Hello,
I've been using Asterisk for a while now. For a large project I think about
using SER, too.
But although I have studied the SER tutorial, I'm not quite sure, how Asterisk
and SER work
together, how Asterisk know about clients that are registered at the SER and so
on.
Can anyone of you
I forwarded your note below to [EMAIL PROTECTED] I found some docas
on the FCT-11M at their site, but it was in Chinese, so I sent them your
problem.
Hope they will respond to this list and maybe to you directly.
Murrah Boswel
- Original Message -
From: "Bill Michaelson" <[EMAIL PROTEC
That's not quite what I meant.
There is no such word as Telestrate. It has started to appear on weather
bulletins lately.
However, as you're having a bad day I'll drop it.
Mark
David Tillman wrote:
On 10/24/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
Syntacticly? Would that be the same a
On Mon, 2005-10-24 at 23:55 -0500, Erick Perez wrote:
> Hi, my employer will send me to Belgium, Bruselles from Nov 11 to Nov
> 26. Will an asterisk meeting take place anytime between those dates?
> Since in my country we have *no* meetings whatsoever I was wondering
> if there was going to be any.
Hi All
Can anyone please let me know if they have got UK CallerID working using a
X101P?
If so please can you let me know which version of asterisk, did you apply
the UK CID patch, what are your settings in zapata.conf, zaptel.conf and
extensions.conf to get it working?
There is a lot of confusi
Hi, my employer will send me to Belgium, Bruselles from Nov 11 to Nov
26. Will an asterisk meeting take place anytime between those dates?
Since in my country we have *no* meetings whatsoever I was wondering if
there was going to be any.
Or a tech fair showing asterisk. Or maybe a local asterisk "s
On 10/24/05, David Tillman <[EMAIL PROTECTED]> wrote:
> In my (inherited) extensions.conf I have some lines of the format:
>
> exten => o,2,GotoIf($["foo${FROM_DID}" =
> "foo"]?from-pstn,s,1:from-pstn,${FROM_DID},1)
>
> and some lines like:
>
> exten => s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo
I recently obtained a FCT-11M GSM-analog converter box. It arrived with
no documentation. So I popped in a SIM chip, and connected the the RJ11
port to an FXO port on my Asterisk box. It worked smoothly right away
for inbound and outbound calls in all respects. For about an hour.
Then eith
Can't help you much here but I know for a fact that a SPA 3000 works
perfectly fine with Asterisk (I got one). It's also correct that
Asterisk will not show anything if an incoming INVITE does not match
anything. Try SIP DEBUG and see what exactly is happening, where
asterisk is looking for the ext
I have not read through the rest of your posts, but try some of the other
variations of switchtype:
; Switchtype: Only used for PRI.
;
; national: National ISDN 2 (default)
; dms100: Nortel DMS100
; 4ess: AT&T 4ESS
; 5ess: Lucent 5ESS
; euroisdn: EuroISDN
;
Anyone have the? The patch is not avaialble in their site.
Would appreciate if you can send the patch
Thanks,
Ehsan
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On 10/24/05, Mark Phillips <[EMAIL PROTECTED]> wrote:
> Syntacticly? Would that be the same as Telestrate?
Look, I've been in the office for 14 hours today.
I just got through arguing with SBC about our T1 (Them: "our tests
show that it is
currently up". Me: "No kidding, we are currently talking
Syntacticly? Would that be the same as Telestrate?
David Tillman wrote:
In my (inherited) extensions.conf I have some lines of the format:
exten => o,2,GotoIf($["foo${FROM_DID}" =
"foo"]?from-pstn,s,1:from-pstn,${FROM_DID},1)
and some lines like:
exten => s,1,GotoIf($[foo${ECID${CALLERI
It's also catering for the fact that ${FROM_DID} might be a string with
embedded spaces, and it's assuming, probably not unreasonably, that
${CALLERIDNUM} doesn't have embedded spaces.
David Tillman wrote:
In my (inherited) extensions.conf I have some lines of the format:
exten => o,2,GotoI
There are none that I’m aware of at
the moment – the different echo cancellers just drop in as code
replacements at compile time, so they’re completely insulated from the rest
of the system, and vice versa.
