Hi,Did anyone try to set up alarmreceiver application over IP network? Which ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck. Maybe I did something wrong with alarmreceiver.conf (I tried diverse settings, but nothing worked).
Sometimes alarmreceiver is able to get some
I have 4 asterisk servers which are Friends and each one has an account for termination. A total of 5 peers each.
Currently, the setup is as follows
iax.conf=
[FriendName]
type=friend
context=server_friend
secret=donttell
host=friend.dyndns.com
qualify=750
=
Hi,
I have an ISDN phone connected to a hfc-s card. I use it to phone via
an iax provider to foreign countries. Inside my country it works reliable,
but to other country it happens very often that the other side hears ringing
and before it can take the phone the line is dropped. What makes me
Hi All,
We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM
3.1.3 as per http://www.polycom.com/resource_center/1,,pw-492,00.html
Can someone help? We have legitimately obtained these phones but even
our official distributor can't get their hands on updated firmware. The
only
Eric,
I have a copy of both.
They are at my office. Send me an email directly and tomorrow I'll forward
you a copy.
- Gabe
- Original Message -
From:
Eric
Bishop
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, March 27, 2006 12:57
The worst thing on all Polycom IP phones is the speaker phone's poor
quality. You could not have a conference call using the speakers, only
the head phone.
Denis.
Hahaha, clearly this guy is on crack. (no offense)
I have uploaded MP3s to my asterisk box and have it programmed to play
At the risk of being redundant, VoIP and Alarm is known not to mix well.
Some of the tones used by an alarm system do not behave in the same way
as conventional DTMF. This will vary greatly based on the actual alarm
format used (and there are at least thirty different formats.) I don't
know the
What is maximum length of name in caller ID? How much charters can I put and be
sure it will work fine?
--
Tomislav Parcina
tparcina#lama.hr
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Hi,I am wondering if it is possible with astcc to make a second call without hangup and be oblige to re-enter all the codes.Any idea how to do?Thank you
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To
Skeeve Stevens wrote:
I just picked up a Cisco 7940 from an Auction… and would like to use
it on an Asterisk box.
Can anyone give me a pointer where I should start so I can get it
working?
http://www.voip-info.org
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty
Yes.
disallow=all
allow=g723
Allow only g723 codec.
roberto
2006/3/26, Mohammad Salaque [EMAIL PROTECTED]:
Hello list,Another newbie question,.if I putdisallow=all andallow=g723my sip.cofdoes it mean thatextension could only communicate usingg723 ?bellow is one of my extension example
Detailed info about snom beta firmware can also be found at snom-wiki
e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes
Regards,
-
Usman Tahir
snom technology AG
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Please stop send me email
Best Regards,
Mr.Peeramate Rochanasmita
Project Manager/General Manager
This message was sent to me?
--
Tomislav Parcina
tparcina#lama.hr
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Anton Krall wrote:
Hi John, yes, Im using native transfer. What I do is use Monitor on the
dialplan of the extension that picks up the call coming from PSTN, so after
that, if the extension forward or transfers the call, monitor keeps
recording all thru the end of the call no matter where it
Hi Guys
Is there anyway to adjust the output volume on the Polycom
501?
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5.5.1b is neither listed on the snom-wiki nor is any changelog for 5.5.1b
listed.
-Dan
On Mon, 27 Mar 2006, Usman Tahir wrote:
Detailed info about snom beta firmware can also be found at snom-wiki
e.g. http://snom.com/wiki/index.php/Beta_Firmware#Release_Notes
Regards,
Hello,
I am trying to register to the asterisk with different phone number,
login and password. This is my setting in the sip.conf:
[246079011]
type=friend
context=cisco
secret=XXX
host=dynamic
username=tomas
allow=alaw
nat=yes
canreinvite=no
mailbox=246079011
but I get this reply:
Mar 27
Guyz,
I wanna test my asterisk load capability before going to production, anyone know is there any call simulator to test this thing?
Thanks in advance,
Voipman
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I can't transfer call which was picked up with feature - group pick up. I'm
running * 1.2.5.
The problem is that asterisk doesn't hear that I have pressed #1 and doesn't
play transfer sound for me.
Regular phone calls I can transfer without problem. Can anybody check is this
a BUG?
