Welcome to the very much less than wonderful world of Cisco software
support. When will those guys simply make the software downloadable
straight away from their website for a modest fee?
Iain
--On Saturday, March 27, 2004 1:43 am -0600 Mitchell S. Sharp
[EMAIL PROTECTED] wrote:
I just
If you pay 8 USD for 1 year support you can download the image :)
Best regards,
Chris HARIGA
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Iain
Stevenson
Sent: Saturday, March 27, 2004 4:06 AM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Cisco
.. not sure this applies outside the US - or I'd reach for the credit card.
Iain
--On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA
[EMAIL PROTECTED] wrote:
If you pay 8 USD for 1 year support you can download the image :)
Best regards,
Chris HARIGA
-Original Message-
From:
Michael
Yes is works simular to the Zap stuff
You put context=whateverIncomingContext in the sip.conf block for the
device or port. Like I have everything that comes in on SIP go into the
context sipfriend execpt my 2 ata ports and the 2 lines of the 7940
phone.
Looks something like this.
I configured agi-test.agi on extension 111 when i dial into asterisk
extension 111 using a IAX softphone and hangup while the AGI is playing
asterisk crashes. Does anyone have any idea why this happens.
--
regards
Vikram (http://www.vicramresearch.com)
Are you making calls out to Nufone or simply from one of your servers to
another? We noticed this problem when we upgraded one of our servers to
the latest CVS and left another one with an older version. Seems that
the latest changes with rtp.c need to be applied everywhere.
When we upgraded
Andres wrote:
Are you making calls out to Nufone or simply from one of your servers to
another? We noticed this problem when we upgraded one of our servers to
the latest CVS and left another one with an older version. Seems that
the latest changes with rtp.c need to be applied everywhere.
Andres wrote:
another? We noticed this problem when we upgraded one of
our servers to
the latest CVS and left another one with an older version.
Seems that
the latest changes with rtp.c need to be applied
everywhere. When we
upgraded all servers then the audio returned to
Iain Stevenson wrote:
.. not sure this applies outside the US - or I'd reach for the credit card.
Iain
--On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA
[EMAIL PROTECTED] wrote:
If you pay 8 USD for 1 year support you can download the image :)
Best regards,
Chris HARIGA
-Original
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct timeval
*delivery)
{
struct timeval now;
unsigned int ms;
if (!rtp-txcore.tv_sec !rtp-txcore.tv_usec) {
gettimeofday(rtp-txcore, NULL);
}
gettimeofday(now, NULL);
Iain Stevenson wrote:
.. not sure this applies outside the US - or I'd reach for
the credit card.
Iain
--On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA
[EMAIL PROTECTED] wrote:
If you pay 8 USD for 1 year support you can download the image :)
Best regards,
I have a few 7940G phones on a LAN hub, and I'm looking at the 3COM
3CNJPSE power injector. Can I put one of these behind my LAN hub and
power all the phones, or do I need one for each phone?
From the spec, it looks like PoE tries to discover whether a device is
powered over ethernet. Can I
Michael Welter wrote:
I have a few 7940G phones on a LAN hub, and I'm looking at the 3COM
3CNJPSE power injector. Can I put one of these behind my LAN hub and
power all the phones, or do I need one for each phone?
From the spec, it looks like PoE tries to discover whether a device is
powered
Hi,
I trying to get agi with perl to stream a gsm file , and wait for a
digit , the agi gets to the stream but doesn't play back, could some one
explain how this works
here is a snip it of code
open(DAT,/etc/asterisk/1571.log) || die(Cannot Open File);
while( $sth-fetch() ) {
print DAT in
--On Saturday, March 27, 2004 4:52 pm -0500 Ray Burkholder
[EMAIL PROTECTED] wrote:
Iain Stevenson wrote:
.. not sure this applies outside the US - or I'd reach for
the credit card.
Iain
--On Friday, March 26, 2004 11:29 am -0500 Chris HARIGA
[EMAIL PROTECTED] wrote:
If you pay 8 USD
All three of my servers are at the same level of cvs checkout (within
minutes of each other), I believe from March 22, 2004. All of my calls
to NuFone are using GSM, though I allow iLBC as well.
Thanks for the response, I was beginning to think my questions were
invisible :-)
Andres wrote:
Does anyone know if it is possible to force a extension to not allow
transcoding? If you spec canreinvite=yes the cal may still transcoded if the
parties do not choose a the same code on each end. In my situation it is
better that the call fail than have it transcoded.
Also, I see some limited
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Hi Ray,
I did report this to Mark Spencer a few days ago. Look at my last
comment in bug: http://bugs.digium.com/bug_view_page.php?bug_id=0001260
Maybe you can open up a bug report and provide your solution.
Ray Burkholder wrote:
static unsigned int calc_txstamp(struct ast_rtp *rtp, struct
Is it possible to maintain two separate and distinct IAX2 connections
between a pair of Asterisk servers?
Here is the scenario: I use VoicePulse Connect for inbound and outbound
calling. I have a friend who also has
an account and I would like to add his profile to my Asterisk Server. I
added
I am trying to get asterisk working with H323 - but I don't seem to be
getting very far. I have downloaded the asterisk-oh323-0.5.10 package, but
cannot seem to compile it ; I get the following errors:
... [ a lot omitted ] .
wrapper_misc.cxx:72: error: parse error before `else'
What you and so may others on this lise seem to forget is that Cisco is a company
offering bsuiness products for businesses. Businesses typically pay by check and wire
transfer, especially for items such as this.
If you want home-user pay-by-credit-card service, buy products from Belkin's
Hi List,
I've just built a new * box (slackware 9.1) and I get
no sound from a Playback(tt-weasels) command.
I've got other slack9.1 boxes running.
* Version is v1.0 stable
exten = 213,1,Answer
exten = 213,2,Playback(tt-weasels)
exten = 213,3,Playback(tt-weasels)
exten = 213,4,Hangup
when
I'm running slackware 9.1 as well, and found that mpg123 doesn't come with
slackware, so you may need to fetch and install it to get playback
working...
http://www.voip-info.org/tiki-index.php?page=mpg123
http://www.mpg123.de/
I know I couldn't get MOH working until I installed mpg123, can't
Darnell Gadberry [EMAIL PROTECTED] wrote:
Is it possible to maintain two separate and distinct IAX2 connections
between a pair of Asterisk servers?
Yes and no, see below. :-)
Here is the scenario: I use VoicePulse Connect for inbound and outbound
calling. I have a friend who also has
an
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