Sounds like a good idea for a trivial
patch… anyone care to contribute such for
In my (inherited) extensions.conf I have some lines of the format:
exten => o,2,GotoIf($["foo${FROM_DID}" =
"foo"]?from-pstn,s,1:from-pstn,${FROM_DID},1)
and some lines like:
exten => s,1,GotoIf($[foo${ECID${CALLERIDNUM}} = foo]?4)
Note the quotes around the "foo${FROM_DID}" and "foo" in t
[EMAIL PROTECTED] wrote on 10/24/2005
08:09:27 PM:
> Also, if you're not already, try using the kb1 echo canceller from
> CVS-HEAD without aggressive cancellation before taking time to do
any of
> the above. It can be dropped into stable if needed by just copying
it
> (and the contents of the hea
On 10/24/05, Gary Reuter <[EMAIL PROTECTED]> wrote:
> Why not attach /etc/zaptel.conf and /etc/asterisk/zapata.conf?
Tacked on below.
> First thing I'd actually check is the wiring: if you jiggle the cable and
> the led changes, you've got a serious problem, but since you've only
> been the
One thing to consider is if there were alarms on the T1 to SBC, they may
have something in place to take the circuit down. Even if you get your
configs right, the T1 just might not come up clean. MCI does this to us
sometimes.
Please post your /etc/zaptel.conf and your /etc/asterisk/zapata.conf
I curently have a user who did not have any problems
on the asterisk. But in the last week or so, maybe
three times a day, when someone calls there extension,
the call goes straight to voicemail without even
ringing. The phone says registered but still does not
ring. Could any one out there have ex
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:asterisk-users-
> [EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
> Sent: Monday, October 24, 2005 12:07 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [Asterisk-Users] Terrible echo with Te110P and Adi
Why not attach /etc/zaptel.conf and /etc/asterisk/zapata.conf?
On my TE405 (PRI, not plain T1), a red alarm detected by software (kernel, zttool, or asterisk), coincides with the red led.
First thing I'd actually check is the wiring: if you jiggle the cable and the led changes, you've got a serio
> how many B channels and what range
> are they on and what channel is the D channel on (usally 24)?
23 B channels 1-23 and the Delta on 24.
I'm going to beat SBC around the head, I'm becoming convinced they are
to blame. It has taken me a bit to troubleshoot it because I am used to
old fashioned
Hi Alvaro,
We're running the same setup - check an earlier (like, within the
last week or so) thread on this list for our .conf details and email
me if you have further questions.
Regards,
--
Anthony Rodgers
Business Systems Analyst
District of North Vancouver
Web: http://www.dnv.org
RSS Fe
What kind of phone do you have on the asterisk side?
Have you tried just having the call from the Meridian go directly to
voicemail? If you hear the announcement and can leave a message,
the problem isn't with your ISDN connect, but on the other side.
If it is, turn on isdn debug (pri debug span..
its PRI line with what type of signaling, how many B channels and what
range are they on and what channel is the D channel on (usally 24)?On 10/24/05, David Tillman <
[EMAIL PROTECTED]> wrote:> well post your setup from the provider and maybe someone can help.
I'm not sure I follow you. More detail
> well post your setup from the provider and maybe someone can help.
I'm not sure I follow you. More detail: We only do VOIP in-house. The asterisk
box connects to a T1 provided by SBC by way of a Digium Wildcard TE110P.
Zaptel.conf indicates that the last admin believed the T1 to be configured fo
Or consider booting the kernel with "no_timer_check". It solved my issue
with a fast timer on a x86_64 AMD laptop with FC4.
Regards,
Patrick
On Mon, 2005-10-24 at 13:44 -0700, Hector Villalobos wrote:
> Uninstall FC4 and install FC2.
