Hi all,
i'm trying to integrate cepstral and asterisk, and i have a problem i'd
appreciate any help with (i know it's a bit tangential, but i figure this is
the place with the most knowledge of app_swift and asterisk).
I've installed swift from cepstral.com with alison's voice, and it works
fine,
Luki wrote:
> Here's how you do it.
>
> 1) In the DHCP server's config (dhcpd.conf) you specify the IP of the
> TFTP server:
> option tftp-server-name "66.55.44.33";
> This can be a remote server, as long as it's accessible by the device.
>
> 2) The factory settings on the Sipura devices (ATAs a
> I've been using mysql databases more and more. I've run
> across a couple
> of instances where I've either made a mistake on the ip
> address of the
> mysql database or for whatever reason, mysql wasn't
> running. In those
> instances, I've noted that the mysql command will hang
> indefinitely
Agreed - handling multiple calls and transferring them on a Snom is a
problem. Too fiddly.
Polycom phones work well in reception situations, if set up well.
Haven't tested the new Aastra's (but the Aastra transfer function works
well) but they would probably be OK too.
PaulH
On Fri, 2007-10-1
Hi Tzafrir!
Thank you for your answer, and my apologies for my delayed
response. I regret to say that the patch test's results were not succesful.
I shall describe the whole procedure in detail for you to establish whether
I did something wrong.
The fact is that I am as much a Linux beginner a
Hi to all!
I'm glad to be of any help. I had the same issue and sent an e-mail directly
to FWD. I paste the answer below.
Greetings,
Aldo
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Aldo,
FWD IAX is experiment
Baji Panchumarti wrote:
> you may be confusing the db server because mysqladmin
I made this name up, it's not the user name that I really use.
>
> no quotes around the password.
>
I'll test this out, thanks.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase
On 10/21/07, Doug Lytle wrote:
> Hey everybody,
>
> I've been using mysql databases more and more. I've run across a couple
> of instances where I've either made a mistake on the ip address of the
> mysql database or for whatever reason, mysql wasn't running. In those
> instances, I've noted
Hello, never posted to a mailing list before. I've been trying to work out this
problem for quite awhile now. I have a PHP script which is run whenever an
emergency situation happens. The script connects to the AMI and originates
calls to previously defined "emergency" extensions. I'm looking fo
> I'd like to be able to templatize a server, add a bunch of new handsets
> into sip.conf and extensions.conf, and then plug the phones into a
> network and have some DHCP and/or TFTP "glue" logic that sees the DHCP
> or TFTP request, and from it generates a boot file (an .XML file) and a
> respons
On Sun, 21 Oct 2007 13:31:28 -0400, Doug Lytle <[EMAIL PROTECTED]>
wrote:
>You'll want to look at the Privacy Manager:
Great :-) I'll take a look... once I can get the TDM card to pass the
CID number to Asterisk when it's actually sent by the telco.
Thanks for the tip.
_
Ian Hodgson wrote:
> Hello,
> Sorry for what may be a basic question, but I have spent a number of
> hours trawling the forums without resolving the problem, and hence this
> post.
>
> I have just started to dabble with Asterisk, as much for the
> learning than anything else. I created
Vincent wrote:
> On Sat, 20 Oct 2007 11:37:56 +0100, Alan Lord <[EMAIL PROTECTED]>
> wrote:
>> Look back a few hours in this mailing list for the message called "
>> IAX2: Incoming calls answered prematurely[RESOLVED]".
>>
>> I have included most of how I setup a simple IVR. It wasn't that har
Hey everybody,
I've been using mysql databases more and more. I've run across a couple
of instances where I've either made a mistake on the ip address of the
mysql database or for whatever reason, mysql wasn't running. In those
instances, I've noted that the mysql command will hang indefinitely
Thats a great step forward. Auto for PRI doesn't make sense... but
two configs to describe the same thing makes no sense.
/b
On Oct 21, 2007, at 1:03 PM, Tzafrir Cohen wrote:
> On Sun, Oct 21, 2007 at 11:57:45AM -0500, Brian West wrote:
>> It actually CAN but because someone was lazy and didn
On Sun, Oct 21, 2007 at 11:57:45AM -0500, Brian West wrote:
> It actually CAN but because someone was lazy and didn't want to
> actually do the work to make it possible to do a full change during a
> reload. The biggest issue is ztcfg would have to be absorbed into
> chan_zap to make it 100%
Vincent wrote:
> How would I go about prompting users for their phone number?
>
>
You'll want to look at the Privacy Manager:
http://www.voip-info.org/wiki/index.php?page=Asterisk+cmd+PrivacyManager
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a littl
It actually CAN but because someone was lazy and didn't want to
actually do the work to make it possible to do a full change during a
reload. The biggest issue is ztcfg would have to be absorbed into
chan_zap to make it 100% possible. In fact if Digium wanted to make
Asterisk easier to co
>
> Anselm Martin Hoffmeister wrote:
>
>> The problem there is that you have a very small "windows". AFAIK there
>> are no tftp servers that can generate files on-the-fly, so your script
>>
>>
>
You could make a perl script that pretends to be a TFTP server. Then it
could generate the
Hello,
Sorry for what may be a basic question, but I have spent a number of
hours trawling the forums without resolving the problem, and hence this
post.
