Previously posted to the Users list (FYI).
We have a system running Asterisk-1.4.40. Queue calls are
distributed using rrmemory with a 20 second timeout. What we are
seeing is; when a call in the queue will call the first agent for 20
seconds, and subsequent attempts will call agents for random
We have a system running Asterisk-1.4.40. Queue calls are
distributed using rrmemory with a 20 second timeout. What we are
seeing is; when a call in the queue will call the first agent for 20
seconds, and subsequent attempts will call agents for random periods
of time (as little as one second),
We have a system running Asterisk-1.4.40. Queue calls are
distributed using rrmemory with a 20 second timeout. What we are
seeing is; when a call in the queue will call the first agent for 20
seconds, and subsequent attempts will call agents for random periods
of time (as little as one second),
Asterisk 1.4.32 (Also 1.4.26, 1.4.33)
Broadvox ITSP (xxx.xxx.xxx.xxx)
Linksys 2102(yyy.yyy.yyy.yyy)
Both peers :
canreinvite=yes
t38pt_udptl = yes
I'm having some trouble getting a T.38 fax call established with
Broadvox. During negotiation, Asterisk sends a SIP re-invite (T38
switchover)
It seems that asterisk-addons and one or more of Digium's licensed
modules such as res_fax_digium have a conflict that doesn't seem to
be documented anywhere I can find.
In a nutshell, asterisk14-addons-core has a fake provide for
asterisk-gplonly :
#
# core subpackage
#
%package core
On 4/1/2010 1:52 PM, Kevin P. Fleming wrote:
Chris Miller wrote:
A comment in the spec file would have been nice... Does anyone know
if this a real technical issue, or simply a licensing conflict
between GPL and Digium?
It is not a technical issue; it is an issue because some of the modules
Darrin Henshaw wrote:
add resetinterval=never in your zaptel.conf, or chan_dahdi.conf
depending on what you are running. zaptel or dahdi.
Can someone confirm when the default was changed from never to
3600 seconds? According to the voip-info wiki, never has always
been the default. I would tend
We've inherited a pair of mostly identical PBX systems, each with a
TDM400P Rev I boards and 4 FXO modules. The production system is
running Asterisk-Now with 1.4.9, and despite some other issues, it
is able to answer inbound calls just fine. The replacement system is
currently running
I've got a system with a TE412P installed under Fedora Core 6 and I continue to
see this message in the logs. The card most certainly does have an EC module
installed. The system is suffering from echo problems, and I suspect this is no
coincidence... I've double checked to ensure the module
I'm chasing down some issues at a call center. Today I received a complaint
that audio file playback
ceased after they upgraded the system from FC4 to FC6, Asterisk 1.2.14 to
1.2.17. Zaptel is at
1.2.16. The system in question takes inbound calls via IAX2 and has a TE410P
with a couple of
Eric ManxPower Wieling wrote:
Chris Miller wrote:
I would tend to agree, but the context that holds these number is an
inbound context which includes additional logic that would fail
normal calls. Yes, I can add the DIDs to the outbound context, but
the point here is not to have a bloated
Asterisk SVN-branch-1.2-r48484
I get a SIP Response 482 (loop detected) back from my SIP provider
whenever I dial from/to DIDs on the same server. The call is assumed
from an unknown peer, then gets routed to
Local/DID@from-sip-external which fails. No SIP headers/messages are
generated
Paul Hales wrote:
But the general thought is that if you build your contexts right, your
internal SIP users should hit those numbers as part of their dialplan.
I would tend to agree, but the context that holds these number is an
inbound context which includes additional logic that would
I'm seeing channel.c: Nobody there, continuing... in the asterisk
full.log. This error is repeated 20+ times per second when it occurs. I
thought this problem was specific to one PBX that performs call
recording on all the call queues, but after disabling all call
recording, the error
Upon investigating a call quality complaint with a conference room, I
discovered this error repeated several times in the log. Looking at the
source, frametype 5 is An empty, useless frame. Does this indicate an
actual problem?
app_meetme.c: Got unrecognized frame on channel
Local/[EMAIL
I'm seeing this error more and more in the full log. When the error
occurs it prints to the log 20+ times per second. At first I thought it
was specific to one PBX that performs a fair amount of call recording,
but after disabling call recording in all the call queues, the error
remains.
, but it sounds like they may be
related to the Trixbox compile of the latest Asterisk.
Regards,
Chris
Chris Miller
President - Rocket Scientist
ScratchSpace Inc.