--
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Absolutely right :)
\ escapes the next character, so if you wants *69 to go through
immediately, you'd put \*69 so that the * gets recognized as a digit.
, returns the dialtone sound. When my users hit 9, they like to hear
the
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
There is no such thing as a 'free' G.729 - The DSP Group has claimed and
defended the Patents they hold against the algorithm and process.
Please do not use Asterisk/Digium related resources to exchange this
information - They are
Could anyone provide me some link in order tovoicemail to email working, I believe I have to give SMTP settings but do not know where.
Thx
Voipman
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Voicemail uses sendmail on your system. If your machine can send mails
using sendmail, so will asterisk.
Rudolf
On 3/27/06, voipman [EMAIL PROTECTED] wrote:
Could anyone provide me some link in order to voicemail to email working, I
believe I have to give SMTP settings but do not know where.
Its in vociemail.conf.
If you built asterisk with a basic running config
there should be examples in there.
Dovid
--- voipman [EMAIL PROTECTED] wrote:
Could anyone provide me some link in order to
voicemail to email working, I
believe I have to give SMTP settings but do not know
where.
I found a solution... I just has to enter an Answer
line and now it behaves as I wanted. Here is the
working code:
[inbound]
exten = 1234567,1,Set(GROUP()=limit)
exten = 1234567,2,GotoIf($[${GROUP_COUNT()}2]?103)
exten = 1234567,3,Dial(Zap/5Zap/6,25,tT)
exten = 1234567,4,Voicemail,u110
MBIT Technologies wrote:
Hi Guys
Is there anyway to adjust the output volume on the Polycom 501?
Yes. I did this over the weekend. Look in your Polycom sip.cfg for a
line tx.digital.handset. I had to set mine to -6 before the levels came
down within tolerance.
There is one for
I got past this by changing spinlock.h in the
/usr/src/kernels/2.6.9-34.EL-x86_64/include/linux/ folder. (I am using 64bit
kernel)
I changed:
#define DEFINE_RWLOCK(x) rw_lock_t x = RW_LOCK_UNLOCKED
#define DEFINE_RWLOCK(x) rwlock_t x = RW_LOCK_UNLOCKED
to:
#define DEFINE_SPINLOCK(x) spinlock_t
You are signed up to the list. If you want out go to
http://lists.digum.com
--- Peeramate @ SIPPhone Thailand
[EMAIL PROTECTED] wrote:
Please stop send me email
Best Regards,
Mr.Peeramate Rochanasmita
Project Manager/General Manager
SIPphone (Thailand) Co., Ltd.
644/19 Moo 1
I'm trying to set up the following application:
When a SIP extensions calls another one which is busy, the caller would be
able to ask for an automatic callback: when the callee becomes available
again, asterisk would ring both the caller's and the callee's phones and
connect them when both
SIPPS is one, I would like to hear of
others.
Of course you could create a dialplan that
loops calls in and out.
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
From: voipman
[mailto:[EMAIL PROTECTED]
Sent: Monday, March 27, 2006 6:39
AM
To:
--- Tomas Komarek [EMAIL PROTECTED] wrote:
Hello,
I am trying to register to the asterisk with
different phone number,
login and password. This is my setting in the
sip.conf:
[246079011]
type=friend
context=cisco
secret=XXX
host=dynamic
username=tomas
allow=alaw
nat=yes
we are thinking about replacing a median 1 pbx system, we have about 40
phone.
i got 4 incoming pot lines (all the same number), i don't know if i can
use one tsu600 port as a fxo (for the pots) and all the rest as fxs, or
should i use a tdm400p with 4 fxo's (for the pots,inside the asterisk
Hi,Thanks for so fast reply.Now, rather than just being a nay-sayer, let me refer you to the BoschC900V2 device. It takes a signal from just about any panel and converts
it into IP to be received by a Bosch receiver.Is it possible to connect C900V2 with Asterisk, (did you do such a thing, did you
The C900V2 only connects with Bosch receivers. In fact, all of the IP
communicators in the industry are proprietary. There is a committee
working towards a standard, but my understanding is that we still have a
decent wait ahead of us.
Bob McDowell
From:
In article [EMAIL PROTECTED], [EMAIL PROTECTED] says...
