Hi:
Does anyone know if it's possible to record channels in raw PCM instead
of GSM format?
Thanks,
-Stephen-
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Do you have a computer connected to the cisco phone?
Carlos Alperin
Senior System Engineer
Seneca Communications, LLC
[EMAIL PROTECTED]
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edwin Lam
Sent: Wednesday, September 28, 2005 8:12 PM
To: Asterisk
Steve,
I hope you didn't feel bad about my opinions. This is why I asked a couple
of weeks ago about to clarify regarding T.38 on the 1.0.9 CVS.
I read on the mailing list that someone sent an e-mail about the release of
the last CVS with the T.38 passthrough included. And then I started to
Thanks Steve about your great work.
I am very antious to test it!
Thanks again.
Rosario
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, September 28, 2005 8:52
Hi,
no, not in french, in italian but the matter is the sameI found the
only solution is to change ${VM_DATE} is to change the source code...
::))
Giorgio
Nathan Pralle wrote:
What exactly are you trying to do? Get it to say the date in French?
Nathan
gincantalupo wrote:
Hi,
Hi all,
Is realtime meetme conference supported by Asterisk?
Regards.
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To
On Wed, 28 Sep 2005 [EMAIL PROTECTED] wrote:
Hello,
I am using a system with an AVM ISDN PCI card (fcpci) and asterisk with
chan_capi-cm-0.6. The hardware is connected to a Siemens Hipath 3550 PBX. As a
BRI connection has 2 channels and allows 2 simultaneous calls, numbers/MSNs
6391 and
I want to see if any
of my fellow Asterisk-Users list members have dealt with these guys. I'm a admin
for a VOIP provider, and have encountered a few PBX customers that want
consulting/support for the IAreaNet provided Asterisk pbxs. These guys are
selling AAH servers to the public, and are
Hello all,
I have someone working for me who has a nice phone voice. I looked at
some available prompts for asterisk, and found both the free and
commercial ones to be pretty horrible. The asterisk ones are good, but
I wish I had more to choose from sometimes.
My question is, what do you think,
Hi,
I would like to know what type of configuration could get me closer to
100% hits in zttest, when testing a TDM400P with 4 FXO ports,
I am currently running kernel 2.4.31, on a IBM Xseries 306, with 3gh
CPU, HT is disabled, PCI latency was changed, i still cant get more then
99.975% in
Hi :)
Am Donnerstag, 29. September 2005 09:03 schrieb Voice over IP:
Hi all,
Is realtime meetme conference supported by Asterisk?
Yes and no.
I wrote a patch for an older CVS-Version and will port it to the latest CVS
version.
Will take 2 or 3 weeks ;)
So current versions do not support
Hi group,
anyone can explain me the exact difference between
pri value in zapata.conf ?
; PRI Dialplan: Only RARELY used for PRI.;; unknown: Unknown; private: Private ISDN; local: Local ISDN; national: National ISDN; international: International ISDN
If I use it, I also must use
Hey.
How would I set up my dialplan if a user wants to call its voicemail
from an external phone?
I'm thinking of getting the user to enter its mailbox number.
Something like this:
1. User calls the dedicated voicemail number.
2. Phone prompts for mailbox number.
3. Voicemail([EMAIL
On Thu, 2005-09-29 at 10:43 +0200, Arne Morten Johansen wrote:
Hey.
How would I set up my dialplan if a user wants to call its voicemail
from an external phone?
I'm thinking of getting the user to enter its mailbox number.
Something like this:
1. User calls the dedicated voicemail
hi,
are any one working on h324 codec with asterisk for 3g video communication ...does asterisk support this
regards
kiran
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We ended up doing it in the c code made it so user can hit * and it will
prompt them for a password. We figured that was the easiest way to go about
it.
Carlos Alcantar
Race Technologies, Inc.
101 Haskins Way
South San Francisco, CA 94080
P: 650.246.8900
F: 650.246.8901
E: [EMAIL PROTECTED]
Hey ho!
We have a functional t.38 implementation for asterisk, but its far from
complete. (meaning it doesnt work for all devices, and i only tested it
on 1 fax).
I hope to take our t.38 developper with me to Astricon and maybe even
demo it there. (Maybe oej could bring a fax or two ? :)
I
for the community, I think it is important to have at least t.38 passthrough
first then the other devolpments.
