Hi,
I'm using an ATL IP400 phone and cant get it to register, it fails with:
chan_sip.c:9405 handle_request_register: Registration from 'xx sip:[EMAIL
PROTECTED]'
failed for 'x.x.x.x'
looking at the register request i notice two things:
Authorization: Digest
saying the call has ended. However, the RTP (audio) stream will
stop. The rtptimeout setting lets you define a time period that after
x seconds of no audio packets, it's assumed the SIP client has gone
away and the call should be terminated.
Jeremy
Stephen J. Wilcox wrote:
Hello,
can
Hello,
can anyone help with my problem below, searching doesnt show any results..
thanks
Steve
On Wed, 3 Aug 2005, Stephen J. Wilcox wrote:
Hi,
I'm seeing a problem where if I place a call, then forcibly quit or turn off
the client the call stays active.
The frames counters stop so
Hi,
I'm seeing a problem where if I place a call, then forcibly quit or turn off
the client the call stays active.
The frames counters stop so its apparent the client has gone away but the call
remains active.
Asterisk is CVS-HEAD 23-Jun-05
What is supposed to happen in this scenario?
send faxes out from your system, what do they look like
at the remote end?
Steve
-Original Message-
From: Stephen J. Wilcox [mailto:[EMAIL PROTECTED]
Sent: 13 July 2004 18:12
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] fax still fails, ideas sought! Re: rxfax/spandsp
fails
clock sync issues.. my zaptel.conf is set to
use the PRI as primary clock, i have no evidence of issues altho dont know how
to check (other than the call quality is fine, no clicks, no pri down/ups).
What can i try?
Steve
On Mon, 12 Jul 2004, Stephen J. Wilcox wrote:
Hi,
I just sent
Hi,
I just sent this to Steve Underwood, but then found a bunch of posts on the
mailing list about similar issues.. does anyone have the fix?
I'm running asterisk CVS-HEAD-06/28/04-18:13:13, spandsp 0.0.1k, libtif 3.5.7
one thing i just noticed is that calls come in with format '72' which is
the syntax is wrong
its $[x operator y]
you need the $ you need the spaces.
Steve
On Mon, 12 Jul 2004, Steve Woolley wrote:
It is possible and probably most likely that it is being interpreted as
a string. The real question is though, why?
I believe it is syntactically correct.
Date:
Both the question and the answer are not talking about QoS.
From the Q, qos does not provide a measure of quality, it provides a system to
allow you to request your data be handled according to priorities.
From the A, qos is confused with the pstn.. qos is a feature of IP, that has
nothing to
its not on there is they sell
GAPS so yeah, its reverse engineering unless you care to pay for their
system.
- Original Message -
From: Stephen J. Wilcox [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, July 08, 2004 4:01 PM
Subject: Re: [Asterisk-Users] sample config file
I was wondering about that too..
Following the instructions on that page for config did not work for me. Setting
up a config file like the sample one made no difference to the phone (I can
confirm it did tftp it okay). Also the method references md5 checks and I dont
see that at all.
I tried
he did i tell you! i'm still in counselling... altho come to think , i was in
counselling anyway..
Steve
On Mon, 5 Jul 2004, Bradley D. Thornton wrote:
. Monday, July 5, 2004
15:50:06 (-08:00hrs UTC)
Hello
A trace to the IP gives valuenet : http://www.valuenet.net/ a colo provider on
Level3.
I have a dislike for this kind of targeted spam on mailing lists, and are they
harvesting email addresses from their subscription .. I suggest nobody contact
them else they will think this is acceptable.
I
FYI I see it only on 1 in about 10-20 pickups...
On Wed, 19 May 2004, Luis Vazquez wrote:
Shaun Ewing wrote:
I'm not seeing this - using stable CVS from 14-05-2004.
Phone types are Grandstream Budgetone 101 using firmware 1.0.4.68, and Cisco
7940 using SIP 6.2.
-Shaun
Just to
I see this (but not using a recent asterisk version)
I had put it down to a software bug in the grandstream phones that I'm using -
are you sure its an asterisk bug or are you using grandstream also?
Steve
On Mon, 17 May 2004, John Vogel wrote:
I upgraded to the latest stable version of
Hi Mark,
its sets the context in zapata.conf.. put a context=contextname somewhere
above where you define the channels
then in your extensions.conf, create the context if you've not already, start
with a exten= _.,1,Dial(blah...) then modify from that to suit your setup
Steve
On Mon, 1 Mar
Hi Jimmy,
I have CDR issues also and was thinking of seeing if I could write a patch to
fix, my particular problem which I think you include below alebit subtley is
that there are numerous circumstances where you would expect multiple CDRs and
you currently get one minus bits of vital detail
As for the rest of the questions I can't reallt answer as I have never
personally connected an E100P to an ISDN30 line.. many on this list have
and will hopefully be able to give you more of the technical details..