Best bet is to get Asterisk Chan_Sccp http://chan-sccp.berlios.de/
1.) setup your /etc/asterisk/sccp.conf with something like:
2.) setup lines 30/31 as a custom extension in astersik (i used amp)
and had it dial SCCP/30 and SCCP/31
Well, I did, but the reason is still the same, if the username is
different from the phone number, asterisk rejects the registration :-(
Dovid Bender napsal(a):
--- Tomas Komarek [EMAIL PROTECTED] wrote:
Hello,
I am trying to register to the asterisk with
different phone number,
login
Hello,I am using Asterisk-java, the Manager. And I have a problem I don't know howto sort it out!:Sometimes, when I send an OriginateAction my code receives an exception withthis message:
Timeout waiting for response to OriginateI don't know what it means as Asterisk receives the action and then
I'm postponing this activity indefinitely but I collected some ideas.
Try something similar to this recipe:
First of all store dialed extension number as
exten = _[2-8]XX,102,SetVar(${UNIQUEID}=${EXTEN})
exten = _[2-8]XX,103,Goto(busyphone,s,1)
then you can use 3
Really?
Mmhh seems you got working what I want and I what you want.. Hehehe try
using monitor instead of mixmonitor.. Maybe there is a difference in apps.
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|John Daragon
|Sent: Monday, March 27, 2006
what do you men adjust? (I guess you already tried the keys
on the pad right)?
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MBIT
TechnologiesSent: Monday, March 27, 2006 4:57 AMTo:
asterisk-users@lists.digium.comSubject: [Asterisk-Users] Polycom
501
If you purchased your phones from an
authorizedreseller they shouId be able to provide
this.
Ican help you. Please contact me off list.
-Mike
Michael Crown Managing Partner www.thevoipconnection.com 321.989.6728
ext. 611 sip:[EMAIL PROTECTED]
From: Eric Bishop
[mailto:[EMAIL
Quoting Andrew Nowrot [EMAIL PROTECTED]:
Hi,
Did anyone try to set up alarmreceiver application over IP network? Which
ATA can I use? I tried to set up it with Linksys PAP-2 but with no luck.
Maybe I did something wrong with alarmreceiver.conf (I tried diverse
settings, but nothing worked).
I have a regular PRI from our CLEC and I *do* get blocked numebrs..
the bit is set to tell me to hide the number. I definately (as the
'phone company') want to be getting all call data for tracing
purposes, should we ever need it, but we can certainly honor that bit
and not display the number.
Actually, I have tested this here with an Aastra 9133i and an
[EMAIL PROTECTED] server, and the 9133i will re-subscribe on its own after
an Asterisk reboot, if you wait long enough. It took on the order of an
hour to do so. Of course, a phone reboot will get it done faster, if
necessary, but it
Hi,
After much searching I have found that it might be possible to get a
bluetooth headset to answer/hangup with SJPhone or Xlite if the headset
supports handsfree mode. My Toshiba bluetooth stack supports this but I
have not been able to figure out how to enable it. Also Windows XP
desktop
Anton Krall wrote:
what do you men adjust? (I guess you already tried the keys on the pad
right)?
On my system, when you watch ztmonitor on a channel, it is maxing out
the output volume, causing local side echo. Reducing the
tx.digital.handset gain bring the graph down to an acceptable
On 3/27/06, Matt [EMAIL PROTECTED] wrote:
I have a regular PRI from our CLEC and I *do* get blocked numebrs..
the bit is set to tell me to hide the number. I definately (as the
'phone company') want to be getting all call data for tracing
purposes, should we ever need it, but we can
I could ask why it can't authenticate against the key, but we've already been
there.
So, if I have 5 asterisk systems, and I want to have a different key on each,
and each system has a user and a peer section, and I have to use different
usernames... oh boy... this sounds like a horrible
Try replacing the XP Bluetooth stack with the widcomm drivers...google
is your friend!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chuck Bunn
Sent: Monday, March 27, 2006 6:21 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
On Monday 27 March 2006 08:58, Matt wrote:
Further, and here is where the legal question comes in. Is it legal to
'unblock' to the end user, that blocked number?Personally I feel
it SHOULD be.. after all it is my time I'm about to spend picking up a
phone and talking to someone, I want to
Hi,
Our asterisk installation will be a man-in-the-middle providing local,long,international VOIP services to our customers and our asterisk will be connect via VOIP to international carriers.