In this way t.38 can be easly spreaded and catch up more supporters.
Rosario
- Original Message -
From: Zoa [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
On Thu, Sep 29, 2005 at 05:12:21AM -0400, Rosario Pingaro wrote:
for the community, I think it is important to have at least t.38
passthrough first then the other devolpments.
In this way t.38 can be easly spreaded and catch up more supporters.
What do you mean more supporters. t.38 is only
So, after some research I can provide you with some more information.
According to our employees on every fourth call the dialtone is choppy.
That happens, not like I said first, when we dial trough phpagi AND when
we dial directly with x-pro (but both times through asterisk).
In X-Pro its a
I have set up extension.conf and sip.con with default
parameter of UNIVOICE server, but Asterisk show this
message when I call a number:
Sep 29 11:34:52 WARNING[4179]: chan_sip.c:1899
create_addr: No such host: univoice,Ttr
Sep 29 11:34:52 NOTICE[4179]: app_dial.c:1109
dial_exec_full: Unable to
Kresimir Petrovic schrieb:
...
What do you mean more supporters. t.38 is only *reliable* way for transporting
fax over ip. Fax over g711 is pure luck...
Hi,
it is rather a question of IP quality than good luck.
I think, 99.9% of all faxes are transported via G.711.
Is there any telecom
Hi!
I have a strange problem. In an AGI I tell Asterisk to playback a number, for
example 31. I then use the AGI SAY NUMBER command and I only hear thirty
and then get:
-- Playing 'digits/30' (language 'de')
Sep 29 11:47:40 WARNING[3401]: file.c:475 ast_openstream: File does not exist
in any
yes
On 28. sep. 2005, at 15.54, Tom Hayden wrote:
You're going to need to explain a little more. When you say central
are you talking about an SMSC?
--
Tom
On 9/28/05, Roy Sigurd Karlsbakk [EMAIL PROTECTED] wrote:
hi
is it possible to use asterisk as an sms central to send SMSes
directly
Try using filename:wav instead of
filename:WAV
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You want something like this:
exten=_+1NXXNXX,1,SIPDtmfMode(inband)
exten=_+1NXXNXX,2,Wait(4)
exten=_+1NXXNXX,3,Playback(please-enter-your)
exten=_+1NXXNXX,4,Background(ha/mailbox)
exten=_+1NXXNXX,5,DigitTimeout,5
exten=_+1NXXNXX,6,ResponseTimeout,10
for the community, I think it is important to have at least t.38
passthrough first then the other devolpments.
In this way t.38 can be easly spreaded and catch up more supporters.
What do you mean more supporters. t.38 is only *reliable* way for
transporting
fax over ip. Fax over g711 is
What do you mean more supporters. t.38 is only *reliable* way for
transporting
fax over ip. Fax over g711 is pure luck...
Hi,
it is rather a question of IP quality than good luck.
I think, 99.9% of all faxes are transported via G.711.
Is there any telecom network operator left using
Hi,
Can anyone tell me if Asterisk sets some variable when doing a call
parking (when someone presses an exten set in features.conf). In can't
find this information on a wiki.
Cheers ;)
Andrew
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It was indeed the problem with the language 'de' setting, setting the SIP
client to US gives me the numbers.
On Thursday 29 September 2005 12:00, Christoph Eicke wrote:
Hi!
I have a strange problem. In an AGI I tell Asterisk to playback a number,
for example 31. I then use the AGI SAY NUMBER
Hi,
I tried to use the version 0.6 of chan_capi-cm for outgoing calls it
works perfectly but for incoming calls it will not work:
--- snip ---
*CLI capi debug
CAPI Debugging Enabled
-- CONNECT_IND
(PLCI=0x101,DID=97,CID=0179903,CIP=0x1,CONTROLLER=0x1)
== reventix: Incoming call
Hello everyone. Ive seen postings for connecting
asterisk to vonage but Im still having trouble achieving that.
I have a vonage softphone and I'm trying to register to vonage using asterisk.
I have not had any luck. I am behind a firewall. I've successfully gotten xlite
to connect and work
Hello,
I have one simple question. Is it bug that for From: 1234 1234
sip:[EMAIL PROTECTED];user=phone; ${CALLERIDNUM} is 1234 instead of 5678 ?