(831) 621-7928
http://www.scratchspace.com
___
--Bandwidth and Colocation provided
I'm chasing down a pop/click type of disturbance on a PBX system.
Strangely, the disturbance is only heard by the outside caller, the
internal recipient hears the caller crystal clear. This seems to have
crept up when upgrading the zaptel driver to the 1.2 series while
running 1.0.10. I went
config file sip.cfg (i.e. via ftp server).
tone.dtmf.rfc2833Control=0
It appears that although Asterisk recognizes and uses inband dtmf
internally, rfc2833 is used on the external channel. I noticed this
behavior when remote IVR systems weren't acknowledging dtmf.
Regards,
Chris
Chris
Mojo with Horan Company, LLC wrote:
The recent suggestion on the list was to not use 1.0.9 zaptel
You mean the driver, or the version of fxotune? fxotune has been removed
from the prior versions of the zaptel driver, it's only included in 1.2
now. As for the driver, is anyone using the 1.2
I've got a cell phone setup as an extension in a queue. On occasion the
cell phone will drop the call due to loss of, or bad, signal. Is there a
clean way in the dial plan to reintroduce a call back into the queue
when the call is dropped on the extension side? I realize this would
occur
)
Found a Wildcard TDM: Wildcard TDM400P REV I (4 modules)
Registered tone zone 0 (United States / North America)
Regards,
Chris
Chris Miller
President - Rocket Scientist
ScratchSpace Inc.
(831) 621-7928
http://www.scratchspace.com
___
--Bandwidth
I'm looking for advise on troubleshooting QOS problems. After much
searching and reading online (Google, Voip-Info Wiki, etc.) I don't feel
any closer to finding the right tools to solve my problem. Any info you
would like to share would be much appreciated, and I'm sure the thread
will
I'm seeing a number of these logged in full while my * system is idle,
but I haven't found a good description of what they mean. Can someone
oblige? I have a single SIP phone registered and an IAX trunk.
Chris
Sep 19 22:13:44 DEBUG[18720]: (Provisional) Stopping retransmission (but
, but support response has been fast via email and the system
has been rock solid in my testing so far. I've seen other folks give
them good marks as well.
Regards,
Chris
Chris Miller
President - Rocket Scientist
ScratchSpace Inc.
http://www.scratchspace.com
Hello,
I've got an Asterisk system with 3 SIP trunks configured. Each SIP
trunk is actually a 4 port Mediatrix PSTN gateway. The current outbound
call routing (via AMP 1.10.007a) uses the 3 trunks in descending order,
all set with max channels to 4. Unfortunately, when the first trunk
I'm running * on an AMD 64 system with FC3 x86_64, everything works fine
so far. Programs can be rewritten to take advantage of the the 64 bit
architecture and the extra computing power. Having seen that many high
end systems are using 32 bit Xeon based systems for call capacity, I'm
wondering
Dave Green wrote:
I've downloaded the latest CVS as of yesterday. Zaptel and libpri
compile and link OK but after issuing the make asterisk command I get
the following:
/usr/bin/ld: cannot find -lidn
collect2: ld returned 1 exit status
make[1]: *** [app_curl.so] Error 1
make[1]: leaving
Christopher L. Wade wrote:
Andrei (MPI) wrote:
Brian West wrote:
Just an FYI to all out there that are upgrading after this weekend's
run of
CVS updates that are in now... MAKE SURE YOU DO make clean. If
you don't
and asterisk acts funny this is why. Anytime any struct like
ast_channel
Lane wrote:
Hi, again.
I've spent a week trying to get asterisk to work on FreeBSD unix, with some
success. Everything works until I plug the box into the TELCO line and then
the line goes off-hook and stays that way.
I wasn't able to get the zaptel stuff working under 5.3, but that has
more to
I had the same problem and I see that this was addressed in udev-043 :
http://lwn.net/Articles/111858/?format=printable
(search for zaptel)
FC3 has udev-039-8.FC3 installed by default. If you run up2date, an
update to udev-039-10.FC3.6 is available that fixes this problem. Also
the typical zaptel
Bruno Hertz wrote:
On Sun, 2004-12-19 at 13:11 +1100, David Uzzell wrote:
I have * running on Mandrake 10.1 and I to had similar problems in the
begging but as soon as I had ztdummy configured correctly everything
seemed to just fall into place and work with IAX and *, not that I have
got a
32 matches
Mail list logo
