Hi,
We are the American distributors for these phones. The link is below.
http://ipphone.eezeephone.com
AT723 is discontinued. A 2 Port ATA and a 4 Port ATA is on the cards very
soon.
Seshu Kanuri
Netweb Group, Inc.
Ph:1-732-387-4133
[EMAIL PROTECTED]
www.netwebgroup.com
"This e-mail message ma
On Tue, 21 Sep 2004 14:35:02 -0600, James Sizemore <[EMAIL PROTECTED]> wrote:
> I have noticed a problem with the Cisco 7940/7960 phones where if
> you put your voice-mail box on hold using soft keys and come back
> you can no longer navigate. I am curious if anyone else can
> duplicate this proble
Try nat=route to correct the rport issue mentioned earlier. I'm told
by sources within Uniden that the firmware supporting STUN will be
released soon.
-Curt
On Tue, 21 Sep 2004 22:44:34 -0600, Ryan Courtnage <[EMAIL PROTECTED]> wrote:
> Lyle,
>
> If you are behind NAT, and * isn't, I'm afraid I
Lyle,
If you are behind NAT, and * isn't, I'm afraid I have some bad news for you.
According to Uniden, STUN support is a "Feature Under Development".
To furthur complicate things for you, the UIP200 currently does not
respond (at all) to an INVITE that has 'rport' in the SIP Via field. In
other
Keep in mind, PPTP will only tunnel through the NAT, as long as GRE (prot
47) is properly tunneled along with tcp 1723. This support is relatively
standard in common NATs, but it's not a given.
-denon
At 09:23 PM 9/21/2004, you wrote:
First,
I assume that you will be running NAT at both locations
On Tue, 21 Sep 2004 21:49:33 -0400, William Suffill
<[EMAIL PROTECTED]> wrote:
> Good idea Matt. Tad far for you unfortunately and too costly for me at
> this time but hearing all the latest and greatest news would be
> supper.
News for supper and images for dinner
beautiful
:)
As I understand it we can either stay at the Marquis for the duration or
try again tomorrow to move to the CC. I'm probably not going to go
through the hassle of packing back up and moving over to the CC,
especially since I already paid for the STSN net access for the duration
of my stay here. :) I
What are they going to do about that anyway?
- Original Message -
From: "Joshua M. Thompson" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL PROTECTED]>
Sent: Tuesday, September 21, 2004 10:19 PM
Subject: Re: [Asterisk-Users] Astricon meets?
> M
On Wed, 2004-09-22 at 00:19 -0400, Joshua M. Thompson wrote:
> More importantly, who else is at the "other" here...the Marriot Marquis.
> I was one of the unlucky ones who didn't get a room at the Century
> Center.
Oh and on a related note...I'm here, I'm bored, and I'm hungry. If
anyone else is i
More importantly, who else is at the "other" here...the Marriot Marquis.
I was one of the unlucky ones who didn't get a room at the Century
Center.
If there is anyone here, is anyone heading back to the Century Center at
a more reasonable hour who has their own transportation or who wants to
split
I've figured out the problem my flash length was too long...
I have set flash=100 in zapata.conf
and this in extensions.conf
ignorepat=3
exten=> _3XX,1,Flash()
exten=> _3XX,2,SendFTMF,${EXTEN:1}
exten=> _3XX,3,Flash()
exten=> _3XX,4,Hangup
all works now :)
On Wed, 22 Sep 2004 11:48:24 +1000
If you have used the FWD Assistant, it will have configured your
Asterisk server to use FWD through IAX, not SIP. Since you seem to be
behind NAT and a firewall, using IAX is a good idea anyway. IAX is NAT
friendly, SIP is not.
I have decided to use the IAX protocol enabled by the FWD Assistant
in
On Tue, 21 Sep 2004 19:54:12 -0500, Joe Woss <[EMAIL PROTECTED]> wrote:
> > X-Lite configuration:
> > Menu | System Settings | SIP Proxy | default
> > Display Name: Full Name
> > User Name & Authorization User:
> > Password:
> > Domain/Realm: x.x.x.x
> > SIP Proxy: x.x.x.x
> >
> This is from the
On Tue, 21 Sep 2004 20:40:38 -0400, Kenton Powell <[EMAIL PROTECTED]> wrote:
> I'm new to Asterisk and have installed it on OS X. I am trying to use
> an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and
> am not having any luck.