Asterisk 1.0.9
--
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On Thu, 29 Sep 2005, Bastian Schern wrote:
Hi,
I tried to use the version 0.6 of chan_capi-cm for outgoing calls it works
perfectly but for incoming calls it will not work:
--- snip ---
*CLI capi debug
CAPI Debugging Enabled
-- CONNECT_IND
On Wed, September 28, 2005 5:41 pm, Matt said: I have heard this issue when on hold with Cisco and Vonage... Idon't think it's an asterisk problem I htink it's a G711 problem... orgsm
problem. Basically they are made for voice, and I think the music goes outside their encoding ranges... sound
Arne, been posted many times do a search on the voip-info site on Disa.
Does exactly what you are after.
Cheers,
Dean
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Arne Morten Johansen
Sent: Thursday, 29 September 2005 4:43 AM
I bought some USB soundcard/handsets from them with
no issues.
I did not deal with them on any PBX or config
issues though.
-- -- Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.--- -
--- - - - -
- - - -- - - - --- -
-- - - --- - -
I have been working on solving a major issue with
IPSwitchBoard. It was reported that IPS would use all available memory and get
the PC to grind to a halt.
I could not understand this as I had it running on
many different PCs in Denmark.
I now found the bug:
IPS would crash on
Can anyone recommend a soft phone for my Zaurus PDA that will play well
with Asterisk?
TIA
Jason
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Well,
I had an issue with them charging funds on PayPal
for stuff they never sent out, and they justsat on their hands for 3 months
till I contacted them to get a refund back (took me some time to check my
paypal), and then it took them 3 weeks to refund me.
Nir S
From: [EMAIL
I just copied the *98 extension to the extension of one of our DID numbers.
So now if I dial 5686 from inside or 1-xxx-xxx-5686 from outside, I get the
same prompts as dialing *98.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---
Well, it depends what country you're in and what kind of protocols you
are using. Here in the US, I prefer to *not* use asterisk and use the
perl module Net::SMPP to handle my SMS traffic between my
gateway/aggregator and the carriers SMSC. It's somewhat easier to
configure with special
I have asterisk 1.0.9 installed with spandsp 0.0.2pre20.
Asterisk crashes on boot while loading app_txfax.so
app_rxfax.so. If I move the files out of /usr/lib/asterisk/
modules asterisk boots fine.
Running on FC3, Linux asterisk.crocker.com 2.6.11-1.27_FC3smp #1
SMP Tue May
I have had problems between the sip/FXO lies and was able to kill the
echo by trying different combinations of the echocancel line to 64 (I
think it has settings in 32 bit increments)
Just kept trying different ones till it went away. Here is my config:
group=1
context=line1
signalling=fxs_ks
Edwin,
They are on the same VLan and on the same Subnet? If that's the case
check you switch log for details, if you havent changed anything on the *
Server. Looks like a serious package lost, even with a high segment this
shouldn't occur. At least for the info you send, these are the POF.
Yup that's what I was going to suggest you do.. we've been using that
and it works great.
On 9/29/05, Steven [EMAIL PROTECTED] wrote:
I just copied the *98 extension to the extension of one of our DID numbers.
So now if I dial 5686 from inside or 1-xxx-xxx-5686 from outside, I get the
same
I listened to all the demos you showed.
My ear discerns a little muffling and minor slushiness in the GSM files
you sent, along with a much more narrow bandwidth, mainly on the high end
side, and Allison either has a mild whistling s or slushy s sound in her
voice or the producer didn't properly
Hi all,
I hope someone can help, as I have an urgent problem.
I've got a production Asterisk server thats been deployed, but we are seeing
a strange voice echo problem. There is about a 250ms echo for the users in
the office, and they are hearing their own voice back at them.
I'm running
You could look up at '/tmp' if you are runing * in safe mode ...guess that
help you
G.
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What kind of POTS trunks/cards are you using?
--
Tom
On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote:
Hi all,
I hope someone can help, as I have an urgent problem.
I've got a production Asterisk server thats been deployed, but we are seeing
a strange voice echo problem. There is about a
Hi,
I've recently reinstalled a Diva in my asterisk server (alongside a
QuadBRI :-) to test the nice features Armin has been adding in
chan_capi.
The capi.conf format has changed, so my question is how do I define a
deflect= statement for different incoming MSN's?
I've tried to define a section
Hi,
I phone with phpagi and/or x-pro.
Sometimes I get this warning in the asterisk-console:
maximum retries exceeded on call.
