I expect the issue is the same problem we have with the 360's.
Quick fix is add the old Snom MWI fix to your dial plan but its not perfect
solution for us
as all our phones with DDI present 6 digits and we have already created our
mailboxes to match
the 3 digit ext number which means the users
On 1/23/06, Doug Lytle [EMAIL PROTECTED] wrote:
Further, Polycom SIP phones have the longest boot time of any phone
I've ever seen (something like 5 min, compared to a Sipure, less than
Give a SIP based Cisco 79XX phone a try, just about as long in boot time.
Huh? My 7905 takes well under
I recommend to use the mass deployment feature to maintain your phones.
http://www.snom.com/wiki/index.php/Massdeployment_Firmware_Release_5
Besides this setting you cannot expect each setting is set by default to
exactly match your needs. Different environments different setup.
Best regards,
thanks buddyOn 1/23/06, Cameron Grant [EMAIL PROTECTED] wrote:
check /etc/asterisk/logger.confregards,cameron -- Forwarded message -- From: Angelito Manansala [EMAIL PROTECTED] To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com Date: Sun,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I have a small simple rule to start with!
Di you whant to have peiple calling in with sip to you domin then you
have to use SER as a SIP server, and ser as a connection to the telco world.
If you are only using the phones for PBX uses - use only
Hi Atif
make is a Unix's command which uses Makefile file for package's compilation.
So after installing the complete development package from distribution disk,
launch make.
Ciao
mauro
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Hi All,
I would like some clarification about licensing. Does this non-commericial
license provide me for usage inside my company (We're not telephony provider
so we are not using telephony services for making money). As I understood
from license agreement, I cannot use it?
Regards and thank you
On Mon, 2006-01-23 at 09:57 +0100, Dusko Tubin wrote:
Hi All,
I would like some clarification about licensing. Does this non-commericial
license provide me for usage inside my company (We're not telephony provider
so we are not using telephony services for making money). As I understood
from
How to set-up LCR ?
a. which companies can be used with LCR?
b. how to set-up maintain LCR?
c. multiple connection to one gateway?
Example:
+886223456789could be reachable via
a. ENUM free
b. Dundifree
c. Voipstunt free
d. Voipbuster free
e. Nufone $
f. Voipstunt $
g.
I am looking for a way to signup users and provide them with a file
which includes all settings, just to put somewhere.
Does something like that exist?
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To
*bump*
Anyone? I still can't find little/no info on DNID... :-/
Regards,
Evert
Evert Meulie wrote:
Hi all!
I'm in the process of configuring an Asterisk server here that, based on
which number was called, should send calls to different extensions:
913 - 1 - ext. 1
913 - 2 -
On Fri, Jan 20, 2006 at 09:20:43PM -0500, Michael Miller wrote:
I have over 50 Asterisk servers geographically distributed in pairs all
connected via DUNDi. Contact me off list and I will be happy to describe
my experience.
I'm also interested in knowing more of this. Why
not write to the list
Ronald Wiplinger a écrit :
How to set-up LCR ?
Easy!
sudo perl -MCPAN -e 'install Asterisk::LCR'
Then create a directory in which to work in, such as:
mkdir /tmp/lcr
Once you're in this directory, create a config file such as:
[comparer]
package = Asterisk::LCR::Comparer::XERAND
Odd you should have this problem as I had exactly the same. In my case
it was a slow DHCP server. Around 7 seconds in the phones tries to time
sync. If the phone hasn't got an IP address then this time sync fails
but it doesn't retry. I emailed Grandstream about it but got nowhere.
I changed
It's been about 2 months since I have updated my asterisk box.
I was running CVS HEAD and I notice a whole lot has changed since
my last update!
I'm running Debian Sarge up to date on a 2.4 Kernel.
I was updating about every 2 or 3 weeks and never had any problems
compiling
Bummer - Possibly a bug
The stable stuff compiles and runs fine :(
Steve
-
It's been about 2 months since I have updated my asterisk box.
I was running CVS HEAD and I notice a whole lot has changed since
my last update!
I'm running Debian Sarge up to date on a 2.4 Kernel.
I was
-BEGIN PGP SIGNED MESSAGE-
Hash: RIPEMD160
right!!! it is an unfortunate description...it does not have anything
to do with xml applications...