>
> I have looked at several settings pages at vo
First, I assume that you will be running NAT at
both locations, if that is not the case, then the configuration will
change.
When you said VPN, are you using PPTP or IPSEC?
Microsoft supports PPTP. In order to connect a PC over VPN to the office, which
has a PPTP VPN Server, you will need
Shawn,
I am running ISA 2004. As you know this
firewall is not SIP aware. I’ve spoken with MS LCS devs and they don’t
know of any SIP filters on the horizon. As far as IAX… not sure why you
would be having problems.
In our environment we have an ISA 2004
firewall at the central offi
How do you know which card is loading first and
which is loading second? And on the tdm33b, module 1 is closest to the
rj45 jacks and 4 is at the open end of the card.
Lyle
- Original Message -
From:
Carlos Medina
To: [EMAIL PROTECTED]
Sent: Monday, September 20,
Good idea Matt. Tad far for you unfortunately and too costly for me at
this time but hearing all the latest and greatest news would be
supper.
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Hi there,
thanks for that.
I'm using x-lite softphone, can I send a flash to the x100p from xlite
directly? or do I need to dial an extension and have asterisk do it?
any examples...?
cheers
Adam
On Thu, 16 Sep 2004 10:14:39 -0300, Marcelo Pacheco <[EMAIL PROTECTED]> wrote:
> The flash applica
On Tuesday 21 September 2004 19:46, jay wilton wrote:
> I have been getting outbound nufone calls dropped
> after about 70 seconds. CLI shows "Attempting native
> bridge of IAX2". I have put "notransfer=yes" in
> iax.conf in the main section and all identifier
> sections.
After 70 seconds? Upgr
Hello,
I have been getting outbound nufone calls dropped
after about 70 seconds. CLI shows "Attempting native
bridge of IAX2". I have put "notransfer=yes" in
iax.conf in the main section and all identifier
sections.
I tried a Tt in the dialstring, but it still tries the
bridge. a cvs update
I got a Uniden UIP200 and started to configure it and I am lost
I have a tftp server setup on my * server and have the files unidencom.txt
and uniden.txt there. But it doesn't quite work yet. It registers as
a sip phone (sip show peers), but I cann't dial it and the display shows #1
disconn
I am new to the Asterisk-users Mailing list, but i assume that you
post your answers to the group?
Kenton,
>
> X-Lite configuration:
> Menu | System Settings | SIP Proxy | default
> Display Name: Full Name
> User Name & Authorization User:
> Password:
> Domain/Realm: x.x.x.x
> SIP Proxy: x.x.x.
I'm new to Asterisk and have installed it on OS X. I am trying to use
an X-Lite softphone to connect through Asterisk to FreeWorld Dialup and
am not having any luck.
I have looked at several settings pages at voip-info.com but have had
no luck. I am on a school network which I assume has some s
I am trying to install the astVICIDIAL_0.7.pl on a working asterisk box. I
keep getting the following error.
Time::HiRes object version 1.59 does not match $Time::HiRes::XS_VERSION 1.55
at /usr/local/ActivePerl-5.8/lib/5.8.4/i686-linux-thread-multi/DynaLoader.pm
line 253.
Compilation failed in
>
>
> On Tue, 21 Sep 2004, Steve Kann wrote:
>
>> Is there a mailing list for people at astricon?
>>
>> I'm here, at the conference Hotel, and was wondering if others as well.
>>Doing some work in the Hotel, but I'd be interested in meeting some
>> others in the Lobby for *-talk or something..
No worries. You are totally right that $8 bucks is pretty neglibable
for that service.
Your comment about the vendor made me think of a fellow I recently
helped with his * who resells Cisco.
His company is called TekSavers and they have the phone that started
this email (Cisco 7905G) for $95.
