and working fine with others. MOH and other audio playback
features seem to work fine. What's different about app_dictate?
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David Josephson
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I wonder if anyone else is having these problems. We are running
Asterisk 1.2.17, with an assortment of SIP users and peers. This is
running on an 600 MHz P3 with CentOS 4.4, and worked properly in
Asterisk 1.2.15. Nothing else running on the server except the usual
support stuff like sshd, a
Luki writes about choppy audio with LiveVOIP. We have an almost
identical situation except that we were switched from the San Diego
gateway to the Van Nuys gateway. Some improvement but still not usable
for real customers. I have an open trouble ticket with them and no
progress. Doesn't matter
bit linear (raw) which becomes .wav if you add file headers to it.
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It is my understanding that TDM is circuit switched and VoIP is packet
switched. It would seem to me that at some point in a TDM-VoIP gateway,
a change from circuit switching to packet switching is happening, and
vice versa depending on the direction of the signals. I was just
wondering if
There has been improvement in the quality of LiveVoip connections. Still
some packet loss and resultant choppy audio, a little worse than with
Vonage or Broadvoice. As noted in several posts over the past months,
they still don't handle indication of ringing on an IAX channel if the
caller has
In some tonelists, as used in Playtones or indications.conf, I've seen a
notation to set levels, for instance [EMAIL PROTECTED] The -10 doesn't
seem to do anything. Is there a patch that will enable setting levels in
a tonelist?
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Rich Adamson responded to an earlier reply (not from me)
Eric, those links have nothing to do with his stated problem. The
problem is 105v AC on the pstn line when on-hook and no ringing.
No, he says the issue is about ringing and strange voltages on his
Digium TDM400 FXS ports, not the PSTN
I see in the list archives that this problem came up before, but there
was no fix for it. Any clues now?
Inbound calls from LiveVoip work (I am assuming they will soon fix their
packet loss issues at the San Diego pop) except for one thing -- no
ringback when the called extension is ringing.
what's possible within them? Let us know
how to use AMP to pass stuff to * unless you are willing and able to
track all the apps within it at once.
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Trying to get some straight info from the VOIP providers is difficult.
Say there's a small Asterisk switch and it's registered with Broadvoice
or LiveVOIP or someone. There are a couple of people using the switch,
one is on an outgoing call with the VOIP provider. What happens when
someone
On Fri, 2005-03-25 at 08:00 -0800, Mark W Wood wrote:
I have searched both the wiki and googled looking for a solution to
a square key configuration. I need to have C.O. lines to appear on
the buttons to facilitate a small office. All of the users can see
each other and calls are put on hold
Can I program a specific C.O. line directly to a button?
Adopting the Critchfield style for a moment, no, *you* probably can't.
But, depending on one's expertise with the hardware and Asterisk, it can
certainly be done. I am by no means an Asterisk expert but have about 35
years of
Greetings list,
We are having a puzzle with * (asteriskathome 0.5) and SIP phones
(SPA2000 ATA's). If callwaiting is enabled, everything (including call
waiting) is normal. If callwaiting is turned off, the phone will not
accept incoming calls and the call goes straight to whatever is
or percent or some arbitrary number. My C is very rusty, but in
chan_zap.c the number seems to be converted to a linear gain number
using a formula for voltage gain rather than power gain. Has anyone
calibrated this?
Thanks
David Josephson
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There is a mention that the current Sangoma T1 cards (A10[1,2,4]) work
with * using their WANPIPE drivers. Has anyone used any older Sangoma
cards that also support WANPIPE ?
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On the various technical issues raised here (OK, we posted the rules, we
won't discuss the legality anymore) I think there is only one main
obstacle to using FRS radios for extensions on *. They are simplex
(push-to-talk, release-to-listen). The protocol for dealing with
Daniel Nystrom wrote
It seems like the Radio discussions is closing in on something I was
interested in.
How about controlling 30 2-way radios via E1 and 30-channel Mux
(channel bank?) with EM signalling?
I think the Mux uses CAS and each channel has Audio out, PTT, Audio
IN, Busy. 6-wire
Rich Adamson writes
GMRS, FRS and MURS radios may not be interconnected with the PSTN (47
CFR 95.141). There has been a lot of talk from lobbyists to clarify this
rule, but as it stands you could conceivably connect a *private* network
to GMRS or MURS radios (you can't make any plugins or
You don't need to reinvent anything to tie radios to *. Ham systems like
IRLP, Echolink, eqso etc all have fairly tight controls to keep from
being abused (although with a little Linux knowledge, the IRLP package
can easily be used to set up your own network using their protocol). Jim
Dixon
Pete VK2YX writes
I've been involved with IRLP for about 5 years and am one of the original
install team. I've gone through the emmotions of allowing other networks
connect to IRLP and I know its caused some lots of headache.
As far as a closed network goes, yes there is LOTS of passion to keep
John Novack writes
Dare I suggest that a MUCH better job of documenting would go a long way
towards eliminating the problems you mention?
Now I realize that programmers are much more interested in writing code
than documentation, as well as moving on to the next hot feature than
making sure
Is there a configuration difference for clone X100P cards versus
compatible? I have a similar problem to what David Shaw posted earlier
today. 0.5 installed OK, but mine just with one X100P clone. Default
config files, edited zapata.conf per the FAQs so it includes the line
channel = 1
without
into AMP and configure some place for
incomming calls to go?
--- David Josephson [EMAIL PROTECTED] wrote:
Is there a configuration difference for clone X100P
cards versus
compatible? I have a similar problem to what David
Shaw posted earlier
Rob at draughon.org writes
I recently obtained a Western Electric multi-line phone and am
seeking help with getting this beast working with *.
The interesting stuff in my * implementation consists of a T100P
card, a TDM400P card, and an Adtran TA750 channel bank with three
Steve Blair writes
I can redirect and relay calls to numerous destinations via
SER but because the Octel needs an SMDI interface for mailbox
identification I am stuck, none of the solutions thus far support
SMDI-SIP munging.
I just started thinking about the possibility of using Asterisk
with a
Has anyone found an inexpensive EM trunk card that will play with *?
Looking for an interface to a legacy electromechanical PBX that's able
to pass answer supervision. Docs on the X100P card would be helpful, we
could probably pull EM out of that. Any ideas?
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