Bohuslav Coufal wrote:
I'm looking for that one too. I had not been succesfull up to now.
Bob.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tomasz
Chmielewski
Sent: Thursday, October 27, 2005 1:58 PM
To: Asterisk Users Mailing List - Non-Commercial
Doug Lytle schrieb:
(...)
Is it possible to somehow read spandsp / txfax exit codes?
Run Asterisk in debug mode [asterisk -d] and use the -debug option on
the spandsp command line. Mine is as follows:
exten = s,3,rxfax(${FAXFILE}.tif,DEBUG)
After I get the debug output, I use cat
Hi,
does anybody have a working sample configuration of a cisco as53xx for
receiving faxes ?
Sending faxes over the as5300 works fine, but if I send a fax from pstn to
asterisk (over the as5300 as pstn/voip gateway) it does not work.
Thx, florian
So much for not stepping on toes.
Incidently, there have been dev-asterisk posts in the past relating to ADSI
tones being processed through a SIP channel, so theoretically, a softphone
'could' exist. I've been hard-pressed just to find any documentation via
google explaining any
We have 60+ members loged into the queue and talking to 5-10k people a day.
I need a better way to track them loggin in and out. The queue_log gets
really big fast. And has data we dont need. Is there anyother way to
track when someone loges in and out. I can write to a different file
when
We would like to beable to listen in and interact with the person in a
queue, talk to our agent and NOT have the other person be able to hear
us. Is there a way to do this?
Kyle
___
--Bandwidth and Colocation sponsored by Easynews.com --
Does anyone know how to silence the audible mwi on
a soundpoint ip500 or ip501 running sip 1.4.1?
I tried changing just about all the se.pat.callProg.11
vars and nothing seems to change.
--
Bob Knight
[-w] the work option
[EMAIL PROTECTED]
925-449-9163
Hi,
Sorry this does not answer your question.
As I am trying to implement fax on Asterisk, can you please tell me if you are using spandsp? Are you sending fax from SIP ATA's?
Thank you.
AK
On 10/27/05, Florian Meister [EMAIL PROTECTED] wrote:
Hi,does anybody have a working sample configuration
[EMAIL PROTECTED]
1.2.0 beta4 writes to the respective voicemail directory and when the call is
hung-up the .wav file disappears.
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--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
I have a small issue with some remote users connecting to my primary
Asterisk server using 1.0 Every few seconds, there is a subtle tick and a
very small amount of jitter. The tick is not consistent i.e. it could be in
2 seconds, could be 5, could be 10. This does not affect core functionality,
On Friday 28 October 2005 12:06, Richard Smith wrote:
[EMAIL PROTECTED] 1.2.0 beta4 writes to the respective voicemail directory and
when the call is hung-up the .wav file disappears.
Sounds like voicemail.conf is setup to delete the message after it is emailed
to the user.
You may also want
Hello all,
I have a problem calling into asterisk on a PRI going out to a SIP phone
(PRI - SIP). The calling party does not hear ringing and after about five
seconds gets an *All circuits are busy* recording. However, the called SIP
phone does ring, and if the called party answers the phone
Olle --
the version at the called end is CVS from the weekend (from 10/24); I
don't know what the version on the calling end is. It is the calling
end that sends the Loop Detected message because my end is re-inviting
too quickly.
Luki
___
--Bandwidth
Hi all,
I'm newbie in asterisk (just first install)
I'm looking some ideas to send info about incoming call to another
process (my app)
I have this problem asterisk is actually installed syde by side with
the legacy pbx, one my program talk with the pbx and offers some
custom services on the
Receiving faxes do not generate a fax tone.
They will generate a modem tone when answered if that is usable/detectable.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- -
It worked.
Thanks for the 1.2 info.
Hopefully it hasn't created any unforeseen issues.
--
--
Steven
May you have the peace and freedom that come from abandoning all hope of
having a better past.
---- --- - - - -- - - -- - - - --- - -- -
- --- - - -- - -
It is registering with the sip provider. When I do a CLI sip show
registry, it shows as registered.
I really think my problem is about contexts. For some reason, I think
I got the flow of how a call comes in and goes out works.
Is this approximately right?:
Incoming SIP Calls:
PSTN - SIP
Yes, many people have had this problem.
Check the mailing list archives... I think the newest code has the fix.
Workaround for older versions is to Answer before Dial, but you may
still need the 'r' option to Dial as ringing may stop for the caller
after about 10 seconds.On 10/27/05, OTR Comm
We are running 1.0.9 STABLE on all of our machines. I am about try and
upgrade one machine to CVS HEAD as all this echo cancellation
improvements sound enticing. Can anyone recommend
a) A procedure to cleanly upgrade from STABLE to HEAD
b) A procedure to ensure I can back out and go back to
Hi,
I have TE406P (2nd gen card with echo cancellation on-board).
We still notice quite often echo on our PBX that is connected to one of
the spans on TE406P (with calls routers to PRI provider on another
span).
I've tried to experiment with the echo cancellation on asterisk.