We use asterisk 1.2.5 with mysql in centos 4.2 Kernel 2.6
I have looked at astbill and it sounds
Who it is legal for or not to display those numbers is not realy the
point here, as in a law suit you will both (you and your provider) be
held liable. But the law clearly states that the end user should NOT
see that number if the number is blocked.
On 3/27/06, Andrew Kohlsmith [EMAIL PROTECTED]
Hello people.
I`m running asterisk 1.0.9.
In a phone call, I want to know who hangup, the caller or the callee.
It this posible?
Thanks in advance.
José Luis
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For the US PSTN network the limit seems to be 15 characters. For
Asterisk you can safely use 20 characters with most VOIP phones.
MATT---
On 3/27/06, Tomislav Parčina [EMAIL PROTECTED] wrote:
What is maximum length of name in caller ID? How much charters can I put and
be sure it will work
Hi,
I have the following requirement.. after a customer
is answered bya Queue, I want him to be redirected to another extensions,
where an IVR would answer and ask for his opinion about the analyst who just
solved his issue.
Is there a way to redirect him automatically, or do
I have to
On 3/27/06, Dov Bigio [EMAIL PROTECTED] wrote:
Hi,
I have the following requirement.. after a customer is answered by a Queue,
I want him to be redirected to another extensions, where an IVR would answer
and ask for his opinion about the analyst who just solved his issue.
Is there a way to
I'm curious (ok, well I admit it - it's for perosnal gain) what methods people
are using to manage asterisk config files when they have multiple asterisk
systems?
Some sort of revision control such as cvs,rcs or subversion?
A central 'config server' where you edit the files and then rsync them
What could have caused a system(on the same side of NAT on our LAN ) that have been working perfectly ie you can call and both parties can hear themselves very well to start having the problem described below (1) the caller can hear the other party very well ,but the other party hears cracked
I have the following python AGI script.
I know it's been abstracted, but it's still pretty easy to
see what's happening.
self.agi.channelAnswer()
self.agi.wait(1)
self.agi.execCmd(background,enter-conf-call-number,)
self.agi.execCmd(Read,confNum|||,)
I'm using CVS. I only have one server right now. I use it on other
clusters to sync files and it works for me..
On 3/27/06, Douglas Garstang [EMAIL PROTECTED] wrote:
I'm curious (ok, well I admit it - it's for perosnal gain) what methods
people are using to manage asterisk config files when
I have asterisk with rxfax txfax modules.I want
to test fax sendig and reciving in one asterisk
instance, in extensions.conf I have :
exten = 1234567,1,rxfax(/home/patryk/fax-new.tif|debug)
exten = s,1,Dial(1234567)
exten = s,2,txfax(/home/patryk/fax.tif|caller|debug)
but I doesn't seem to
FreePBX allows you to set up multiple companies as well as determine what
level of access each user has.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original Message-
From: [EMAIL
Just remember that a lot of email systems don't accept email from
unverifiable domains. If your using a domain for your Linux/Asterisk server
that does not resolve to a public IP then you may not be able to receive
voicemail to email.
I know that Hotmail WILL work no matter what so try that
I was playing with the fax stuff over IP on Friday. Unless you're
receiving faxes from a PSTN circuit, it doesn't work so well.
Also, I don't think you can chain txfax and rxfax like that. When you
hit the s,2 part, it's going to play the fax out to the handset you
dialed from. You'll need
I tried again and you are correct. It does work on the Aastra 9133i but
takes about an hour with no way to change that that I can find. The GXP2000
happens a lot sooner. I think it can be configured on the GXP2000.
Turns out the problem I had is that the Aastra 9133i does not resubscribe to
You could always use System() to copy a call spool file to launch the
outbound fax call. I don't really think a 3rd party app is necessary.
-Corey
On Mon, 27 Mar 2006, Gary Richardson wrote:
I was playing with the fax stuff over IP on Friday. Unless you're
receiving faxes from a PSTN
Hello All:
I used the Authenticate command against a list of 4 passwords, however is
there anyway I can get these to echo in CLI for debugging purposes?
My auth line looks like this:
exten = s,2,Authenticate(/home/listofnumbers|[|a])
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Jeremy wrote:
Hello All:
I used the Authenticate command against a list of 4 passwords, however is
there anyway I can get these to echo in CLI for debugging purposes?