Hirosh
[EMAIL PROTECTED] wrote:
| On Thu, 19 Jan 2006, Hirosh Dabui wrote:
|
| [EMAIL PROTECTED] wrote: | On Wed, 18 Jan 2006, Hirosh Dabui
|
To reply to your rant the reason why I got this
account is because I am out of the US (in the us for
lists i used asterisk AT Dovid DOT net for the biz
list asteriskusers AT dovid DOT net for users list
etc). Here where I am I only have access to yahoo. I
needed a name that I could remember and
[Jan 23 19:56:44] -- Got SIP response 300 Multiple choice back
from 194.120.0.201
[Jan 23 19:56:44] -- Now forwarding SIP/601-fc4d to
'Local/194.120.0.211:5060,sip:194.221.62.211:5060,sip:80.239.235.211:[EMAIL PROTECTED]'
(thanks to SIP/voipstunt-5c8c)
[Jan 23 19:56:44] NOTICE[3439]:
As a general rule if the phone is behind NAT there
should be no issues. Server behind NAT = Lots of
issues (which can all be worked out). You will have to
specify NAT=YES in the dial plan.
Regards,
Dovid
--- Moises Silva [EMAIL PROTECTED] wrote:
you can redirect the ports of the router as
Hi,
I am new to asteriks,
ON Fedora Core 4
I installed asteriks with Asteriks Management Portal
İt seems working.
In the wen configuration panel i added a user and it seems
added but i can not connect from sjphone soft phone.
On the other way,
I start asteriks console and run sip
Technical Support wrote:
Check out www.generationd.com for their fax2mail and mail2fax scripts. It
might make life simpler
There is no description how to set-up!
The scripts are working with asterisk, but how?
bye
Ronald Wiplinger
-Original Message-
From: [EMAIL PROTECTED]
Please stop plugging the book. Its annoying. We know
its out there.
http://asteriskdocs.org deserves all mentions it receives and the
people behind it like Leif have done a great service to the community.
The entire book is still available online free so why stop plugging
it. For two years, the
I don't think you can beat the Polycom's for design, features, configuration
options and functionality tho. :)
Polycoms (I only have experience with a ip500) have many qualities.
However, I think it's only a matter of time before entries at the
$180-$200 price point begin beating it in many
hi
how can i debug with ser and use log() command in SER?
where it will log ?
thanks
--
Giti
Data products Trading Company
Mob : +971 508715610
Tel : +971 4 2973961
Fax : +971 4 2976404
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This has been discussed before. The decision was that manager in
Asterisk should *not* be XML. That's why we started to create the
AstManproxy that converts to XML.
I do believe that an XML formatted manager will help a lot of
developers, so having the option of both is a good thing.
Before
This has been discussed before. The decision was that manager in
Asterisk should *not* be XML. That's why we started to create the
AstManproxy that converts to XML.
I do believe that an XML formatted manager will help a lot of
developers, so having the option of both is a good thing.
Before
What is Native MoH, what file formats it has and how we use them. Is it
SLN files or GSM? How we enable Native MoH?
I've tried everything but my MP3 MoH is not going to work (very
distorted). GSM voice prompts play ok over the phones. I converted
fpm-calm-river.mp3 etc to GSM using sox, but
On Sunday 22 January 2006 14:11, Chris Mason wrote:
I am trying to construct a macro for long distance dialling. I have two
internet feeds, I have all routes including Teliax on Internet A and a
static route to Voxee on Internet B. I thought I could use the dialplan
entry below which uses the
Its ulaw.
sox -V foo.mp3 -t au -r 8000 -U -b -c 1 foo.ulaw resample -ql
is one way to get there.
You could also take a look at format_mp3 in asterisk-addons which is what I
use.
Chris
- Original Message -
From: Zach A [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List -
Sorry, I haven't received a message in a few hours, just testing to see if
it is alive.
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To UNSUBSCRIBE or update options visit:
Hi all,
Despite of www.openh323.org and some other sites claim the cvs has an empty
password for anonymous, I am unable to download the code from it. Any clue?
Logging in to :pserver:[EMAIL PROTECTED]:2401/cvsroot/openh323
CVS password:
cvs [login aborted]: reading from server:
On Fri, Jan 20, 2006 at 12:32:32PM -0500, Script Head wrote:
I am seeking to implement the following behavor:
When a headset on phone1 is picked up, phone2 rings right away, without any
need to dial numbers on phone1. Is this possible to implement?