I
Setup:
System Phones <--> PBX <--E1--> * <--> SIP Phones
Calls work in both directions. However, ringing feedback to caller only
works in the SIP->System direction. System callers to a SIP endpoint get
silence, until call is picked up or dropped into something else.
I've tried both Dial() with
Looked at one of these phones about a month ago (the ATA 323, haven't seen
the ATA 723), the base unit is very light and the rubber stoppers are crap
so the phone slides across the desk whenever you pickup the handset. There
was no visual MWI (message waiting indicator) on the phone and the defaul
On Tue, 21 Sep 2004, Steve Kann wrote:
> Is there a mailing list for people at astricon?
>
> I'm here, at the conference Hotel, and was wondering if others as well.
>Doing some work in the Hotel, but I'd be interested in meeting some
> others in the Lobby for *-talk or something..
Well
--On Tuesday, September 21, 2004 15:58 -0700 "Wiley E. Siler"
<[EMAIL PROTECTED]> wrote:
A completely valid point you make...
Just remember to multiply that 8 dollars times the number of phones
times the number of years you need to have them in service
Extend that to the other Cisco items yo
Miguel,
I don't know about the Gatekeeper, but if you install Asterisk with
the sample configuration files, then it will be already setup to have
a few extensions, along with the ability to dial out through the IAX
protocol to Digium.
That's how I started with Asterisk too.
Regards,
Craig Fo
A completely valid point you make...
Just remember to multiply that 8 dollars times the number of phones
times the number of years you need to have them in service
Extend that to the other Cisco items you need to maintain and purchase
and the costs keeps rising
Mine is just a simple obse
I'm pretty sure SMARTnet on the VoIP phones is like $8/phone/yearMost
orgs spend more than that per person on electricity for CRTs.
--On Tuesday, September 21, 2004 15:25 -0700 "Wiley E. Siler"
<[EMAIL PROTECTED]> wrote:
See my post of a few moments ago and you have hit on the exact reason
See my post of a few moments ago and you have hit on the exact reason I
will not use Cisco beyond my firewall (a purchase you will never regret
if you need a good firewall). Cisco makes arguably some of (if not
totally) the best equipment out there. I just have one problem.
Their licensing model
Hi All
Just received my first 7905G from a distributer here in Sweden. According to the spec
this phone should be able to use SIP. Now I been looking on Ciscos home pages for
several hours trying to find a "SIP image" for this phone.
No luck at all, need special access to be able to download sof
Her is the 7905-12 page
http://www.cisco.com/cgi-bin/tablebuild.pl/ip-phone-7905
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Vallee
Sent: Tuesday, September 21, 2004 4:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subje
Hello again.
I'm stll struggling trying to terminate calls from SIP through Asterisk
and throught my H323 gateways...
Basically the call is accepted by GnuGK but then dropped with
reason = unreachableDestination <>
I did a debug trc 10 on GnuGK and looked at the sessions... one
from X-Lite
I use my XLite softphone from my Win XP box over VPN to my
Cisco PIX with no issues so this can be done.
How that works for an ISA box is unknown to me. I
dumped ISA several years ago do to it's (IMHO) unpredictability and low
performance.
Are you using the built in VPN of WinXP or an ISA
On 21 Sep 2004 at 15:40, Kristian Kielhofner wrote:
> Hey,
>
> I am here at Astricon and about to go down to registration. Is there
> any
> interest in pictures if I take my digital camera? I am sure that
> someone is already doing this. (Probably someone official). I would
> take pictures o
Use this url:
If you have a valid userid.
http://www.cisco.com/kobayashi/sw-center/sw-voice.shtml
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Christian
Vallee
Sent: Tuesday, September 21, 2004 4:27 PM
To: 'Asterisk Users Mailing List - Non-Commercial D
I have just finished compiling and installing Asterisk on a
test Debian system. All is working well. We are now attempting to get remote
offices to test the system I have installed both a SIP and an IAX client at a
remote office. Then I connect to our office via Microsoft ISA firewall and t
Hi, I don't know if it the only way to do that, but it is how I did it.