I noticed when this message shows up, asterisk hangs up the call (even
when i'am in the middle of a call, according to our employess)
When they restart x-pro it seems
Hi there:
Is there any way to change the language in asterisk to Spanish...I
mean I want to change all the dialogs to Spanish in my * box can u help me
pls.
Hector
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We have contracted with an outside call center to provide sales for a
certain product. We want to be able to transfer people over to those
dedicated sales agents using an attended transfer (so we can prepare them
with as much information as we have), to a regular extension. So far, so
good. All
Zoa wrote:
Hey ho!
We have a functional t.38 implementation for asterisk, but its far from
complete. (meaning it doesnt work for all devices, and i only tested it
on 1 fax).
I hope to take our t.38 developper with me to Astricon and maybe even
demo it there. (Maybe oej could bring a fax or
Ok. I've been searching the wiki and google for a long time now. HOW do
I enable asterisk to send mail when users get new messeages in there
mailbox?
Do i need to change mailcmd in voicemail.conf?
Regards,
Arne morten
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Solo probando
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Arne Morten
Johansen
Enviado el: Jueves, 29 de Septiembre de 2005 07:50 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: [Asterisk-Users] Getting asterisk to send e-mail
I have a Polycom IP600 serving as a receptionist phone. We
developed a call manager via c/gtk that runs on a touchpad. It
allows them to transfer calls, transfer to voicemail, page, etc.
The problem is this: When paging another phone from the touchpad,
I have to open a channel to the receptionist
I'm using a Voicetronix OpenSwitch12 analouge card. It's on the latest
driver for Asterisk (2.4.9) and has echo cancellation turned on. This works
fairly well on on SIP-POTS calls after it trains up, but there is still a
small echo. The SIP-SIP calls are really echoy though.
Cheers,
Ian
On Thu, Sep 29, 2005 at 12:38:40PM +0200, Roy Sigurd Karlsbakk wrote:
for the community, I think it is important to have at least t.38
passthrough first then the other devolpments.
In this way t.38 can be easly spreaded and catch up more supporters.
What do you mean more supporters. t.38
I didnt implement anything myself and am not very familiar with t.38,
but i think its udptl, sip, iax2, and soon gateway too.
I will try to get a little more info from the developper when he gets
back.. Maybe its even based on your work, i should check.
Sorry for the incomplete reply, i just
Why is what he is doing different than having the fax machine on a
Sipura ATA?
Just because both those ports are on the pci card that doesn't make them
not Voice in betweenif I'm wrongeh...oh well
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Solo probando
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hi Hector. Just use the * command SetLanguage(), passing as argument es
http://www.voip-info.org/tiki-index.php?page=Asterisk+cmd+SetLanguage
for more info check the link, and other related links at the bottom of that page.
best regardsOn 9/29/05, Hector Elias Menjivar [EMAIL PROTECTED] wrote:
Hi Arne,
In /etc/asterisk/voicemail.conf, under the [default] section, you need to
declare the users like this :
box# = passnumber for box, Name of User,email address
e.g.
221 = 1234,Ian Bonham,[EMAIL PROTECTED]
Do that for each mailbox you require.
Then in the sources directory, under
Prueba
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Zoa
Enviado el: Jueves, 29 de Septiembre de 2005 07:57 a.m.
Para: Asterisk Users Mailing List - Non-Commercial Discussion
Asunto: Re: [Asterisk-Users] T.38 Faxing - at astricon ?
I didnt implement
Well, I think what he means is that it's not VoIP, because you are
using TDM on both ends. It looks like this:
fax machine - TDM - * - TDM - PSTN
If you had a SIP ATA attached to a fax machine, you would be using
VoIP. That would look like this:
fax machine - SIP/VoIP - * - TDM - PSTN
I have
Arne Morten Johansen wrote:
Ok. I've been searching the wiki and google for a long time now. HOW do
I enable asterisk to send mail when users get new messeages in there
mailbox?
Do i need to change mailcmd in voicemail.conf?
Make sure sendmail is installed.
Doug
PRI
dialplan, in a nutshell, sets dialled digits from your Asterisk box to a pattern
that your telco expects. For example, if your telco expects numbers in XXX-
format ALWAYS, then you would set it to Local so the MSD of whatever your user
dials is stripped off by Asterisk, leaving only
This looks like the info you want:
http://www.voip-info.org/tiki-index.php?page=Polycom+auto-answer+config
BTW, is your touchpad app publicly available?
