I'm trying to figure out how to start *.
Rh7.3,CVS,TE410P,TA750
If I just try the way the docs spell it out
/usr/sbin/asterisk -vvvc it fails..
/var/log/asterisk/messages
Oct 3 22:23:34 WARNING[1024]: File chan_zap.c, Line 610 (zt_open):
Unable to open '/dev/zap/channel': No such
Even Birmingham UK is not near enough :(
--
Dave Cotton [EMAIL PROTECTED]
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Any ideas on the following? (CVS 10/01/2003)
Only reference I could find was a Zaptel change log update...
2003-09-02 18:23 martinp
* wct4xxp.c (1.6): Get rid of the Double missed interrupt message
every time you load the driver
and an email refering this to serial console usage.
Something I
The differences in VM2 and the ability to create VM contexts
for things like virtual PBX's on one box, VM2 allows you to
modify the email that gets sent when a voicemail is recieved
and a few more config features..
How do you modify the emails ?
Are there other configfiles ?
/M
Chris Albertson wrote:
I haven't found any open source/freeware software phones that run under
Windows 2000 that I like or that even work well. What are other people using?
X-Lite is about the best you are going to find..
Later..
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Micke Andersson wrote:
The differences in VM2 and the ability to create VM contexts
for things like virtual PBX's on one box, VM2 allows you to
modify the email that gets sent when a voicemail is recieved
and a few more config features..
How do you modify the emails ?
Are there other
Martin Pycko wrote:
take out usecallerid=yes in zapata.conf
Martin
Thanks Martin, but my zapata.conf is :
[channels]
echocancel=yes
echocancelwhenbridged=yes
busydetect=yes
busycount=6
context=incoming
signalling=fxs_ks
group=1
channel = 1-2
Perhaps I need a usecallerid=no in there. I'll
You can also use the AGI interface function RECORD FILE and specify a max
record duration of 5s and silence detection of 1s. Time the duration of the
call to asterisk - if its longer than 1 second you know you've got voice. If
you need to check for voice over a longer period of time - repeat the
Hi, Most embarrased newbie evere here again.
Possibly another daft question. I have the digium
starter kit lite, so I've got the single FXO and FXS lines
All is working well with local sip phones able to dial other phones,
conferencing, MOH (Thanks Asterisrk-users list!) along with the one
Everyone seems to be working on their own servers
that are in there homes, offices and elsewhere. The only common thread is this
list-serv.
I haveseveral * servers set up here in
Austin, Texas. I propose we set up one of them with with all the list-serv
members. At this time the calls would be
sip wrote:
Everyone seems to be working on their own servers that are in there
homes, offices and elsewhere. The only common thread is this list-serv.
I have several * servers set up here in Austin, Texas. I propose we
set up one of them with with all the list-serv members. At this time
the
IAXTEL is a 1-700 system designed for on-net calls.
We have very low-cost PSTN lines to 48 states for no-cost long-distance
dialing. I plan to add some of these lines to the system. As servers are
added in other cities around the world I envision dialing Berlin, Germany
from Austin, Texas and
I welocme the idea.After all we are deploying and using a voice based
technology so we can have something like Voice Mailing List where we will
be having voice messages and realtime talks with other programmers and
developers.
I think this should set a precedent for other mailing list also.
sip wrote:
IAXTEL is a 1-700 system designed for on-net calls.
We have very low-cost PSTN lines to 48 states for no-cost long-distance
dialing. I plan to add some of these lines to the system. As servers are
added in other cities around the world I envision dialing Berlin, Germany
from Austin,
It won't cause any sort of serious problem, but you are getting it with an
unusually high frequency. What's particularly interesting is that it
occurs almost exactly once per minute, on the minute. This would seem to
suggest that you have some hardware in your system which, once per minute,
is
That is right!!! We will enable the Voice-Mail Email-Mail function. That way
if you call a member and he is not there you leave him a message...it is
packaged in a wav file...then Emailed. The next time he checks his email he
will have the voice mail also!
- Original Message -
From:
Each server would update from a master SQL database at a predetermined time.
This way all servers would be in sync.
- Original Message -
From: WipeOut [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, October 04, 2003 9:55 AM
Subject: Re: [Asterisk-Users] Let's TALK ABOUT IT!!!
At 09:01 4-10-2003 -0500, you wrote:
IAXTEL is a 1-700 system designed for on-net calls.
We have very low-cost PSTN lines to 48 states for no-cost long-distance
dialing. I plan to add some of these lines to the system. As servers are
added in other cities around the world I envision dialing
sip wrote:
Each server would update from a master SQL database at a predetermined time.
This way all servers would be in sync.
Ok I get it, so there is going to be some app or script that will update
the DB from a central source, and an AGI written for Asterisk that will
do the lookup in the
After re-installing, I am finding that when dialling into an X100P, that
Answer is now answering on the second ring, where it always used to
answer on the first before.
In the console, Starting simple switch on 'Zap/1-1' appears halfway
through the first ring and Executing
But it would be a free call to the common man who had a fast internet
connection and a softphone or IP phone. He doesn't have to have a server or
know the tech stuff... just a free softphone and he is in.
After all, we are all working to develop this industry...build
servers...sell phones...etc.