My auth line looks like this:
exten = s,2,Authenticate(/home/listofnumbers|[|a])
show application NoOp
How does the hinting work on the polycoms? I've got a polycom set up with
hinting, I can see when the shared line rings, but I can't tell if
someone's on the line. Any suggestions?
--
Aaron Daniel
Computer Systems Technician
Sam Houston State University
[EMAIL PROTECTED]
(936) 294-4198
Can somebody send me a config of how to authorize SIP client by IP?
Sam
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Can Somebody send a working instruction to me on how to install g729 and
9723.1?
I could not open the
http://aussievoip.com.au/tiki-index.php?page=G729-Install
Thank you,
Goksie
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Peeters
I am using some Cisco
7940s with the 8.0 CM SIP image on them, and was wondering if there is a way to
have the caller ID display as just NAME number as opposed to NAME
[EMAIL PROTECTED].
The way it currently
is, the missed calls directory cant be dialed, and my users really want
this
On 3/27/06, Aaron Daniel [EMAIL PROTECTED] wrote:
How does the hinting work on the polycoms? I've got a polycom set up with
hinting, I can see when the shared line rings, but I can't tell if
someone's on the line. Any suggestions?
Shared lines still don't work with Asterisk on the
Does anyone know if FreePBX can be installed on a Linux box that was built
using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over
the AAH build. I have just not had good luck building an Asterisk system
from scratch and the Centos based Amp ISO and prebuilt config files are
http://www.readytechnology.co.uk/open/ipp-codecs-g729-g723.1/
I think this give a pretty good how to on installing the g729 and 723.
Sam
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of ADEGOKE ARUNA
Sent: Tuesday, March 28, 2006 12:55 AM
To: 'Asterisk
Yes, my mistake in /tftpboot/SEPMAC.cnf.xml. Having said that, Please
double check that you have set the line:
permit=192.168.1.90/255.255.255.255 ; This device can register only
using this ip address
or in your case:
permit=10.0.0.175 /255.255.255.255 ; This device can register
Yes, you can.
On 3/27/06, Jim Houser [EMAIL PROTECTED] wrote:
Does anyone know if FreePBX can be installed on a Linux box that was built
using [EMAIL PROTECTED] I would prefer to manage Asterisk with FreePBX over
the AAH build. I have just not had good luck building an Asterisk system
from
Worked fine for me. I did lose my MAINT link off the Portal, but I simply
added it back.
Thank you,
Jyran Glucky
Advisory Programmer
BlueWare, Inc.
Strategic HealthWare Solutions
3060 W. 13th Street
Cadillac, MI 49601
Phone: (231) 779-0224 ext. 111
Fax: 231-779-1002
Skype: Jyran Glucky
AIM:
Hi,
I am not having trouble with the bluetooth stack since the Toshiba stack
has the headset profile which supports a subset of AT commands
http://en.wikipedia.org/wiki/AT_command from GSM 07.07 for minimal
controls including the ability to ring, answer a call, hang up and
adjust the volume.
Hi
I have a litle question, what is then version stable, in
the web server i can see unicall version x.2.x and version
x.3.x, and the time is same
unicall-0.0.2e/ 11-Nov-2005 18:33 unicall-0.0.3pre8/
11-Nov-2005 18:37Where i can find the change log or the diference from this
Pardon the question, but what I understand of FreePBX is that it's
basically Asterisk with a web interface and some additional modules.
Is that correct? Can you install FreePBX on a system which ALREADY
has asterisk up and running or does it require ITS version of asterisk?
Thanks,
Waldo
OK, if I see well, this is the key idea here:
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
that is, putting the caller and callee number into AstDB under the CallBack
family.
Can you confirm that Asterisk takes care of the rest? If there is a record
like this in the database
My understanding is you can install it on any Linux server running Asterisk.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Waldo
Rubinstein
Sent: Monday, March 27, 2006 11:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
How can I edit the DB?
Tamás Bondár wrote:
OK, if I see well, this is the key idea here:
exten = 9,1,DBPut(CallBack/${${UNIQUEID}}=${CALLERIDNUM})
that is, putting the caller and callee number into AstDB under the CallBack
family.