On 9/29/05, Eric Lawman [EMAIL PROTECTED] wrote:
I have a Polycom IP600 serving as a receptionist phone. We
developed a call manager via c/gtk
Hi thanks.
Where can i find this variable
Hector
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Moises Silva
Enviado el: Jueves, 29 de
Septiembre de 2005 08:03 a.m.
Para: Asterisk Users Mailing List
- Non-Commercial Discussion
Asunto: Re:
hi
has any one used OOH323C i tried this it is installed but do not know how to
configure has any one used this, what is the best h323 addon to use with
asterisk
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Roy Sigurd Karlsbakk schrieb:
...
see http://soft-switch.org/foip.html for a brief explaination of why
this generally doesn't work...
Hi,
maybe one should update this link.
I think, you agree, that VoIP is somewhat similar to ISDN, as it
transports analog audio data in a digitally coded
Hello
When I dial out from my Asterisk (using Digium analog TDM04B card over pstn
line), calls appear to be from +34rest of number
I am in UK which is +44 so cannot work out why seeing +34.
In my zapata.conf I have:
loadzone = uk
defaultzone = uk
I can't find any country specific stuff in
Are you certain that the echo on sip--sip calls is not being caused
by either a spakerphone or extremely loud handset?
On 9/29/05, Ian Bonham [EMAIL PROTECTED] wrote:
I'm using a Voicetronix OpenSwitch12 analouge card. It's on the latest
driver for Asterisk (2.4.9) and has echo cancellation
Perfect, thanks very much
hth. I just set it to unknown, but it doesnt work.
Have I to use also prilocaldialplan
?
Thanks again
Giordano
Da:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Per conto di Colin Anderson
Inviato: giovedì 29 settembre 2005
16.22
A: 'Asterisk
hi guys
I was working on asterisk and h323 for the past 2 weeks
i have the following feedback please let me know if i am wrong
h323 implementation
I managed to install this it works, but the problem is it accecpts all calls
from all ips. there is no way i can let it accecpt calls only from the
Not sure about the Digium, but I can tell you +34 is Spain, if that helps
you track anything down? I assume you've tested the line with a normal phone
to make sure it's not a telco fault?
Ian
From: Angus Comber [EMAIL PROTECTED]
Reply-To: Asterisk Users Mailing List - Non-Commercial
Hi Matt,
I've tried using both speaker phone and handset and had the volume levels
really low and it still occurs. The transmit volumes on the Polycom IP600's
I have are a fixed transmit volume however, which is set to TIA/EIA-810-A
standard.
I have changed the gain settings in the VPB
Roger Schreiter wrote:
Roy Sigurd Karlsbakk schrieb:
...
see http://soft-switch.org/foip.html for a brief explaination of why
this generally doesn't work...
Hi,
maybe one should update this link.
Update it in what way?
I think, you agree, that VoIP is somewhat similar to ISDN, as
Sherwood McGowan wrote:
I listened to all the demos you showed.
My ear discerns a little muffling and minor slushiness in the GSM files
you sent, along with a much more narrow bandwidth, mainly on the high end
side, and Allison either has a mild whistling s or slushy s sound in her
voice
Ziaxphone might fit your needs. http://www.kauss.org/Stephan/ziaxphone/
Haven't used it recently since someone broke the screen on my Zaurus =(
-- William
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On Thu, Sep 29, 2005 at 08:33:03AM -0400, Tony Nichols wrote:
I have had problems between the sip/FXO lies and was able to kill the echo
by trying different combinations of the echocancel line to 64 (I think it
has settings in 32 bit increments)
Just kept trying different ones till it went
Am Donnerstag 29 September 2005 16:28, Kanishka Somaratne schrieb:
hi
has any one used OOH323C i tried this it is installed but do not know how
to configure has any one used this, what is the best h323 addon to use with
asterisk
Hello,
for me it seems that the OOH323 Development is not
Have I to use also prilocaldialplan ?
Can be left unknown.
Explains what you expect as the incoming number to look like
Thanks again
Giordano
; PRI Dialplan: Only RARELY used for PRI.
;
; unknown:Unknown
don't expect anything
; private:Private
Hi guys,
Need some advise.
Is there some kind of call center software which can interconnect with
asterisk?
