Hi all,
I've put together a quick Asterisk survey, in an effort to attempt to get
some insight into the overall direction of the project. I'd appreciate if
you could spend a couple minutes and run through this - I'll be happy to
share the detailed results with Digium or anyone else who's
inte
denon wrote:
At 06:44 PM 6/11/2003 -0400, you wrote:
On Wed, Jun 11, 2003 at 12:42:57AM -0500, denon wrote:
> We're doing a new * installation at a remote office soon, and I was
just
> curious what people's opinions were on hardware these days .. I've had
> decent luck with T100Ps and Adtran, bu
At 16:38 13-6-2003 -0500, you wrote:
Then I guess in zapata.conf before the definition of the
callerid=asreceived
channel => 1;FXO port
More generic would be to store the original callerid in a variable when the
call comes in at the receptionist desk and then reset it in the context
before t
> >Why not have dial just dial, then have applications like WaitForAnswer,
> >WaitForDisconnect etc...?
> >
> >This would give more granularity to the call flow control and allow
> >someone to get brave and write a WaitForHuman or whatever.
>
>
> Hmm... I can't think of too many instances where
Actually the fix isn't very hard and needs to be done anyway because the
same thing happens if you are checking voicemail and are left a message.
Let me add that to my localized list of things to work on here on my
vacation and see if I can get that knocked out, although committing to CVS
will like
Are you using voicemail2 or voicemail?
Can you confirm that /var/spool/asterisk/vm/403/INBOX has messages and/or
/var/spool/asterisk/voicemail/default/403/INBOX has messages?
Mark
On Thu, 12 Jun 2003, Derek Beaumont wrote:
> I have edited my zapata.conf file and I still cannot get voicemail
> n
The stutter dialtone should be longer now than it used to be (about twice
as long).
Mark
On Thu, 12 Jun 2003, Derek Beaumont wrote:
> In case anybody is wondering...
> I am in Canada, which apparently makes a difference. I found this in
> the archives:
>
> >>And this does actually work. There i
Just tell it to use wcfxs or something instead of torisa. most people
load the driver and don't count on autoloading of modules by the kernel
when they run asterisk.
Mark
On Thu, 12 Jun 2003, Steve wrote:
> Well after moving to a diff slot and rebooting twice I finally got:
>
> Zapata Telephony
Can you run with valgrind instead so we can find the true source? install
asterisk with "make valgrind" and then run:
valgrind --attach-gdb=yes ./asterisk -vvvg
and then "asterisk -rx reload" and see where it tries to make you attach.
Talk to Martin for more info ([EMAIL PROTECTED]) next week if
It would take a fair amount of modification to add "review and commit"
functionality, but if someone wants to do it, just look at
"play_and_record" and add another parameter to add the review
functionality.
Mark
On Thu, 12 Jun 2003, Test wrote:
> Hi,
>
> The voicemail app allows you to record yo
Actually the "easy" way would be to use the seek functionality with
absolute time relative to when the recording began, combining Steven
Critchfield's work with Mahmut's. That way even if there is silence in
the middle where there are no packets, we "seek" to the right place in
time to record.
Ma
hi all
the help output for DISA ends like below, with the half-sentence 'Note that in
the case'
what's the rest of that sentence?
The file that contains the passcodes (if used) allows specification
of either just a passcode (defaulting to the "disa" context, or
passcode|context on each line of
When you say "stops responding" do you mean no more pings, telnet dead,
etc? Or do you mean asterisk stops responding? Is there a segfault or
kernel panic, or any other failure diagnostic?
Mark
On Thu, 12 Jun 2003, Alex Zarubin wrote:
> Zaptel was compiled with -D__SMP__
>
> We've installed ir
Oh, and one more thing... Does the problem occur with SMP mode and only
one card?
Mark
On Thu, 12 Jun 2003, Alex Zarubin wrote:
> Zaptel was compiled with -D__SMP__
>
> We've installed irqbalance and the picture improved a lot
> (thanks to Jared Smith). Do you still see problems in our /proc/in
A friend pointed out this url
http://www.cisco.com/univercd/cc/td/doc/pcat/clmn32.htm where it lists
intercom/auto-answer as being a feature in Cisco Call Manager (which as I
understand it, uses SIP predominately for handsets). I've come
across comment somewhere that intercom isn't supported in th
Can anyone point us to a document which would describe the difference
between Telstra PRI and everything else?
Mark
On Fri, 13 Jun 2003, Peter Armstrong wrote:
> You need to get the ETSI or Euro version of PRA from Telstra and then it
> will work, they offer it as well as their quaint version of
John,
On Wed, 2003-06-11 at 21:58, John Todd wrote:
> >This is slightly off-topic I suppose, but:
>
> 2) The ENUM system is centralized. TRIP can be established between
> two telephone systems, independently of any third party's cooperation
> or assistance. Routes can be exchanged in any
A list of Dialogic compatible hardware is on our web site. We also have
another contact at Dialogic that has expressed more interest in Asterisk
compatibility so perhaps we can pull some more from there, too. I'd
rather try to figure out what is different about your E1 switch that
causes us not t
If someone wants the message, just buy it from thevoice.digium.com and
i'll add it to Asterisk as a sound.
Mark
On Thu, 12 Jun 2003, Dave Packham wrote:
> Yeah thats a good one but it does not give them the message "this number does not
> accept telemarketing calls, if you are a telemarketer t
Sounds like a timing issue most likely. Do you get them frequently or
infrequently? It's okay if you get one every few hours or days but if
you're getting them every few minutes with normal call volumes then
something (likely timing) is wrong.