I enabled echo
On Oct 27, 2005, at 9:52 PM, Eric Bishop wrote:
We are running 1.0.9 STABLE on all of our machines. I am about try
and upgrade one machine to CVS HEAD as all this echo cancellation
improvements sound enticing. Can anyone recommend
a) A procedure to cleanly upgrade from STABLE to HEAD
b)
Boris Bakchiev wrote:
I enabled echo cancellation in Zapata.conf to see if I can improve the
situation and users started reporting warping bubble (description I
got from one of the users) sound on calls from PABX-Asterisk-PRI (and
other way).
I was expecting that asterisk would disable its
I'm having this exact same issue. It's not critical, just annoying to
see that error every 60 seconds.
Dave Grey wrote:
I get these notices constantly:
Oct 22 01:26:35 NOTICE[383]: sched.c:296 ast_sched_del: Attempted to
delete nonexistent schedule entry 1!
They start with entry 1
I replaced a TE410P (1st Gen) with a TE411P (2nd gen with hardware echo
canceller) and the echo actually got much worse! Very disappointing!
On 10/28/05, Boris Bakchiev [EMAIL PROTECTED] wrote:
Hi,I have TE406P (2nd gen card with echo cancellation on-board).We still notice quite often echo on our
Hi Kevin,
Thanks for your reply.
That probably what it was. :)
Could echo cancellation on PBX conflict with VPM module and create the
warping babble sound that my users are reporting?
Do echocancelwhenbridged and echotraining do anything when VPM module is
used? Should I be using them?
Regards
Boris Bakchiev wrote:
Could echo cancellation on PBX conflict with VPM module and create the
warping babble sound that my users are reporting?
I don't think so, but anything is possible :-)
Do echocancelwhenbridged and echotraining do anything when VPM module is
used? Should I be using
I've had similar problems with no fix.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Boris Bakchiev
Sent: Thursday, October 27, 2005 10:02 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] Echo canceller on TE406
Thanks for the suggestion, but in my experience fax machines on ATAs can yield
unpredictable results, even at LAN speeds and uncompressed codecs.
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria, Australia. 3205
I have similar problems with performance degradation over time.
I'm about to post another message to the list (once I have some more
information). Stay tuned.
==
Rod Bacon
Empowered Communications
Ground Floor, 102 York St. South Melbourne
Victoria,
I've had similar problems with IAX2 and ticks. Jit buffer on and off don't
seem to change things much.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Colin Anderson
Sent: Thursday, October 27, 2005 7:13 PM
To: 'Asterisk Users Mailing List -
Does anyone have a full list of places Asterisk puts all config files
and binaries. I need this to be able to fully rollback if I have a
failed upgrade of Asterisk/Zaptel/LibPRI. So far I have:
/etc/zaptel.conf
/etc/asterisk/
/usr/sbin/safe_asterisk
/usr/sbin/asterisk
/usr/lib/asterisk/modules/
Agreed.
PaulH
- Original Message -
From: Rod Bacon [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Friday, October 28, 2005 1:12 PM
Subject: Re: [Asterisk-Users] Wanted to Swap! TDM400 FXO module(s) for FXS
Thanks
On Friday 28 October 2005 16:22, Eric Bishop wrote:
Does anyone have a full list of places Asterisk puts all config files and
binaries. I need this to be able to fully rollback if I have a failed
upgrade of Asterisk/Zaptel/LibPRI. So far I have:
/etc/zaptel.conf
/etc/asterisk/
I agree... I've got wy to many customers out there who are pissed
because they thought VOIP would be just as reliable (or even close) as POTS.
SKM
--Original Message-
-From: [EMAIL PROTECTED]
-[mailto:[EMAIL PROTECTED] On Behalf Of
-[EMAIL PROTECTED]
-Sent: Thursday, October 27,
If your 1.0.9 install is (on the /usr/src/asterisk tree) complete, you might
unpack the CVS source somewhere else other than /usr/src (maybe
/usr/local/src or /usr/src/cvs). Most importantly, PLAN AHEAD. It would
seem that the more Asterisk evolves, the less-tolerant it is natually
becoming
At 11:39 AM 10/27/2005, you wrote:
Um, well the easiest thing to do is:
1) stick the files on your webserver somewhere (e.g. www.mydomain.com/pcom)
2) Modify the top lines of each .php file so that the ip address is
that of your asterisk server, and the username and password match a
username
Hello ,
I have a old TDM04B card and newly bought TDM04B card. iam have a
asterisk,zaptel verison 1.0.7. now i need to add another card to my server.
then i bought a new TDM04B card.and changed the zapte.conf..but card is
not loading. when i type zttool . it shows only one board.
then i used
As I said it could exist, but I'm only guessing here that the posts
about ADSI over SIP channels are (again this just my guess) only for
the SIP channel to allow for the ADSI scripts to be downloaded into
the phones. Since it's like faxing that doesn't really work nicely
over SIP (or VoIP for that
Does anyone know how to redirect or pipe the processing of voicemail
sound file from
Asterisk to another application for what I want to do as described below?
Any input is welcome.
TIA
Kuni
-- Forwarded message --
From: Kuniyoshi Murata [EMAIL PROTECTED]
Date: Oct 22, 2005 8:54
I'm pretty new to Asterisk, and have the CVS head from a week ago installed,
so I guess the fix is hidden someplace not so obvious.
I don't really understand what you mean by *Answer before Dial,* could you
explain that? And I don't know how to use the *r* option for dialing on a
Cisco 7960.
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