I have, for those unsure its no different from a T1 except for how many
Ok I'm not going to tell you but I will only include this one line from your
config:
cannel = 1-15,17-31
See if you can work it out...
Steve
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi,
does anyone know if its feasible to run asterisk with a PRI card but not run
any q931 signalling.. basically push calls down the PRI and tell asterisk in
some other way to pickup a particular Zap channel?
Steve
___
Asterisk-Users mailing list
On Tue, 13 Jan 2004, John Todd wrote:
does anyone know if its feasible to run asterisk with a PRI card but not run
any q931 signalling.. basically push calls down the PRI and tell asterisk in
some other way to pickup a particular Zap channel?
Well, not quite PRI nor quite what you're
If you've got spans from different providers...you're in for an
adventure. You'll be able to do one of the following (which one is telco
and luck dependant):
So what you're saying is that the TE410P is not capable of *independently*
clocking each of the T1s. Hell even the venerable old
What are the practical effects with in-correct clock sync
-like to you hear odd buzzing, or dropped voice or gaps of audio ??
You may get gaps where frames are discarded, this will be across all timeslots
so an individual loss isnt a lot of data, you'll probably get away with the odd
one but
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Stephen J.
Wilcox
Sent: Thursday, January 08, 2004 6:57 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] Asterisk hanging?
I did a search but didnt find it, unless you know exactly what to search for
its
hard to locate specific things
dialling
through analog lines).
Matteo.
Il gio, 2004-01-08 alle 18:27, Stephen J. Wilcox ha scritto:
Hi,
okay heres what I want to do .. simple ivr, we take a call, answer it, play a
menu, dial out based on options. No problems so far.
The CDR always shows the call as answered
Hi,
I compiled and am running the latest CVS but strange things are now happening..
it looks like asterisk is randomly declaring my calls to be fax calls,
complaining and then sending the calls into a black hole... if I hangup the
calls below (soft hangup) asterisk locks up and I have to kill
Hi,
okay heres what I want to do .. simple ivr, we take a call, answer it, play a
menu, dial out based on options. No problems so far.
The CDR always shows the call as answered as I answer the 1st leg to play the
prompts, I am actually more interested in if the 2nd leg - the outbound part -
..
Steve
On Thu, 8 Jan 2004, Eric Wieling wrote:
The fix that is in the mailinglist archives didn't help you?
On Thu, 2004-01-08 at 09:35, Stephen J. Wilcox wrote:
Hi,
I compiled and am running the latest CVS but strange things are now happening..
it looks like asterisk is randomly
Hi,
does anyone have any recommended (read tried and tested) way of making asterisk
be able to handle incoming faxes.
I've a PC running asterisk with a digium E1 card in and simply want to be able
to route a call to some application which will take a fax call and save the fax
as an image.
I
You are trying to have both ends act as users, cisco can support emulating a
network interface (isdn protocol-emulate in serial interface config) but in my
experience i could get the circuit up but it would bounce and i couldnt get
signalling to work.. to be fair my IOS is quite old and wouldnt
Can you set outbound callerid on a channelized T1?
DTMF passed before the call. It is unlikely though that you will find
someone interested in doing that though. It is easier/cheaper to drop a
PRI into somewhere and then outbound caller ID isn't kludgey with DTMF.
The service you might
cant give each channel
their own settings]
Steve
On Fri, 26 Dec 2003, Brian West wrote:
I understand PRI can do that but CT1 was my question. CT1 != PRI
bkw
On Fri, 26 Dec 2003, Stephen J. Wilcox wrote:
Can you set outbound callerid on a channelized T1?
DTMF passed before
I'm using grandstream phones, when on a call and a second call comes in
the
call waiting indication is to play ringing which means you cant actually
hear
your original call. I want to stop this but cant, heres my options
1. Change the callwaiting indication, I assume this is produced
Hi,
I'm using grandstream phones, when on a call and a second call comes in the
call waiting indication is to play ringing which means you cant actually hear
your original call. I want to stop this but cant, heres my options
1. Change the callwaiting indication, I assume this is produced by
34 matches
Mail list logo
