your
parallel port? I'm using a USB-based device for digital IO and that
works great.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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the only one-way
audio problems I've experienced. Setting reinvite=no fixed everything
for me.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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a catch all pattern: exten = _.*,1,Nop(${EXTEN})
I had some random issues and I ended up defining another entry in
SIP.conf for the cisco by by IP address, e.g.
[172.31.2.7]
type=friend
host=172.31.2.7
insecure=very
context=cisco
qualify=2000
dtmfmode=inband
That works for me.
Norman Franke
. They are reasonably priced, come with a lifetime
warranty and free software updates. (Unlike with Cisco!)
Norman Franke
Answering Service for Directors, Inc.
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Has anyone got res_cepstral.so to work with Asterisk 1.4.21.1? It
appears to crash Asterisk on my box (kernel 2.6.26 gcc 4.1.2). Tech
supports doesn't seem to have any ideas.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
Server. I've had a few crashes with 0.82, I think, and I haven't
used 0.64.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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,
with this setup.
Norman Franke
Answering Service for Directors, Inc.
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asterisk
type fxs-loop-start
ds0-group 9 timeslots 10 type fxs-loop-start
ds0-group 10 timeslots 11 type fxs-loop-start
ds0-group 11 timeslots 12 type fxs-loop-start
The cisco then sends calls to Asterisk, and that part works great from
a PRI.
Norman Franke
Answering Service for Directors, Inc
.
Norman Franke
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works just
fine. If I call a local SIP soft client, everything works fine
(instead something via the Cisco.)
If I set canreinvite=no for the Cisco everything works. It seems
like the g option should disable canreinvite for that call, so why
the difference?
Norman Franke
Answering Service
is being transfered to
your call is being transfered
an (to go along with a)
supervisor
manager
incorporated
enter the four-digit extension of the person you are trying to reach
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
to at least 2.6.23.11. That
worked for me.
Norman Franke
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,
you'll get an error that it can't find a zap device (and playback
then works.)
Norman Franke
Answering Service for Directors, Inc.
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in late and didn't see they wanted Xen support, but I figure
others may find it helpful via google.
Norman Franke
Answering Service for Directors, Inc.
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asterisk
into the CentOS 5.x stuff, so they got the 2.6.18 kernel,
which I am
told works much better, and doesn't have the issues the old kernel
did.
I've also found that I can't get ztdummy working on anything less
than 2.6.23.11. Previous versions seem to have a broken RTC.
Norman Franke
Answering
On Mar 19, 2008, at 5:56 PM, [EMAIL PROTECTED]
wrote:
Anyone? Just a user?
I'm just a user, although I also develop things for internal use.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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office if they move and
just plug it in. Companies can also better monitor employees.
Norman Franke
Answering Service for Directors, Inc.
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, of
course. I've tried relaxdtmf=yes in sip.conf, as well. I added
dtmfmode=inband to the SIP peer for the Cisco. Nothing. Tones are
generated by phones connected to our PBX that we don't have a problem
with otherwise or even from a cell phone.
Norman Franke
Answering Service for Directors
a phone that can do SIP, as far as
I've seen. As soon as the iPhone software 2.0 is out, there will be
one for that.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
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that other hardware solutions. But, it's
cheaper and more flexible. You may not care about cheap and flexible,
and if not, maybe it's not what you want.
I've not tested products like CallWeaver or others. People claim some
of these are more reliable, but Asterisk seems more popular.
Norman
a few SIP-related patches from various bug reports
and things are much, much more stable.
1.4.17, which you mentioned, is also very buggy. 1.4.18 fixed many
issues.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
On Mar 18, 2008, at 7:40 AM, [EMAIL PROTECTED]
wrote:
We
I believe most of them will be in 1.4.19-rc3 (and in SVN), but I
applied patches to 1.4.19-rc2 from:
Patches from 11712 and 12098. Plus another one I reported as 12162.
Norman Franke
Answering Service for Directors, Inc.
www.myasd.com
On Mar 18, 2008, at 12:11 PM, [EMAIL PROTECTED]
wrote
and our custom solutions, it wouldn't help anyone, I'm sure.
Norman Franke
Answering Service for Directors, Inc.
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On Feb 26, 2008, at 4:13 PM, [EMAIL PROTECTED]
wrote:
On Tue, Feb 26, 2008 at 3:10 PM, Matt [EMAIL PROTECTED] wrote:
I've had it with Dell server garbage.They seem to change RAID
controllers as much as I change socks, and then the controllers
don't work
with Linux, unless you load a
.)
Norman Franke
ASD, Inc.
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agents to one side.)
How have people handled this? I'm experimenting with a noise gate
that will lower the volume when the agent isn't talking, but that
won't help when the agent is talking.
Norman Franke
ASD, Inc.
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and T.38 FAX processing in a single SIP
entity.
Apparently, indeed. I've been unable to get it to send faxes via a
Cisco gateway. (Receive is OK.) The other side always reports errors,
so it may or may not work for you.
Norman Franke
ASD, Inc
Is anyone aware of a service where we can lookup phone numbers to
determine a name and/or name + address available in bulk?
We want to look up every number called to our call center, so it will
be tens of thousands per day. Services that charge 3 to 5 cents per
lookup will get way too
eventually got this working, but not with Asterisk. It's for our
legacy voice mail system.
-Norman Franke
ASD, Inc.
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the polycom's to auto provision wasn't very
straight forward. I do provision some the linksys PAP2Ts via HTTP and
that works quite well, so I suspect the SPA's to be relatively similar.
Norman Franke
ASD, Inc.
www.myasd.com
On Sep 24, 2007, at 7:06 AM, [EMAIL PROTECTED]
wrote:
Linksys
Noah,
Thanks for the input. I'm thinking the problem with the stop
gracefully is that it would confuse the auto fail-over appliance, in
that it would either detect the server is dead and hard switch the
T1s or keep sending calls there which Asterisk would reject.
I'm thinking a better
On Jul 19, 2007, at 5:16 PM, [EMAIL PROTECTED]
wrote:
On 7/18/07, Norman Franke [EMAIL PROTECTED] wrote:
I've been evaluating Asterisk for a while, and things seem to be
going very well. The issue of redundancy and automatic fail-over is
now on my mind. I searched the archives and googled
I've been evaluating Asterisk for a while, and things seem to be
going very well. The issue of redundancy and automatic fail-over is
now on my mind. I searched the archives and googled for solutions,
but didn't really come up with much.
We'll be using queues (modified), which precludes some
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