So, for example, agents can see on their pc's all info about calling client
(based on clid)
before they pick up the phone.
And that outbound calls are also automated.
Commercial
The
values are mutually exclusive so you can only set it once. What you want to do
is from the Asterisk console type in PRI DEBUG SPAN 1 (if you only have 1 PRI)
and place a call. PRI DEBUG will throw up everything on the screen concerning
call setup and teardown at the PRI network layer. A
Roger Schreiter wrote:
I think, you agree, that VoIP is somewhat similar to ISDN, as it
transports analog audio data in a digitally coded way.
Noone doubts, that ISDN is suitable to transport analog fax.
Finally the PSTN is 99,9% digital (ISDN/SS7), even if some
subscriber lines are still
On Wed, 28 Sep 2005, Stephen Bosch wrote:
[EMAIL PROTECTED] wrote:
When I listen to the GSM compressed prompts, I can hear subtle noise
when the person is speaking -- this is irrespective of whether I listen
to the prompts through the TDM-400 on an analogue phone or whether I do
so
AstGUIClient and VICIDIAL seem to be a good tool for the task. I cannot
verify, as I have not used them before.
http://astguiclient.sourceforge.net/
Nathan
Bartosz Jozwiak wrote:
Hi guys,
Need some advise.
Is there some kind of call center software which can interconnect with
asterisk?
Hi Andrew,
Not sure if I understand your question, but this may help - * has the
following settings in features.conf that are related to parking:
parkext = ;the extension that users xfer calls to in order to
park them
parkpos = - ;the extension range that * will use to park
www.inconcertCC.com has a solution based on Asterisk.
regards,
srsergio
-Mensaje original-
De: Bartosz Jozwiak [mailto:[EMAIL PROTECTED]
Enviado el: jueves, 29 de septiembre de 2005 17:17
Para: Asterisk Users Mailing List - Non-Commercial Discussion
CC: Commercial and
The problem as I see it is that if people start expecting it to work then
rather than being pleasantly surprised when it does, they will be bitterly
disappointed when it doesn't. IMHO analog fax over IP is too flaky to
encourage the general public to utilise, and any suggestion to the contrary
Have you checked that the TDM400P isn't sharing an IRQ with anything
else? Don't trust /proc/interrupts - run lspci -v to confirm this.
We have * running on an x206 and found that the only way to stop the
TDP400P sharing an IRQ with other devices was to juggle cards between slots.
Hope this
Message: 7
Date: Thu, 29 Sep 2005 09:53:27 -0400
From: Eric Lawman [EMAIL PROTECTED]
Subject: [Asterisk-Users] Remotely dialing calls from a polycom phone
To: Asterisk-Users@lists.digium.com
Message-ID: [EMAIL PROTECTED]
Content-Type: text/plain; charset=iso-8859-1
I have a Polycom IP600 serving
I hate to bother the list with this potentially minor issue but
I just wonder if it's a symtom of some other problem.
Every time I make a call the BT-102, with the latest firmware, she just
keeps the LED display lit and the timer counting after hangup.
I check the CLI and the hangup is
On 9/29/05, William Suffill [EMAIL PROTECTED] wrote:
Ziaxphone might fit your needs.
http://www.kauss.org/Stephan/ziaxphone/ Haven't used it
recently since someone broke the screen on my Zaurus =(
I can vouch for the software. I haven't used it in some time, but it
DID work when I tried it on
Hi,
My TDM is on its own IRQ, and the x306 has only one full-size PCI slot..
so no playing with it,
what results do you get from zttest ? what IRQ is the card on ?
Marco.
Damian Funnell wrote:
Have you checked that the TDM400P isn't sharing an IRQ with anything
else? Don't trust
All the config match. Just to make sure, how did you make your loopback cable? Which pins are conected were?
Thanks in advance,
-f
From: "Steve Totaro" [EMAIL PROTECTED]Reply-To: Asterisk Users Mailing List - Non-Commercial Discussionasterisk-users@lists.digium.comTo: "Asterisk Users Mailing
This might seem a silly question but, what is the true meaning of the numbers zttest spits out?On 9/29/05, Marco Supino
[EMAIL PROTECTED] wrote:Hi,My TDM is on its own IRQ, and the x306 has only one full-size PCI slot..
so no playing with it,what results do you get from zttest ? what IRQ is the
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