You need to have a valid cco account and your Smartnet contract have to be
associated with your cco login.
Next go to
http://www.cisco.com/public/sw-center/
and when you are logged in, you can find the firmware of your choic
Someone from Belgium is using my wireless stuff now, so I guess it does
work :)
Trying to use speedtest.nl.secureax.be -- dunno if I appreciate "speed
tests" though :)
-SteveK
On Sep 21, 2004, at 3:52 PM, Steve Kann wrote:
The STSN devices are here in the rooms. I'm currently "sharing" the
co
Hi,
Anybody know if I can register my Asterisk in more
than one h323 Gatekeeper.
I need to call to diferents providers depending on
convenients destinations prices.
Thanks!!!
--
Este mensaje ha sido analizado por Antivirus de Grupo Alternativa S.A.
en busca de virus y otros contenidos peligr
their permission might be a good idea too =) Don't want anyone to get
hostile when you show the pics to the community.
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On Tuesday 21 September 2004 16:40, Kristian Kielhofner wrote:
> I am here at Astricon and about to go down to registration. Is there any
> interest in pictures if I take my digital camera? I am sure that someone
> is already doing this. (Probably someone official). I would take
> pictures
Hey,
I am here at Astricon and about to go down to registration. Is there any
interest in pictures if I take my digital camera? I am sure that someone
is already doing this. (Probably someone official). I would take
pictures of each day and upload them to my website if anyone is
interes
Hello!
I am semi-new to asterisk, I've been toying with it for about a month
now. I'm using the current version from the CVS server (as of last
week) and have succesfully connected it to Broadvoice, as well as a
few softphones. I got a Nortel i2004 IP phone from a friend to setup
at home as a hom
Hi people, is there any good document on setup asterisk using only its voip
functions , i mean without e1/t1 hardware, i have a cisco 5400 gateway and a
cisco 7000 working as gatekeeper, can asterisk register with this GK?, im
thinking in using cisco voip phones
Any pointer will be apreciated, and
Hi there,
I just recently installed asterisk from cvs on SuSE 8.2 successfully. I was
quite surprised since asterisk seems to be a quite complex and powerful
application. I also installed chan_capi 0.3.5 and am using a Fritz!PCI card at
an ISDN line (PtMP).
I already got some basic extension sch
kevin, i will give the latest cvs of asterisk (and libpri / zaptel ) as
seems good practice
would you be happy to share with me (off topic if necessary) your
zapata.conf (for X100P ??)
GT
- Original Message -
From: "Kevin Walsh" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non
The STSN devices are here in the rooms. I'm currently "sharing" the
connection via my powerbook. Dunno if anyone will be able to reach it
from other rooms or not...
It is $9.99/day, and when you buy it, you apparently also get unlimited
local/LD telephone calls, so, if your SIP device doesn't
I have noticed a problem with the Cisco 7940/7960 phones where if
you put your voice-mail box on hold using soft keys and come back
you can no longer navigate. I am curious if anyone else can
duplicate this problem. Happens reliably for me with the 7940
phones.
I use rfc2833 for DTMF. I would
Is there a mailing list for people at astricon?
I'm here, at the conference Hotel, and was wondering if others as well.
Doing some work in the Hotel, but I'd be interested in meeting some
others in the Lobby for *-talk or something..
-SteveK
___
Aste
It's benign,
From memory, I think it happens when a channel calls ast_read on an
iax2 channel, and is waiting on the scheduler, and gets interrupted.
I think that probably the message should just be removed or something.
On Sep 20, 2004, at 9:38 AM, Mike Taht wrote:
I've been testing a new aster
Here is the best starting point. It is all driven out
of extentions.conf
http://www.voip-info.org/tiki-index.php?page=Asterisk%20tips%20ivr%20menu
Search for IVR and you will find good
info...
Cheers,
W
From: Luis Czop [mailto:[EMAIL PROTECTED]
Sent: Tuesday, September 21, 2004 1
Hi friends,
Does anyone know
where can I find an "How To" to config the auto attendant?
Anything
else?