>
> -Original Message-
> From: [EMAIL PROTECTED]
> [m
well post your setup from the provider and maybe someone can help.On 10/24/05, David Tillman <[EMAIL PROTECTED]
> wrote:I am still getting up to speed on the Asterisk system in place at my
new employer.Today we are getting a lot of this:Oct 24 17:21:33 WARNING[2828]: Detected alarm on channel 1: Re
Hi list, i have the next situation
I've a asterisk connect with a Nortel Meridian Op 11, via a PRI CARD with 5ess switch
[Asterisk] -- PRI - NET --- PRI- CPE --[Nortell]
I can call from Asterisk to Nortell with no problem, but when Nortell
place a call to me, i have the channel bridge b
On Thursday 20 October 2005 10:45, Giordano Grandis wrote:
> Hi all,
>
> i'm looking for an utility that let me trace an ISDN trunk (or all ISDN
> traffic) on HFC PCI card.
I see no one is replying, so here is a little spam :) :
You may want to look at the tracing capabilities in vISDN:
http://ww
I am still getting up to speed on the Asterisk system in place at my
new employer.
Today we are getting a lot of this:
Oct 24 17:21:33 WARNING[2828]: Detected alarm on channel 1: Red Alarm
Oct 24 17:21:33 WARNING[2828]: Unable to disable echo cancellation on channel 1
[snip]
Oct 24 17:21:33 WARNI
We don't have a complete package quite yet. I think we have most of
what you will need but we do not have support at present yet to accept
customers payments. We can do that easily via 3rd party sofware but we
can't do it ourselves yet. Anyway, www.aleph-com.net/astpp is the link.
Darren Wi
Hi all.
Any body knows somethings about this issue ?
Some calls fails due to this cause , i have runing UnicallPre5 and
spandsp2.pre20
this is my unicall.conf
loglevel=255
protocolclass=mfcr2
protocolvariant=mx,10,4
protocolend=cpe
group = 1
context=incoming
channel => 1-10
;channel => 17-31
th
I have FC4 and 1.2.0beta. mpg123 0.59r worked fine in
my system.
Min
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Hector Villalobos
> Sent: Monday, October 24, 2005 4:45 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subj
Uninstall FC4 and install FC2.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dave
Sent: Monday, October 24, 2005 10:31 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Urgent - Need Help - Audio Issues
Hello, I am running asterisk v 1.2
On Monday 24 October 2005 16:34, Jorge Cisneros wrote:
> My wildcards TDM 400P's with 8 FXO port(connexts to pstn ) has ticking
> noise.
No interrupt sharing, good... you're using IOAPIC which should work just
fine... there's nothing obvious at this point that I've seen.
Try booting without th
Yes, for sure, it works. With TE110P from Digium using E1/ISDN/Pri
signalling.
By heart, I remember the following:
1. Configure Siemens E1 port as "station" and Asterisk as "Pri_Net" (or
Central Office).
2. At Siemens, set the E1 port as "S2 Point-to-Point net line without CRC4"
or somethin
Hi
My wildcards TDM 400P's with 8 FXO port(connexts to pstn ) has tickingnoise.I have followed http://www.voip-info.org/wiki-Asterisk+Hardware .
and make sure wctdm is not shareing interrupt with any other devices.The sever hard disk is a scsi, so i can't run " /sbin/hdparm -u1 /dev/hda1" to
We have our Asterisk boxes setup to connect our 3 offices in the US and
Canada over IP.
We found that this has saved us a lot of cost (we do least cost routing
to Canada through our Toronto PBX). One of the other locations we call
a lot is the Netherlands due to a concentrated customer base ther
Hi all,
I experienced more and more of the messages showed below.
It looks some kind of stuff accumulated in my system. It
don't seem to be cleared even after I reload the system.
What can cause this? My system is FC4 + 1.2.0 beta.
Thanks a lot,
Min
...
Destroying call '[EMAIL PROTECTED]'
Des
Greetings,
I work for a municipal government in BC, Canada, which is in the
throes of implementing Asterisk as a legacy PBX replacement
throughout our enterprise of around 400 users. If there is sufficient
interest from other government or enterprise users of Asterisk, I
would be interest
For the archives
Appears the issue was 1.1 code. Upgraded to 1.2 and all are
registering fine now.
