP.Due to the nature of
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