Don't know about asterisk, but some
We conducted focus groups, looking at several different vendors, before we
decided to go with the Polycom. From the user interface perspective, the
Polycom's won hands down. I was never involved with it, but apparently to
configure the Cisco's you need to be converting hex??? Yuk!
Example:
log (L_INFO,test)
It will go to syslog, ie /var/log/messages.
Douglas.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Monday, January 23, 2006 6:48 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] debug with ser
hi
how can i
Douglas Garstang wrote:
We conducted focus groups, looking at several different vendors, before we
decided to go with the Polycom. From the user interface perspective, the
Polycom's won hands down. I was never involved with it, but apparently to
configure the Cisco's you need to be
Greg Boehnlein wrote:
(Steve Totaro wrote:)
What I would really like to do is have one D channel coming in on the T3
and have it split between each of the T1/PRI or even better one D
channel per quad (I know Asterisk can do that).
Is it possible?
No.
Actually, it is, using an Adtran
Greg Boehnlein wrote:
Hehehe.. Ask your Telco if they can provision E1 for you. ;) The Digium
cards can handle E1 or T1, and if you go E1 you'll get 30 channels instead
of 24 on the span.
I have talked to a number of telcos in the US about this... they don't
have the ability to do it.
Greg Oliver wrote:
I am unsure of * capabilities on NFAS (we do not use PCs to terminate
any PRIs), but it allows bonding of desparate PRIs to use a single
d-channel. ie, you can have 1 d-channel (optional backups) for the
entire DS3. Not sure if * can communicate across cards like that in the
Wilson Pickett wrote:
Please stop plugging the book. Its annoying. We know
its out there.
http://asteriskdocs.org deserves all mentions it receives and the
people behind it like Leif have done a great service to the community.
The entire book is still available online free so why stop
The problem is when reception is busy she doesn't always wait for
someone to answer the call, however hanging up a ringing transfer on
attended also hangs up the caller.
If you have enabled Disconnect Call feature, then you can hangup
with *0 for example, that will hangup only the current
Doug Lytle wrote:
Douglas Garstang wrote:
We conducted focus groups, looking at several different vendors,
before we decided to go with the Polycom. From the user interface
perspective, the Polycom's won hands down. I was never involved with
it, but apparently to configure the Cisco's you
I am trying to construct a macro for long distance dialling. I have
two internet feeds, I have all routes including Teliax on
Internet A
and a static route to Voxee on Internet B.
Here's an AEL macro I use on our boxes. Modify for your needs.
// dial a number with a range of routing
Andrew Furey wrote:
Huh? My 7905 takes well under 10 seconds, including Asterisk
registration and NTP update. Granted, if it were DHCP it might take
marginally longer, but 5 _minutes_?
Yeah, the Polycoms *do* take a while to boot -- but not five minutes.
I've timed mine (Polycom 501's) and
I have a quick Caller*ID question.
I have an inbound call to my PBX which I am attempting to bridge with
a PSTN number (specifically my cell phone, so when someone dials my
extension the cell phone rings).
In my extentions.conf I have:
; Daniel -- 1102
exten = 1102,1,Answer()
exten =
Kevin P. Fleming wrote:
Greg Boehnlein wrote:
Hehehe.. Ask your Telco if they can provision E1 for you. ;) The
Digium cards can handle E1 or T1, and if you go E1 you'll get 30
channels instead of 24 on the span.
I have talked to a number of telcos in the US about this... they don't
have
Steve Underwood wrote:
Actually every US made switch I've ever seen is 2.048MHz to the core.
They then rate change to the T1s. It makes export easier to handle. Any
mixof E1/2/3.. or T1/2/3.. cards will just plug in and go timing wise.
The issue is probably more of not being set up for mixed
Hi
iam not able to start asterisk
give me following error
any help
STARTING ASTERISK/usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core dumped) ${ASTSBINDIR}/asterisk ${CLIARGS} ${ASTARGS} /dev/${TTY} /dev/${TTY}Asterisk ended with exit status 132
Asterisk exited on signal
OK, so if I were using SVN, the stable branch would still be changing and my
problem was that I was using the Tarballs? Correct?
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of having
a better past.
---- --- - - - -- - - --
My first guess would be to
Use 'tail /var/log/asterisk/full' to find out why.
On 1/23/06, ram [EMAIL PROTECTED] wrote:
Hi
iam not able to start asterisk
give me following error
any help
STARTING ASTERISK
/usr/sbin/safe_asterisk: line 42: 4633 Illegal instruction (core
dumped)
What I did so
far:
My
[EMAIL PROTECTED] PBX is working fine, with four Grandstream Budge Tone –100
phones.