On Oct 23, 2005, at 1:39 PM, Michael Welter wrote:
I just installed several 3104s in S. Calif. Didn't have any
problems--I was able to call from one line to another on the same
unit a
On 10/22/05, Crystal Stream, Incorporated <[EMAIL PROTECTED]> wrote:
You could have your customers call in and enter all ofthat -- then give them a confirmation number and theycould fill out the rest online.
Couple of notes on this topic.
First off, trixter's experience with the name being require
I want to connect an Ericsson MD110 with asterisk using a TE205P. Could someone
tell me if i need some especial media converter or any adapter to connect the
E1 port of the MD110 to E1 port of the digium card
Best Regards
Gabriel Astudillo
___
--Band
anybody can connect a Siemens HI-PATH to ASterisk via e1 ?
i need your help please.
___
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Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asteris
Carlos Alperin wrote:
Download from where? There is not such files on the
http://www.softswitch.org place.
Carlos Alperin
http://www.soft-switch.org/downloads/spandsp/
Doug
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-
Well I'm not sure I 100% understand your question, however Authorize.net
provides a payment gateway and merchant services if you don't currently
offer a merchant account, you can handle customers online or over the
phone.
I do tons of ecommerce design and offer merchant accounts to customers
Download from where? There is not such files on the
http://www.softswitch.org place.
Carlos Alperin
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Craig Guy
Sent: Tuesday, October 18, 2005 1:52 AM
To: Asterisk Users Mailing List - Non-Commercial Discuss
You could have your customers call in and enter all of
that -- then give them a confirmation number and they
could fill out the rest online.
--- trixter aka Bret McDanel <[EMAIL PROTECTED]>
wrote:
> I am interested in hearing some user experiences of
> anyone using a
> merchant account. The cons
On 10/24/05, Gavin Spurgeon <[EMAIL PROTECTED]> wrote:
> Hi List..
>
> How do I offer my help (and bandwidth) to become a mirror for this Book ?
I've gotten a couple of emails regarding hosting the book. If you'd be
interested in being a world mirror (outside the USA), then please
email me off lis
Hi,
I'm trying to configure a few service providers in asterisk. I could
configure correctly one only, messagenet.it.
I'm trying to register to sipgate and sipphone, with no success
(cannot register).
I'm inside a (departmental) firewall, not sure of the ports that are
closed, but I can configure
On 10/24/05, Gavin Spurgeon <[EMAIL PROTECTED]> wrote:
> Hi List..
>
> How do I offer my help (and bandwidth) to become a mirror for this Book ?
We pretty much have enough mirrors for the USA, but if you have a
server in the UK, that might be a good place to have another mirror.
I'm going to say
On Mon, 2005-10-24 at 10:01 -0700, Luki wrote:
> > [InfoWorld: Top News] Aruba unveils portable access point for VoIP
>
> Funny you mention this... I'm currently testing a similar setup,
> Asterisk on OpenWRT:
>
> WAN -> WRT54G -> [SSL] -> UDPTunnel -> [IAX] -> Asterisk -> [SIP] -> WiFi
>
There
This is probably the best option for you. Upgrade to 1.5.2 if that
works for you and set this option as Jerry mentions. It's accessible on
the phone menu.
Jerry Jones wrote:
set calls per button to one - in 1.5 and later code
On Oct 24, 2005, at 11:21 AM, Chris HARIGA wrote:
Bartosz Jozw
Hi,
I've a strange problem here. I can dial out via an AVM B1 card.
I have a sip client running. I can hear my conversational partner
but he can't here me. I'm using * 1.0.
Has anyone got this behavior?
Sascha
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Hello, I am running asterisk v 1.2.0Beta on a HP AMD
64 box. Redhat FC4 is installed on the box. I was not
able to compile mpg123 and am therefore using the
moh_native for moh. The snd_atiixp driver is loaded
for the sound card and I can play the demo sound using
desktop utility just fine.
Asteris
Adam Rybak wrote:
s,1,DaeadAGI,test.php,parameter1
How get value of parameter1 in php script?