Each server would update from a master SQL database at a predetermined
time.
This way all servers would be in sync.
Ok I get it, so there is going to be some app or script that will update
the DB from a central source, and an AGI written for Asterisk that will
do the lookup in the DB that
Everyone seems to be working on their own servers that are in there
homes, offices and elsewhere. The only common thread is this
list-serv. I have several * servers set up here in Austin, Texas. I
propose we set up one of them with with all the list-serv members.
At this time the calls
This entire idea will only work for the USA (and the few other countries
that have unmetered local calls). HOWEVER, in most countries ALL calls,
even if you call the person across the street are billed by the min.
--Eric
On Sat, 2003-10-04 at 10:33, sip wrote:
But it would be a free call to
On Sat, 2003-10-04 at 09:57, Mark Spencer wrote:
It won't cause any sort of serious problem, but you are getting it with an
unusually high frequency. What's particularly interesting is that it
occurs almost exactly once per minute, on the minute. This would seem to
suggest that you have some
Yeah, I'd put usecallerid=no since I bet it's set by default as yes.
Martin
On Sat, 4 Oct 2003, Richard Scobie wrote:
Martin Pycko wrote:
take out usecallerid=yes in zapata.conf
Martin
Thanks Martin, but my zapata.conf is :
[channels]
echocancel=yes
echocancelwhenbridged=yes
But would you then not be able to use caller id from the telco?
On Sat, 4
Oct 2003, Martin Pycko wrote:
Yeah, I'd put usecallerid=no since I bet it's set by default as yes.
Martin
On Sat, 4 Oct 2003, Richard Scobie wrote:
Martin Pycko wrote:
take out usecallerid=yes in
Actually, if this was to be done, it might be an idea to do it with DNS, so
client machines would just do
Dial(IAX2/[EMAIL PROTECTED]/442071234567) and the DNS
system would resolve which machine is the correct target - no cleverness at
all required at the client end, so implementation would
Hi,
I am trying to set up asterisk to register to gnophone/IAX for me.
BUT: The e-mail i received when I set up my account shows that
there is an @ sign contained within my password
So, when I try to place a call I see the following: (No, that is NOT my
real password, and yes, I have
Hi All
Just to query if anyone on this list or rather working
on/using/developing on Asterisk in India (especially
Mumbai)?
Regards.
Sanjay.
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I think that assumption is wrong. The repeated messages are when syslog
decides to empty the buffer. As far as I know, it isn't possible to
turn off that and get real timing information.
Oct 3 22:48:01 cti-350 kernel: TE410P: Double/missed interrupt detected
Oct 3 22:49:01 cti-350
Eric == Eric Wieling [EMAIL PROTECTED] writes:
Eric Check /proc/interrupts to make sure the cards are not shareing
Eric IRQs with anything.
Is there anything that can be done so that this is not a requirement?
There are (many) setups where this is simply not possible.
Other cards can share
Steve == Steve Meyers [EMAIL PROTECTED] writes:
Steve On Thu, 2003-10-02 at 12:04, Jan Rychter wrote:
I'm also hearing this, with an analog phone (connected to an
S100U). Rather annoying.
Incoming calls have an entirely different problem for me, a
disastrous 5-8 second
Mark Spencer:
The anti-patent clause was dropped ages ago.
What do you mean? I can still see it in the LICENSE file in Asterisk.
--J.
On Fri, 3 Oct 2003, Uriel Carrasquilla wrote:
So, is Astrisk being changed to an OSI-compliant license without the
anti-patent clause?
Uriel
A little off-topic, but does anyone know if it is possible to disable call
waiting on the Grandstream phones? I can't find anything in the
configuration that would allow this.
Linus
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On Sat, 2003-10-04 at 12:02, Lists wrote:
But would you then not be able to use caller id from the telco?
CallerID on an analog line is sent between the first and second ring
normally. So if the requester wants callerid and first ring answer, he
will have to move to PRI.
Some choices have
I think he means the old You can't use Asterisk if you pursue patent
litigation against any Open Source project
Mark
On Sat, 4 Oct 2003, Jan Rychter wrote:
Mark Spencer:
The anti-patent clause was dropped ages ago.
What do you mean? I can still see it in the LICENSE file in Asterisk.
I had exactly the same problem. You change your password at the gnuphone
website and it says it doesn't update your IAXTEL password, but it does! I
changed mine, used the password I selected, and was able to use IAXTEL with
no problem after that.
Cheers
Paul
I've bought a X100P two months ago and I didn't get
to make it working properly with another non-Asterisk
solution(http://www.planet.com.tw) even trying everything
suggested here..
1) Uncomment the -DWANT_G729
2) Apply fragmentation patch
3) Use chan_h323 or chan_oh323
4) Uses G729A, B and AB...
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I have tried loads of configurations but I cannot get it to work.
I basically have three computers and want to use soft phones on two of them
to connect to asterisk so I can use them instead of dialling into my X100P
card as the phone bill is getting bigger.
I assume that I can configure * to
I have a pair of 7940 ip phones and a standard analog phone on my test system. I have
a few issues:
1. When dialing out (using m option) via SIP to another another phone Zap or SIP, I do
not hear anything moh or ringing.
2. When two SIP phones are talking and either phone is placed on hold
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