Regarding the
MoH feature, I did the following:
Checked the presence of mpg123. It is.
2.
In
/etc/asterisk/zapata.conf, I added the line "musiconhold=default" under
Hello Asterisk Community.
While sitting at lunch the other day I had a typical napkin-prototype idea:
What if I could make my Asterisk Voicemail accessible as a Podcast in iTunes?
Three hours later with the help of two friends I had a working proof of
concept. Now we are releasing the
At 05:06 AM 01/23/2006, you wrote:
http://asteriskdocs.org deserves all mentions it receives and the
Though you really should mention that it's a 1.0 document and trying
to make a 1.2 installation work using that book is somewhat futile.
Ira
I have the same issue. I just bought the commercial version from Digium
to see if that has the same problem. I wanted to use the free one to
test out g729. My Polycom 301 had no issues using the free codec though
(testing via VM, etc)
Bill
-Original Message-
From: [EMAIL PROTECTED]
Hi,
I know it is a FAQ but I'm interested in latest news (if any...) about SIP
over TCP support in Asterisk.
I found this:
https://savannah.nongnu.org/projects/asterisk-tcp/
but I'm not able to understand if project is active and what is its level of
development.
Thanks
Mimmus
In article [EMAIL PROTECTED],
Steven [EMAIL PROTECTED] wrote:
OK, so if I were using SVN, the stable branch would still be changing and my
problem was
that I was using the Tarballs? Correct?
I guess so. The stable branch (now branches/1.2) has bug fixes applied
to it as they get done. Every
Hello,
I am wondering about the ability of a server that is simply passing G729
through it to have the ability to record the calls. I know for
voicemail, meetme, and things like that to work, a G729 license must be
installed on the machine since there is transcoding going on.
Is this also
On 1/23/06, Koopmann, Jan-Peter [EMAIL PROTECTED] wrote:
On Saturday, January 21, 2006 8:02 PM Alexander Chemeris wrote:
What is the problem with step 3?
See this example as basis for modifications:
http://www.voip-info.org/wiki/view/Asterisk+Dynamic+conferences+macro
Unless I have
Same thing...even with the commercial Digium G729 codec. I have to
specifiy G729br8 on the Cisco.
Cisco issue?
Bill
-Original Message-
From: Bill Gibbs
Sent: Monday, January 23, 2006 12:01 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users]
On 08:08, Mon 23 Jan 06, Alex Barnes wrote:
I expect the issue is the same problem we have with the 360's.
Quick fix is add the old Snom MWI fix to your dial plan but its not perfect
solution for us
as all our phones with DDI present 6 digits and we have already created our
mailboxes to
there is no File called that name in that place
that is the reason i have mailed here
ram
On 1/23/06, John Broome [EMAIL PROTECTED] wrote:
My first guess would be toUse 'tail /var/log/asterisk/full' to find out why.On 1/23/06, ram
[EMAIL PROTECTED] wrote: Hi iam not able to start asterisk give
Hi All,
I configured Asterisk and it is working successfully with Express Talk. Now I am
trying to work with some other client which supports only GSM and now Asterisk
never worked and tried to make a call out. In sip.conf I disallowed all and
allowed only GSM also. I also heard that Asterisk
I am trying to build an silent non moving parts (fans,HD.etc) embedded
system...Firewall/Asterisk/FXo/FXs/CF/etc
Looking for anyone running asterisk with Coyote, IPcop, m0n0wal, Shorewall,
etc in the same system/box!!!
Offlist please...
Thanks in advance!!
Manny
Hello,
Has anyone used Polycom's VSX line of videoconferencing equipment with
Asterisk?
It seems some of their models, namely the newer VSX 5000, supports SIP.
--
The Internet used to be a lot of smart people sitting at dumb terminals,
but now its a lot of dumb people sitting at smart
Ive had problems for the last
couple of weeks regarding incoming calls. Cant hear the party calling me (their
voice sounds garbled/scrambled). If you havent done so yet, I would
recommend you post your complaint on their online forum as well under bugs.
You usually get some good responses
Hi everybody!
I'm from Argentina, so you'll have to sorry me for my English.
I have a Linux box with asterisk and want to buy an ATA.