This is actually a PHP question. You can find it in the PHP manual
online at http://www.php.net
$_SERVER['argv'][1]
Kevin
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After thinking about it for a few days, I realized that one way to prevent
non-admin users from entering the conference room is to use an AGI script
that actually performs the authentication. But, I would rather have the
functionality built into the Meetme application. Are there is any plans in
t
Hi List..
How do I offer my help (and bandwidth) to become a mirror for this Book ?
Best Regards
Gavin Spurgeon
Assistant Systems Administrator
[EMAIL PROTECTED]
http://www.leighctc.kent.sch.uk
Tel: 01322 620501
Fax: 01322 620599
IS HelpDesk : Ext 541
--
This message has been scanned for vi
I am trying to use a SIP provider for outgoing and incoming calls under
Asterisk. I am running a recent CVS-head 1.09 build and the SIP
provider is using a SPA-3000. I can register with the SIP provider's
server and outgoing calls seem to work just fine.
But I cannot get incoming calls to work a
On 10/24/05, Kevin Walsh <[EMAIL PROTECTED]> wrote:
> Steve Davies [EMAIL PROTECTED] wrote:
> > I hope this is not a FAQ - I have not been able to find it if it is
> > covered already...
> >
> > I have a dial-plan on my asterisk system that is becoming potentially
> > quite large and complex - Of t
Bartosz Jozwiak wrote:
Hi Guys,
I have a small problem.
I would like to disable call waiting function in Polycom phones while all
calls are handled by queues.
So far nothing, could not find an option in Polycom config to disable it.
Any help would be appreciated.
Bartosz
Hi,
U can doit
set calls per button to one - in 1.5 and later code
On Oct 24, 2005, at 11:21 AM, Chris HARIGA wrote:
Bartosz Jozwiak wrote:
Hi Guys,
I have a small problem.
I would like to disable call waiting function in Polycom phones
while all
calls are handled by queues.
So far nothing, could not f
> [InfoWorld: Top News] Aruba unveils portable access point for VoIP
Funny you mention this... I'm currently testing a similar setup,
Asterisk on OpenWRT:
WAN -> WRT54G -> [SSL] -> UDPTunnel -> [IAX] -> Asterisk -> [SIP] -> WiFi
And SIP clients via WiFi, or via wired LAN. At this point this is N
On 10/24/05, Chrispen Chisvo <[EMAIL PROTECTED]> wrote:
> Where is the book?
>
> Link please?
>From my post 11 hours ago in this thread...
Available here: http://www.asteriskdocs.org/modules/tinycontent/index.php?id=11
--
Leif Madsen - http://www.leifmadsen.com
http://www.oreilly.com/catalog/ast
Where is the book?
Link please?
On Monday 24 October 2005 10:30, Dave Grey wrote:
> On Oct 23, 2005, at 11:57 PM, Leif Madsen wrote:
> > On 10/20/05, Darrick Hartman <[EMAIL PROTECTED]> wrote:
> >> Leif Madsen wrote:
> >>> For those of you who are able to obtain the
> >>> full copy, please consi
On Oct 23, 2005, at 11:57 PM, Leif Madsen wrote:
On 10/20/05, Darrick Hartman <[EMAIL PROTECTED]> wrote:
Leif Madsen wrote:
For those of you who are able to obtain the
full copy, please consider helping us out by creating mirrors and
torrents and posting them to the list by replying to thi
We use use RS232 to Ethernet converters to solve this kind of
applications, for instance Moxa.
Jorge Mendoza
Jean-Michel Hiver wrote:
Hi,
I have a potential client who has legacy alarm systems which use
modems to transmit encoded data to a remote location through the PSTN.
They wish to repl
Bartosz Jozwiak wrote:
Hi Guys,
I have a small problem.
I would like to disable call waiting function in Polycom phones while all
calls are handled by queues.
So far nothing, could not find an option in Polycom config to disable it.
Any help would be appreciated.
Bartosz
_
Steve Davies [EMAIL PROTECTED] wrote:
> I hope this is not a FAQ - I have not been able to find it if it is
> covered already...