Mark
On 13 Jun 2003, Levent Guendogdu wrote:
> Hi
Note that in the case of specifying the numeric-passcode, the context must
be specified if the callerid is specified also.
If login is successful, the application parses the dialed number in the
specified (or default) context, and returns 0 with the new extension context
filled-in and the priority
Hi,
I am sure this question has been asked a hundred times. I am planning on
purchasing the Dev Kit to experience with * and do some voip configurations.
However, I see that Quicknet phonejack and linejacks are compatible. IS it
worth the time to try Quicknet hardware or is it a waste ? My plans ar
I have asterisk configured on a machine connected to Internet by a cable
modem with a public ip. The same network card has a private lan address
which I'm trying to use to play with an asterisk configuration with X-lite
or an softip phone.
sip.conf has bindaddr=192.168.0.1 [the private address of
--On Saturday, June 14, 2003 12:56:09 -0400 Francisco Perez-Landaeta
<[EMAIL PROTECTED]> wrote:
Hi,
I am sure this question has been asked a hundred times.
Yes - search the list.
I am planning on
purchasing the Dev Kit to experience with * and do some voip
configurations. However, I see that Q
I recently had a Active InternetPhoneWizard USB sent to me from iconnecthere.
Although they push the software that iconnecthere
runs, I figured there is a way to make use of it exclusive of iconnecthere. Has anyone played with this device? It would
make for a cheap way to get a connection t
> >Why not have dial just dial, then have applications like WaitForAnswer,
>WaitForDisconnect etc...?
>
>This would give more granularity to the call flow control and allow
>someone to get brave and write a WaitForHuman or whatever.
Hmm... I can't think of too many instances where the functio
The answer is "No", categorically, nor is their documentation correct
in saying that there is a web server running on SIP phone software
(it's only on CCM images.)
.
.
.
However...
A bad hack would be to use "expect" and "telnet" to open a connection
to the 7960, and activate it's testing mode
Willi -
I think you misunderstand my point. The fact that TRIP is not open
to all parties is it's strength, and does not "break" it. ENUM is
for identifying individual destinations for completely-qualified
phone numbers. TRIP is for identifying gateways to aggregated blocks
of phone number
The "tone" being referenced here is the 2100Hz echo cancel disable tone.
Mark
On Fri, 13 Jun 2003, Martin Pycko wrote:
> I think when you exceed the txgain or rxgain settings than the echo
> canceller might turn off.
>
> You can find if the pending call has echo canceller turned on when you do
>
You're not trying to run it on a BRI are you?
Mark
On Fri, 13 Jun 2003, Steve Underwood wrote:
> Jorge wrote:
>
> >Hi,
> >
> >I have an E100P card.
> >
> >My zaptel.conf is:
> >span=1,0,0,cas,ami,crc4
> >bchan=1-2
> >dchan=3
> >loadzone = us
> >defaultzone=us
> >
> >after execute ztcfg red led b
> I've seen this too, and have wondered exactly what it means. Does it
> mean that the echo canceller has been disabled for that call? That
> channel? That entire span? The whole card? When does the echo
> canceller get turned back on, if ever?
The 2100 hz echo cancel disable tone only disabl
Is it possible your "asterisk.conf" reflects different directories for
storing your modules?
Mark
On Fri, 13 Jun 2003, Moshe Yudkowsky wrote:
> At 09:32 2003-06-13 -0500, you wrote:
> >Since you're using the sound card for testing you need to
> >change in the /etc/asterisk/alsa.conf or oss.conf
Yes, that's right, but you might need to use:
switch => IAX2/user:[EMAIL PROTECTED]/context
Mark
On 13 Jun 2003, Eric Wieling wrote:
> As I understand it (and my understanding is obviously incorrect) the
> switch => statement sells the Asterisk box to resolve (aka lookup)
> extensions by queryi
That's the right way.
MJark
On 13 Jun 2003, Brancaleoni Matteo wrote:
> Hi.
>
> I was wondering how can I make incoming calls to wait if the phone
> operator is busy. I've 8 incoming lines, with 30 extensions.
> What I need is if the operator is busy with call nr #1 , the new
> incoming call wai
In general it has to support a full duplex API. I would suggest
contacting Gerry Gilmore to ask about compatibility, although as I
understand it all the ones on the web site are the ones that are
supported.
Mark
On Fri, 13 Jun 2003 [EMAIL PROTECTED] wrote:
> Hello all,
>
> I have a couple of Di
Hi All,
OS: RedHat Linux 7.2
Machine; x86
Does Asterisk supports Dialogic D/41E?
regards,
-Ahmed.
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
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It should work, but there is a fee of $30 per channel for the software.
Check the archives
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Saturday, June 14, 2003 12:44 PM
Subject: [Asterisk-Users] Dialogic D/41E
> Hi All,
>
> OS: RedHat Linux 7.2
> Machine
I finally got the power supply for my 7960 and am having problems getting
it working. What should be in sip.conf and the SIP(macaddr).cnf file?
This is what I have in SIP0002FD3BA8F7.cnf
# SIP Configuration Generic File
# Line 1 appearance
line1_name: Asterisk Test
# Line 1 Registration Authenti
I checked that; the directories point to the correct places.
As I mentioned in a follow-up message, "strace" shows that my system
doesn't even try to load the modules. I can "load app_playback.so", etc.,
followed by "load pbx_config.so" to get a dialplan -- but then the system
doesn't find the
On Sat, Jun 14, 2003 at 09:33:27AM -0500, Mark Spencer wrote:
> The stutter dialtone should be longer now than it used to be (about twice
> as long).
VZ NYC stutter dialtone is about two seconds I think, maybe three. I
can't verify it now as I have cancelled the voicemail service. :-)
--
Scot
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