Many thanks in
advance
Luis Eduardo Czop
Gte. de Tecnología y Servicios
PMS Argentina SA
Av. Alicia M. de Justo 170 - Piso 1°
(C1107AAD) Ciudad de Buenos Aires
Tel: 5217 9311
_
Yes my registration is also set to 60 sec. I'm going to upgrade them to 2.10
and see what happens.
Thanks for your reply demitri.
Regards,
Mohammed Salim
EZZI Telecom, Inc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of reseaux
Sent: Tuesday, Septemb
On Tue, Sep 21, 2004 at 09:00:18PM +0300, Vlasis Chatzistayrou wrote:
> Hello,
>
> I have been wrestling with installing the CAPI drivers for AVM Fritz in order
> to use chan_capi with Asterisk.
>
> I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers
> (namely: capi4k-utils-
I use the following extension in my dial plan to pickup the ZAP
channel but not dial anything.
This works very well and allows me to pickup a call from a VOIP phone
after the call has already been answered by someone else in the house
on the POTS line.
exten => ,1,Dial,ZAP/1 ; Pickup
Anyone know of a Canadian dealer (not reseller, I only need one unit) of
these asterisk compatible phones?
TIA,
Matt
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Hello all,
I feel dumb asking this, but does anyone have a link to the SIP
firmware for the 7912 on Cisco's site?
I have a SmartNet contract, but I just can't find the link (you can
search for "7960 sip firmware" and find that fast).
Thanks for the help,
Jeb Campbell
[EMAIL PROTECTED]
__
Title: Message
Does anyone have a
working iax1.conf for Gnophone (the softphone version) of www.voiceglo.com?
Thanks
__
Stig
Hess
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I am sorry if this seems like an overly obvious question but is this
patch available on mantis? I am having the same problem and wasn't aware
a patch was available at all.
Thank you,
Jamie Ferrara
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian
Cu
Dear Mohammed
I have notice the same problem of UNREGISTRATION of my Sipura 2000 i have
some of this with R2.07 and other with 2.09 i also try to put registration
expire to 60 sec but seems the same i try to made more debug on this..
Bye
Dimitri
On Tuesday 21 September 2004 17:17, Mohammed Sal
Hello,
I have been wrestling with installing the CAPI drivers for AVM Fritz in order
to use chan_capi with Asterisk.
I use an SMP machine, RH 9. I have found rpm's for CAPI and AVM drivers
(namely: capi4k-utils-2003.06.16-08.mungo.RH9.i686.rpm and kernel-2.4.20-8-
avmfcpci-03.11.02-08.mungo.RH9
Nevermind, I had to upgrade to 1.4.2 first then 1.5
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matthew
Marlowe
Sent: Tuesday, September 21, 2004 1:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Polycom IP
Is anybody familiar with these IP phones AT-723 or AT-323
I think it is made by this company:
http://www.atcom.com.cn/at723E.html
--
#Joseph
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On Tue, 21 Sep 2004, Andrew Thompson wrote:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Look a few blocks into the examples section, you need an underscore in
there.
When I saw this message, I realized that I goofed in my example for the
.call file earlier to
On Sun, 19 Sep 2004 09:40:19 -0400, Austin M. Brower
<[EMAIL PROTECTED]> wrote:
> Paul,
>However, this bug:
>http://bugs.digium.com/bug_view_page.php?bug_id=0001693
> may provide the tools we need to work around this problem, namely, not
> putting Agents statically into queues.conf,
Hi,
I'm trying to get a Zyxel P2000W (reportedly also sold as WiSIP by Pulver)
to work with an asterisk box.
The phone connects nicely to an external VoIP company (sipgate.de
reportedly using asterisk themselves) but there is a strange problem with
my asterisk:
- Incoming calls via ISDN (chan_c
I've had an IP300 for a while now and it's been working fine. I just
got an IP500 and when it connects to the FTP server it downloads the new
bootrom and says error loading.
The bootrom is fine and works on the 300... In addition, I downloaded a
new copy to be sure and it still doesn't work.