Fist, I thought about the Grandstream HandyTone but I read some
reviews which says that it has a lot of echo. Some people recommended
me Sipura 2000 but I don't
we hear you loud and clear
2006/1/23, [EMAIL PROTECTED] [EMAIL PROTECTED]:
Sorry, I haven't received a message in a few hours, just testing to see if
it is alive.
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On 01/23/2006 08:55:09 AM, Omadon wrote:
On Fri, Jan 20, 2006 at 12:32:32PM -0500, Script Head wrote:
I am seeking to implement the following behavor:
When a headset on phone1 is picked up, phone2 rings right away,
without any
need to dial numbers on phone1. Is this possible to implement?
I hate to burst your bubble but DOS attacks are a fact of life for IP based services. The bigger you get the more of a target you are. There are a ton of DOS prevention/mitigation appliances/services available in today's world. But they all rely on the same thing: having more bandwidth/capacity
Hi there,Can anyone know how to view asterisk disconnect code.?-- Best Regards,Angelito Manansalawww.voicefidelity.netMobile: +63 917 542 5807
DID: (+63) 44 7906770US DID: +1 619 399 0128msn: [EMAIL PROTECTED]skype: bulcrack
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There seems to be a queue of some sort the messages fit through. No
matter how long it seems to take, all messages I've sent get through,
even 12+ hours later. I've done my share of double-postings and have
learned to wait ;) I haven't discerned anything broken with the list,
just slow maybe
Hi,
Does any body knows some thing about it?
Thanks in advance.
Telles
Rodrigo P. Telles wrote:
Hi Folks,
I've been searching for an specific feature on asterisk and I found an e-mail
from John Todd asking for the same thing.
Hello:
I´ll like to know if asterisk is capable of making
H.323 videoconferencing and if it can also transcode fromH.323 to SIP
Any help will be appreciate
Tanks
Erick W.
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Asterisk-Users
Hi,
If I have an AGI script that calls user A and then calls user B and
connects them... it seems to work fine (for accounting) if I call a
local call (out my PRI).. however if I go out my IAX... the CDR
terminates the long distance call after 3 seconds (after the IAX trunk
picks up).. and what
Anybody seen some client/server asterisk add-on
script for "live" answering services to provide call handling and message taking
from an Operator?
Bart
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To
Hi,
Try
exten = h,1,NoOp(${HANGUPCAUSE})
in your extensions.conf
Cheers.
Andy
On 1/23/06, Angelito Manansala [EMAIL PROTECTED] wrote:
Hi there,
Can anyone know how to view asterisk disconnect code.?
--
Best Regards,
Angelito Manansala
www.voicefidelity.net
Mobile: +63 917 542 5807
Not sure what you mean but a basic PBX does what I have read.
-Original Message-
From: Bart Fisher
Sent: Mon 1/23/2006 1:47 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Cc:
Subject: [Asterisk-Users] Answering Service
Hi,
According to the wiki, we need to have both callwaiting=yes and
callwaitingcallerid=yes , and that's what I have in zapata.conf.
I can hear the call waiting alert tone when a 2nd call comes in during
an established call, and I can switch between the calls without
problems. However, CallerID
Yep I did the same.
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Francesco
Peeters (Asterisk)
Sent: Saturday, 21 January 2006 5:34 PM
To: [EMAIL PROTECTED]; Asterisk Users Mailing List - Non-Commercial Discussion
Cc: Asterisk Users Mailing List -
Hola Facundo, saludos desde Panama.
If you're running asterisk at home or some other asterisk project and
you're only concerned about the ATA, well, a HT-286 (entry level,
cheap) is a good start. Yes, there are reported issues with the
GrandStream equipment but all the others have issues too (ok
I have a doubt... is it posible to do Video Conferencing using asterisk?
--
Facundo Ameal.
famealatgmaildotcom
Linux User #395088
Open your mind, use open source.
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look at this:
http://www.voip-info.org/wiki-VICIDIAL+Dialer
perhaps it's what you are looking for...
2006/1/23, Danish Samad [EMAIL PROTECTED]:
Hi,
In a normal PBX environment a user usually calls in and IVR's are played
according to a predefined dialplan.
Iam trying to develop an
The option for MeetMe() to record(r) the conference does not seem to be working.
I see a CLI message that it is starting recording, however no file is ever
created. No error or warnings messages are seen either.
Starting recording of MeetMe Conference 100 into file
Erick Muchas Gracias por la respuesta.