>
> I have a dial-plan on my asterisk system that is becoming potentially
> quite large and complex - Of the order of 12 lines of dialplan per
> extension number. Most
In CVS-HEAD and 1.2Beta the "new" KB1 echocan is enabled by default and
has solved most of our echo issues.
stoffell wrote:
On 10/24/05, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
Yes I did notice it immediately. I intend to tweak more, but for the
moment it seems like echo is minimized to
Hi,
I hope this is not a FAQ - I have not been able to find it if it is
covered already...
I have a dial-plan on my asterisk system that is becoming potentially
quite large and complex - Of the order of 12 lines of dialplan per
extension number. Most of this is in order to record suitable CDR
dat
hi
Anyone to help me add extensions to my Asterisk PBX. Step by step please, help me.
I have been doing the following: you can correct me if I am missing anything:
on the xlite client running Windows XP, on ip address 192.168.1.35:
-Enable: Yes
- Display name: anytext
- Username:135
- Authorisa
On 10/24/05, Olle E. Johansson <[EMAIL PROTECTED]> wrote:
> Steve Davies wrote:
> > On 10/21/05, Steve Davies <[EMAIL PROTECTED]> wrote:
> >
> >>I have noticed that when a SIP redirect is sent back to Asterisk by a
> >>SIP peer, that Asterisk will (quite appropriately) do a
> >>Dial(LOCAL/redirect-
Hi Guys,
I have a small problem.
I would like to disable call waiting function in Polycom phones while all
calls are handled by queues.
So far nothing, could not find an option in Polycom config to disable it.
Any help would be appreciated.
Bartosz
__
Giovanni Miano wrote:
I've 2 hfc billion and one TDM400P 1fxs/1fxo with bristuff 0.2.0-RC8o
and * 1.0.9
I dont recive callerid from TDM400P fxo port but isdn hasnt problems
If i try to use only TDM400P 1fxs/1fxo without bristuff.. all work ok
is it bug of bristuff ?
Maybe, why not try brist
Dear Colleagues,
I Have my * with a X100P clon card. When a call in from the PSTN and
nobody answer the call go to the voicemail, then the caller my hangup
or press #. If the caller hangup the ZAP channel never hangup, but if
the caller press # the ZAP channel hangup. Even every time the outside
p
We have Snom 190s in an office of about 30. Trying to use the 5 lit
buttons on the right to be used for parked calls/calls on hold. In other
words, want to be able to transfer someone to either an extension that
maps to the buttons or anyone on hold gets put into that "queue" of lit
buttons so a
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] nombre de Robert Webb
Enviado el: Domingo, 23 de Octubre de 2005 06:24 p.m.
Para: asterisk-users@lists.digium.com
Asunto: [Asterisk-Users] Hardware setup question
I have just a quick setup question about how some of you
Steve Davies wrote:
> On 10/21/05, Steve Davies <[EMAIL PROTECTED]> wrote:
>
>>I have noticed that when a SIP redirect is sent back to Asterisk by a
>>SIP peer, that Asterisk will (quite appropriately) do a
>>Dial(LOCAL/redirect-number) in the context of the original callee.
>>
>>It also forks the
On 10/21/05, Steve Davies <[EMAIL PROTECTED]> wrote:
>
> I have noticed that when a SIP redirect is sent back to Asterisk by a
> SIP peer, that Asterisk will (quite appropriately) do a
> Dial(LOCAL/redirect-number) in the context of the original callee.
>
> It also forks the CDR, which is excellent
Dear colleagues,
I read on the web that implementations of thousand SIP extension on
asterisk became worse and people suggest SER + asterisk.
Anybody checks this? what is the problem?
Any comment
Kind regards,
Juanjo
___
--Bandwidth and Colocation spon
Carlos Alperin wrote:
Doug,
Thank you very much, that is exactly the operation mode that I'm looking
for. Are you using a TDM405P card for the PRI?
Regards,
TE110P
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary Safety,
deserve
Thanks I'll try it.
Regards,
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Juan Jose
Comellas
Sent: Monday, October 24, 2005 12:30 AM
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Trying to clarify ideas about spands,libtiff
Doug,
Thank you very much, that is exactly the operation mode that I'm looking
for. Are you using a TDM405P card for the PRI?