C
Hello All,
I am planning on setting up an * server for a customer and was hoping to get
a sanity check on my Plan. What I am trying to accomplish is a * voice and
16 data channel T-1 connection (ESF/B8ZS). I am planning on using a 2.8 ghz
P4, 1gig ram, on an Abit AS* Mobo, probably 3Com 10/100nic,
On Tue, 21 Sep 2004 12:05:29 -0500 (CDT), Nate Carlson
<[EMAIL PROTECTED]> wrote:
> Hey all,
>
> Someone's posted one of my 800#'s on a poster in California for free
> concert tickets, so I'm getting calls from California AC's at all times of
> the day asking for tickets. I'm just using the 800# f
Nate Carlson wrote:
But if I try:
exten => 8005551212/408XXX,1,Congestion
exten => 8005551212/408XXX,2,Hangup()
It doesn't catch it. Is there any way to do something similar and allow
wildcards? Thanks!
See:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20config%20extensions.conf
Craig:
Thanks very much for the pointer. I suppose a guy could use dual
monitors on a Reception PC running Flash Operators Panel. This would
work well for my application.
Thanks Very much for the info.
(SCCP sucks!)
Jesse Tyler
On 21-Sep-04, at 11:00 AM, Craig Guy wrote:
___
Hey all,
Someone's posted one of my 800#'s on a poster in California for free
concert tickets, so I'm getting calls from California AC's at all times of
the day asking for tickets. I'm just using the 800# for friends and
family, and don't know anyone in these area codes, so I'd like to just
giv
Michael Bielicki wrote:
Stable seized to exist quite some time ago.
To expand on Michael's answer, stable wasn't being kept up to date like
it should have been, so the statement "get the latest stable version"
became "get the latest cvs version" as the standard answer for resolving
people's iss
Hi Jesse,
I would strongly recommend changing over to the SIP image and uisng
something like the Flash Operators Panel (www.asternic.org) instead of the
7914's. I experimented with chan_sccp2 a few weeks ago and decided that it
wasn't for me right now due to both the very limited support for the
Thanks Matt:
(damn cisco) :) == > is right!!
I have already compiled the chan_sccp module. It is working just fine.
My main issue is actually configuring/loading the software the 7914 and
then using it like a main switchboard.
Thanks Again,
Jesse Tyler
On 21-Sep-04, at 9:53 AM, Matthew Boehm wro
OK, that's it. I wont use the RDSI/ISDN connection and will get the
ANALOG :) (sorry about my english) lines.
Thanks a lot for your help.
RODOLFO
Michael Loftis wrote:
I'm not sure if he means RDSI/ISDN and *ANALOG* (POTS)if
'analogic' means POTS then yes, he needs that ... TDM400P is an
P
I was thinking that there could be a way to do this through IAX, without
re-encoding of course. You could for example specify a special extension
on the remote server which would then pickup the stream like a regular VM.
Dominique
Kris Boutilier wrote:
Agreed, however these rely on foreknowledge
Hallo Daniel Eboa
On Tue, 21 Sep 2004 16:16:44 +0100 you wrote:
> Hello to all,
>
>
>
> I'm new user of Asterisk. I'm running Asterisk on a RedHat 9 platform.
> Everything seems to be ok but I got lot of error messages and I don't
> know their meaning. Can somebody help me ??
>
>
>
> These
hmmm I have no problems with 7960's and lates CVS since weeks
On Tue, 21 Sep 2004 10:41:54 -0400, Brian Cuthie <[EMAIL PROTECTED]> wrote:
>
> After downloading the latest CVS head and testing it with the Cisco 7960
> (SIP v7.2), it appears that it's still necessary to patch rtp.c to avoid
> audio
Hi everybody.
I have a Cisco 7905 IP Phone and as I see, the device isn't send the
registration message to the server, so to receive calls need to configure
static ip address.
Is there some way to make the Cisco send any sip registration? or Is there
some way to make the Cisco phone receive calls w
Stable seized to exist quite some time ago.