I'm not using any of that projects, it's my own Asterisk installation
onto slackware 10.
well what can you tell about sipura ones?
2006/1/23, Erick Perez [EMAIL PROTECTED]:
Hola Facundo, saludos desde Panama.
If you're running asterisk at home or some
Hi,
In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan.
Iam trying to develop an application where asterisk dials out to a user
and initiates an IVR instead (please note that the IVR is not static
and may vary according to different
An example of this would be Outcall Voice Mail?
Mark, G7LTT/KC2ENI
Randolph, NJ
http://www.g7ltt.com
Danish Samad wrote:
Hi,
In a normal PBX environment a user usually calls in and IVR's are
played according to a predefined dialplan.
Iam trying to develop an application where asterisk
Hi,
I installed Asterisk yesterden with amportal,
I added 2 sip extensions, and it is seen in mysql too.
But when i try to register from any device or softphones,
invalid username/secret message comes,
İn tne cl, i us esip show users -- no users
But user is in mysql db
What can i do
Danish Samad wrote:
Hi,
In a normal PBX environment a user usually calls in and IVR's are
played according to a predefined dialplan.
Iam trying to develop an application where asterisk dials out to a
user and initiates an IVR instead (please note that the IVR is not
static and may vary
It's possible to do point to point but not point to multipoint.
I tried to get development for this some time ago and no one responded,
check out my Video Conference Bounty on www.voip-info.org, since then we
have developed our own solution that we have decided to market, it will
cost $2,000 for
I haven't worked with sipura. So I can't write about it. If I stick to
the reviews, then it is a good/stable product with some
minor/strange/rarely-ocurred issues regarding phantom calls.
spanish-onno creas que no hablo español, pero sabes que aqui solo
puedes postear en ingles no?spanish-off
On
Hi,
I search in the archives and I don't find that
case.
I'm wanted todo Asterisk+spandsp working. I
have installed spandsp and apply the patch without any errors. I have recompiled
Asterisk and When I try to start it, the output say :
[app_txfax.so]Jan 23 15:17:12 WARNING[3022]:
On Tue, 24 Jan 2006, Danish Samad wrote:
Hi,
-users questions
In a normal PBX environment a user usually calls in and IVR's are played
according to a predefined dialplan.
Iam trying to develop an application where asterisk dials out to a user and
initiates an IVR instead (please note
The party's over folks, the new official cisco/linksys/sipura policy is to
no longer sell SPA-3000's to end users.
Buy them while you still can :-(
-Dan
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To
Cute?
But it can use LDAP...
PaulH
- Original Message -
From: Ben Klang [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 24, 2006 3:58 AM
Subject: [Asterisk-Users] Announcing PodMail 1.0 (GPL)
Hello
Is it possible to background SayDigits()?
I know you can manually Background() each digit individually, but this
does not solve the problem when you need to do something like
SayDigits(${EXTEN}) or SayDigits(${CALLERID(number)})
-Dan
___
We have sold thousands of these with no reports of echo problems. Perhaps
the reviews were referring to a different Grandstream product? Some of the
phones have had some echo issues. BTW, the Sipura 2000 has been replaced by
the 2002.
Michael Crown
Managing Partner
www.thevoipconnection.com
Support Internet.net wrote:
I'm wanted to do Asterisk+spandsp working. I have installed spandsp
and apply the patch without any errors. I have recompiled Asterisk and
When I try to start it, the output say :
[app_txfax.so]Jan 23 15:17:12 WARNING[3022]: loader.c:325
__load_resource:
Greetings fellow list members,
I am trying to add some tricky functionality to
Asterisk dialplan and I was curious if anyone else has come up with a solution
to something like this.
Basically I have phone representatives that log
into one of several queues (not using chan Agent, welog inby
What
version of SpanDSP are you running? You should be running
-pre21
-Original Message-From: Support Internet.net
[mailto:[EMAIL PROTECTED]Sent: Monday, January 23, 2006 1:18
PMTo: asterisk-users@lists.digium.comSubject:
[Asterisk-Users] app_rxfax.so and app_txfax.so
Hi,I've got a CCM ( Cisco Call Manager ), with a Cisco Unity VM server and about 45 SCCP phones on the ccm, and 200 users on unity. we want to migrate all users to IP Phones to ditch our ancient phone system. I would love to get Linksys-Sipura SPA-941s for the 150 users not on IP phones yet and
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