Can you confirm that Asterisk takes care of the rest? If
FreePBX is a configuration manager for Asterisk. It is NOT its own version
of Asterisk, it is simply a GUI to manage the config files.
Kerry Garrison
Publisher - http://GeekGazette.com - http://VOIPSpeak.net
(949) 502-7819 x200 - [EMAIL PROTECTED]
http://www.techdatapros.com
-Original
On Friday 24 March 2006 16:05, Benoit Panizzon wrote:
Hi all
Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
http://bugs.digium.com/view.php?id=5884
Haven't tried it out yet.
I can now confirm: No freezes/crashes anymore since I applied the patch.
-Benoit-
Where can I do a keyword search of the posting in biz and users forums? asterisk.org just links to http://lists.digium.com/pipermail/and that doesn't let me do a string search across all postings.
thanks,
-- ---Erick PerezLinux User
I'm a bit newbie, could you tell me how to i apply the patch?
Thanks in advance
Marco Mouta
On 3/27/06, Benoit Panizzon [EMAIL PROTECTED] wrote:
On Friday 24 March 2006 16:05, Benoit Panizzon wrote:
Hi all
Apparently there is a patch for those 1.2.4/5 MeetMe Freezes:
Sorry for thread breaking... I'm on digest.
I'm curious (ok, well I admit it - it's for perosnal gain) what
methods people are using to manage asterisk config files when they
have multiple asterisk systems?
I'm using CVS. I only have one server right now. I use it on other
clusters to sync
Thanks for all the comments on the 3Com phones. Thankfully, there
is a large number of phones out there to dig through looking for the
right solution.
What I have not been able to find, after spending all weekend
looking, is a good solution for an attendant console. We have 2
http://www.google.com/search?sourceid=navclientie=UTF-8rls=GGLD,GGLD:2
004-48,GGLD:enq=apply+patch+linux
patch -p0 patch-file-name-here
Thanks,
Steve Totaro
http://www.asteriskhelpdesk.com
-Original Message-
From: Marco Mouta [mailto:[EMAIL PROTECTED]
Sent: Monday, March 27, 2006
On Monday 27 March 2006 20.07, Daniel wrote:
How can I edit the DB?
This may be a starting point for you:
http://www.voip-info.org/wiki/view/Asterisk+database
Or the related section of the book Asterisk: TFOT
Hi Erick -
Where can I do a keyword search of the posting in biz and users forums?
asterisk.org just
links to http://lists.digium.com/pipermail/ and that doesn't let me do a
string search across
all postings.
I'm guessing you mean the mailing lists rather than the forums. If
so, you can
We would be interested in the same. We have had only limited success
getting Snom's phones to do this. And, you're right, this is such a
common thing, there MUST be something out there that can do the job.
Darrell S. Long
BestWeb Corporation
Daniel Hazelbaker wrote:
Thanks for all
You could always use System() to copy a call spool file to launch the
outbound fax call. I don't really think a 3rd party app is necessary.
Could You explain this please? Or maybe some links to
documentation and examples ?
Thanks Patryk.
___
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Daniel
Hazelbaker
Sent: Monday, March 27, 2006 1:32 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Receptionist Phones (was 3Com Phones)
Thanks for all the
On Sun, Mar 26, 2006 at 08:03:55PM -0600, Darrick Hartman said:
Denis Galv?o - iSolve wrote:
The worst thing on all Polycom IP phones is the speaker phone's poor
quality. You could not have a conference call using the speakers, only
the head phone.
WHAT! The Polycom phones that have
The Asterisk Development Team is pleased to announce the release of
Asterisk 1.2.6 and Zaptel 1.2.5. Both of these releases include a number
of important bug fixes, and users are encouraged to upgrade their
systems when possible. See the included ChangeLog files for more details
on what has been
The Asterisk Development Team is pleased to announce the release of
Asterisk 1.2.6 and Zaptel 1.2.5. Both of these releases include a number
of important bug fixes, and users are encouraged to upgrade their
systems when possible. See the included ChangeLog files for more details
on what has been
Superb replies.
Thanks to Jon and Noah
On 3/27/06, Noah Miller [EMAIL PROTECTED] wrote:
Hi Erick - Where can I do a keyword search of the posting in biz and users forums?
asterisk.org just links to http://lists.digium.com/pipermail/ and that doesn't let me do a string search across all
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