Regards,
Carlos
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Doug Lytle
Sent: Monday, October 24, 2005 5:41 AM
To: Asterisk
Hi,
I'm trying to configure a few service providers in asterisk. I could
configure correctly one only, messagenet.it.
I'm trying to register to sipgate and sipphone, with no success
(cannot register). I could configure in iax.conf voipbuster and
sipdiscount, it registers but it does not allow me t
Sergio Chersovani schrieb:
Tomasz Chmielewski ha scritto:
I searched the whole "Cisco IP Phone 7905 Series Administration
Guide", but besides the copyright notes, logo is not mentioned.
lol
http://www.google.com/search?q=site%3Acisco.com+7905+bmp2logo
I see, it's called bmp2logo.exe and it
Olle E. Johansson wrote:
Cirelle Enterprises wrote:
is the cvs head version considered 1.0 or 1.1 with
regard to spandsp
CVS head would be considered 1.1 at this time.
/O
___
Thanks
Tom Rymes wrote:
> On Oct 24, 2005, at 6:25 AM, Olle E. Johansson wrote:
>
>> Guido Hecken wrote:
>>
> I was looking for the text in the /etc/asterisk directory, but it must
> be somewhere else. Can anybody tell me where? And can it include
> Chinese
> as well?
Check
Tomasz Chmielewski ha scritto:
I searched the whole "Cisco IP Phone 7905 Series Administration
Guide", but besides the copyright notes, logo is not mentioned.
lol
http://www.google.com/search?q=site%3Acisco.com+7905+bmp2logo
___
--Bandwidth and Colo
Cirelle Enterprises wrote:
> is the cvs head version considered 1.0 or 1.1 with
> regard to spandsp
>
CVS head would be considered 1.1 at this time.
/O
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Us
On Oct 24, 2005, at 6:25 AM, Olle E. Johansson wrote:Guido Hecken wrote:I was looking for the text in the /etc/asterisk directory, but it mustbe somewhere else. Can anybody tell me where? And can it include Chineseas well?Check voicemail.conf in /etc/asterisk or voicemail.conf.sample in the/configs
is the cvs head version considered 1.0 or 1.1 with
regard to spandsp
--
Best Regards
Greg Cirino
Spam and Virus Free Email
included with every email account
Cirelle Enterprises Inc.
25 Indian Rock Rd #421
Windham NH, 03087
603-425-2221
___
--Bandw
Erik schrieb:
Logo file for a 7905 isn't a BMP file, it has to be converted with a cisco util
aah, now I see.
and what tool is that and where can I get this?
in my firmware package I have only two tools:
- cfgfmt.linux (a tool for converting text configuration into cisco
format, which doesn't
[InfoWorld: Top News] Aruba unveils portable access point for VoIP
http://www.infoworld.com/cgi-bin/redirect?source=rss&url=http://www.infoworld.com/article/05/10/24/HNaruba_1.html
basically it creates a VPN connection to let remote users connect with
some level of security. It also has an access
Tomasz Chmielewski ha scritto:
any idea why a custom logo isn't displayed on a 7905G phone?
The logo image file need to be encoded. You will find the tools at the
cisco website
Sergio
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A
Logo file for a 7905 isn't a BMP file, it has to be converted with a cisco util
Tomasz Chmielewski wrote:
> Sergio Chersovani schrieb:
>
>> Tomasz Chmielewski ha scritto:
>>
>>> But I have (had?) a problem with 7905 phone (still minor problems
>>> with that, like a wrong timezone).
>>
>>
>>
>> Y
Sergio Chersovani schrieb:
Tomasz Chmielewski ha scritto:
But I have (had?) a problem with 7905 phone (still minor problems with
that, like a wrong timezone).
You can easy change it with the phone web page.
yup, I just figured that out :)
one more issue though.
any idea why a custom logo
I am trying to use a SIP provider for outgoing and incoming calls under
Asterisk. I am running a recent CVS-head 1.09 build and the SIP
provider is using a SPA-3000. I can register with the SIP provider's
server and outgoing calls seem to work just fine.
But I cannot get incoming calls to wo
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