On Tue, 14 Sep 2004 16:35:28 +0500, Atif Rasheed
<[EMAIL PROTECTED]> wrote:
> on the asterisk site, it was stated while ago, how to download stable
> version. like
> cvs checkout -r v1-0_stable asterisk-addons zaptel libpri
>
> but now it's not their.
Hallo Martin Mielke
On Tue, 21 Sep 2004 17:03:54 +0200 you wrote:
> Thomas Niesel wrote:
>
> [ snip ]
>
> >
> >Does the phone had the same MSN?
> >
> >
>
> I think so. It could dial outside without a problem...
>
> >Is there maybe a PBX needs a leading "Digit" to get outside line?
> >
> >
Agreed, however these rely on foreknowledge of the remote end configuration
and are non-transactional. I was thinking more along the lines of VPIM
(http://www.google.ca/search?q=%22Voice+Profile+for+Internet+Mail).
Consider a large-scale private networking scenario - it would be very nice
to have
This certainly works, if you want to have a remote VM - but still does
not forward a received VM to another server.
Dominique
Matthew Boehm wrote:
Just have the two * servers login to eachother via IAX, then in your
extensions plan where you normally have:
exten => 8899,1,Dial(SIP/8899,15,tr)
e
If you are going to use the 7914 (which yes, unfortunatly isn't supported on
SIP, dammit Cisco) you might want to check out
http://chan-sccp.sourceforge.net
an alternative sccp module for *. Before we switched all our 7960's to SIP
we used this and it seemed alot better than the built in one.
Ma
Hello,
I am receiving an error in my error logs any time I receive a call on
the third line in our hunt group.
Sep 20 13:15:03 WARNING[1116939584]: Ring/Off-hook in strange state 6 on
channel 3
The weird part is that the calls seem to work fine, just this error
message is logged. Currently, I h
I'm in the process of setting up a queue system where the position
message and thankyou message are required to play every 90 seconds.
However, if a caller comes in to a queue with active agents logged in,
and no one else is in the queue, the messages play immediately, and
then the agents are poll
Just have the two * servers login to eachother via IAX, then in your
extensions plan where you normally have:
exten => 8899,1,Dial(SIP/8899,15,tr)
exten => 8899,2,Voicemail([EMAIL PROTECTED])
change it to
exten => 8899,1,Dial(SIP/8899,15,tr)
exten => 8899,2,Dial(IAX2//)
We have two * server
I'm not sure if he means RDSI/ISDN and *ANALOG* (POTS)if 'analogic'
means POTS then yes, he needs that ... TDM400P is an POTS/Analog NOT ISDN
device
--On Tuesday, September 21, 2004 11:28 -0300 Marconi Rivello
<[EMAIL PROTECTED]> wrote:
On Tue, 21 Sep 2004 15:52:57 +0200, Rodolfo Grave <[
Hi all,
I have a Wildcard that is flip floping between internally clocked and
the PRI. It is showing Red Alarm/Recovering. After a long run around
with the telco, they said I have lost the D channel on my side. I am
seeing this message:
== Restart on requested on entire span 1
Sep 21 08:29:48
Why not just use rsync or netcat? There are about a dozen different
ways to do this.
John
Kris Boutilier wrote:
I was having this thought also and I couldn't find any implementations.
Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery receipt
Hello,
We've released another update to our Asterisk GUI Client suite: 1.0.4
http://astguiclient.sf.net/
Screen shots: http://astguiclient.sourceforge.net/screenshots.html
The client suite runs on both Windows and UNIX, includes the VICIDIAL
auto-dialer and is free as in GPL.
(the suite is not
I was having this thought also and I couldn't find any implementations.
Likely it could be done using the sendmail 'pipe to shell' facility,
combined with some kind of delivery receipt system and a few minor hacks on
app_voicemail.c
> -Original Message-
> From: Dominique Kull [mailto:[EMA
"Matthew Boehm" <[EMAIL PROTECTED]> writes:
> Do you see how you had to put 2 SetCIDNum entries for 2 seperate
> dial-out numbers? Why can I not make 1 SetCIDNum entry for all
> outgoing numbers below it like I tried to do with the 's' extension?
You can, you just did it the wrong way. ;-)
> Is
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