I have just started to dabble with Asterisk, as much for the learning
than anything else. I created an account on FWD and used the Aster
On Sun, 21 Oct 2007 17:23:03 +0200, Tzafrir Cohen
<[EMAIL PROTECTED]> wrote:
>chan_zap cannot change signalling of a channel on reload. So that
>parameter is ignored on reload.
>
>False warning...
OK. So to check that Zaptel is correctly configured, I can just type
"zap show channels" in the CLI.
On Sun, 21 Oct 2007 10:59:49 -0400, "C F" <[EMAIL PROTECTED]> wrote:
>I believe that by reloading without restarting asterisk doesnt reload
>the signalling part
Thanks for the help. I did read this somewhere, so I typed "stop now"
in the CLI, followed by "safe_asterisk", "asterisk -r", and "reload
Hi
The first step I have to go through when users call into our
IVR is to handle the case where users' PBX hides their CID number. In
that case, I need to have them type their phone number (ten digits).
OTOH, those who call without hiding their CID number are sent directly
to the main men
On Sun, Oct 21, 2007 at 04:27:17PM +0200, Vincent wrote:
> ubuntu*CLI> reload chan_zap.so
> -- Reloading module 'chan_zap.so' (Zapata Telephony)
> == Parsing '/etc/asterisk/zapata.conf': Found
> [Oct 21 16:22:37] WARNING[8240]: chan_zap.c:11120 process_zap:
> Ignoring signalling
chan_zap ca
I believe that by reloading without restarting asterisk doesnt reload
the signalling part
On 10/21/07, Vincent <[EMAIL PROTECTED]> wrote:
> Hello
>
> I've been googling for this message, but can't find why
> Asterisk sends a warning. The configuration files look similar to
> http://www.voip-
Hello
I've been googling for this message, but can't find why
Asterisk sends a warning. The configuration files look similar to
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+zapata.conf.sample
It's a TDM card with just one FXO module on it, and I connected an
RJ11 cable to
Hello,
I have two issues which I would like to know whether someone has an answer to
them:
1. Our institute has over 8,000 phone numbers and I would like to allow
people to search it from the phone. I am willing to write some XHTML
scripts to run through the microbrowser, but I cannot fin
> Yehavi Bourvine +972-8-9489444 wrote:
>> In any case, I'll try this week to upgrade to 1.4.6 version and then add
>> IMAP
>> support and inform what happens.
>
> There have been _many_ IMAP related fixes sine 1.4.6. Please try the latest
> version, 1.4.13, instead.
>
> --
> Russell Bryant
So
Anselm Martin Hoffmeister wrote:
> Am Samstag, den 20.10.2007, 22:58 -0700 schrieb Philip Prindeville:
>> I'd like to be able to templatize a server, add a bunch of new
>> handsets into sip.conf and extensions.conf, and then plug the phones
>> into a network and have some DHCP and/or TFTP "glue" l
I have had ongoing echo problems with snom 360's, maybe the problem lies
with your phones...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stefan
Guenther
Sent: Sunday, October 21, 2007 5:20 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-us
Hello,
I have read the articles on echo cancellation
(http://www.voip-info.org/wiki/view/Asterisk), but couldn't find a
solution to my problem.
We are running Asterisk 1.4.12 together with an EICON DIVA SERVER BRI-2M
PCI (current driver from EICON) and some SNOM 300/360.
There are few clients whe
On Sat, 20 Oct 2007 11:37:56 +0100, Alan Lord <[EMAIL PROTECTED]>
wrote:
>Look back a few hours in this mailing list for the message called "
>IAX2: Incoming calls answered prematurely [RESOLVED]".
>
>I have included most of how I setup a simple IVR. It wasn't that hard to
>do and I have onl
Am Samstag, den 20.10.2007, 22:58 -0700 schrieb Philip Prindeville:
> Erik Anderson wrote:
> > On 10/20/07, [EMAIL PROTECTED] <[EMAIL PROTECTED]> wrote:
> >
> >> If you are trying to use non-complied ("XML") profiles... don't even
> >> bother wasting your time.
> >>
> >
> > Why is that? I'
[EMAIL PROTECTED] wrote:
> If you are trying to use non-complied ("XML") profiles... don't even
> bother wasting your time.
Oh. I _am_ using the XML format. When I initiate a resync over the
http server, it works fine, except the SPA doesn't start the regular
resync.
/Per Jessen, Zürich
--
This is little risky,
if some one got his account username/pws he will be able to send the traffic
allowing only IPs means he need to assign his IP then he can send traffic.
Is there no possibilities in asterisk to adding more host?
Thank You
> If you are only going to receive and
>not
Hi
Need help on this setup:
Incoming DID in H323 > Asterisk Server --> SIP Phone
please tell me to achieve this above setup what needs to be done in
Asterisk.
thanks
Arun
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