a reload command. At least back then it
was still too unstable for production. But I do look forward to a more
completed realtimehopefully you'll be able to name tables whatever
you want then too! Like voicemessages
-mark
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.
Disclaimer: It is possible that I just have a defective SPA-2100...but
what are the odds? Besides, my Azacall has supported T.38 for almost a
year now and the SPA-2100 still doesn't have it (at least, not in the
current firmware). Not that any of that matters anyhow...
-mark
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been fixed.
Seems like a very Microsoft-like attitude: If it don't work, don't
bother to
fix it, just go back to what did work.
Which res_config_* where you using? What did the debug say? What did
the
backtrace say?
-Matthew
Mark Eissler wrote:
On Mar 11, 2005, at 2:38 PM, Cirelle Internet
cordless phone (that
is, multiple handsets for less than $400) then you will need an
adapter.
For now, I've chosen to ride things out with my old Siemens Gigaset and
have ordered a Sipura 2100 as the only immediate upgrade (not to
impressed with my current adapter).
-mark
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using trunk mode.
-mark
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On Feb 25, 2005, at 11:41 AM, Steve Underwood wrote:
Mark,
In the time it took to write all that you could probably have read up
enough about T.38 to realise you were talking complete rubbish :-)
Gee thanks Steve. And your insight has been absolutely beneficial as
well.
-mark
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an analog adapter that supports T.38. If you want to hookup a
fax machine to a port on a channel bank or a zap card then you're going
to be out of luck unless the zaptel driver supports T.38.
-mark
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for their 7410 all in one machine and, more importantly,
they include support for changing transmission speeds. Way to go HP!
-mark
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doesn't really have a fax problem per se. Fax was
developed for analog lines. What the world needs is fax machines that
are completely IP-based and only fall back to using a gateway when
connecting to POTS-based fax machines.
-mark
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the e911 system by clogging
their lines that goes beyond illegal. Find another get rich quick
scheme.
-Chuji
E, let's keep in mind that the original poster may not be located
in North America. Either way, I'm sure the OP got the message that they
won't find help here.
-mark
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providers. No need
for TDM cards. Your system will work fine. I've done this on an old
Pentium Pro with only 128MBs for a small system. You only need bigger
hardware if you're going to add many more users. Two SIP phones is
nothing.
-mark
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properly while others must suffer at the
hands of an ISP that does a really crummy job.
-mark
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for me, does it mean all wireless device?
I haven't dug into the source looking to fix this problem so, no, I
haven't tried enabling that option. Perhaps someone on the list knows
more about this option? I'll certainly try it out as well.
-mark
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posts
in long response anyhow! ;-)
-mark
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Vonage account I'll just hang on to the ATA [he casually comments as
the collection of VOIP adapters steadily grows in the basement...].
-mark
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realy slow then it seems to work more reliably.
BTW, Steve, if you're still reading, what is the RADIO_RELAX option
intended to be for in dsp.c?
-mark
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in dsp.c)
I remark an big improvement in the detection of the dtmf over GSM.
have you ever test this option?
RADIO is obscur for me, does it mean all wireless device?
Florian
-mark
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On Feb 14, 2005, at 1:25 PM, Pedro wrote:
Is it just a bad implementation of g729 compression with the Sipura
product line?
That would be my guess.
-mark
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don't get it. My assumption is that by background you mean the
applicant turns out to be a VOIP provider as opposed to say a lot of
people on this list that would be interested in buying one of these
devices but can't because they're really just end users.
-mark
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will be released for the Linksys-branded stuff or
what level of support there will be.
-mark
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?
is there any way to prevent this
I suffer from this problem all of the time but I'm configured for
out-of-band dtmf on the asterisk side thanks to IAX2 trunking. I think
the problem is GSM or maybe cell phones in general. I know this has
been discussed before.
-mark
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with
Broadvox almost exactly one year earlier.
-mark
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agree. I think the technology needs to be rebuilt for IP, but I
don't think the concept is going to go away anytime soon.
-mark
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initiated within *.
Interesting. I wasn't aware of that. I'm more inclined to blame my CPE
at the moment. Will probably switch to a Sipura 2100 soon.
-mark
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be purchased retail anymore. Then again,
you're probably better off with a Sipura-branded unit anyhow.
-mark
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for faxing are all related to outbound faxing (using the same
Voicepulse setup or Sixtel iax.cc). Not sure why outbound is giving me
problems when inbound isn't giving me any. shrug Of course I need to
fax outbound more often than I need to receive inbound!
-mark
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,
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my opinion on How things should work!. ;-)
-mark
On Feb 11, 2005, at 10:06 AM, Chris Wade wrote:
It basically works out that CVS *is always a moving target*, tarballs
are the only things that don't change. Make sense?
-Chris
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version? I don't
want
my system to seg fault as the cvs list would indicate it could.
So there are known issues with the latest CVS Stable? What is the best
known
version of Asterisk to date? 1.03? 1.05? I'm not interested in new
features,
just stability and quality.
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failure :-)
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with:
exten = _2.,1,Dial(IAX2/user:[EMAIL PROTECTED]/${EXTEN:1})
Is trunking dependent upon the use of 'register'? Or, dependent on the
above exten=_2., referencing a context (instead of the IP directly)?
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for the
dial
command, but maybe this is a message that we should start an official
beta
release period before a release so that people can test pre-releases
even
for just a week to report problems before it is unleashed upon the
world as
an official release
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Mixtur
over the phone
or IM. Not very promising.
All I want is a toll free DID that works on * and isn't too expensive.
Any suggestions for a provider? I don't even care if it can be ported
away!
On Thu, 3 Feb 2005 10:12:02 -0500, Mark Eissler [EMAIL PROTECTED]
wrote:
Based on the support and management
that noted this might be the case and if
the documentation is lacking...well, that's a problem.
-mark
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just fine, but
while in the call, after a few seconds, the pbx starts the simple
switch once more and keeps ringing the voip extensions log as
follows:
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Mixtur Interactive, Inc. [EMAIL PROTECTED] http://www.mixtur.com
report for this in Mantis. Is anyone else experiencing
these problems?: 1) incorrect callerid; 2) SIP extensions not
disconnecting?
-mark
On Feb 3, 2005, at 6:05 PM, Mark Eissler wrote:
I seem to be experiencing the same problem (I'm using IAX trunks). My
SIP phone keeps ringing after a call has
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iax.conf file, is that a trunked connection to my DiD
provider?
Thanks!
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Yes, you are correct. Instead of separating the RTP streams for each
call, trunking combines them in order to save on overhead.
-mark
On Jan 31, 2005, at 9:38 AM, Andrew Kohlsmith wrote:
On January 31, 2005 08:57 am, Mark Eissler wrote:
You must set trunk=yes in the context of the relevant
.
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. We are looking at this version as well as 1.0.3
for some other issues now but, Asterisk is not our only platform.
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out of
curiosity
and interesting enough, when I called into my IVR w/ my cell phone, it
recognized 1234 and whatever other digits I entered. So inbound DTMF
worked
using ULaw, however I never tried outbound. Could have been a fluke
though.
Give it a shot.
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was with asterisk users, I have not upgrade or install any patches and
all is fine now.
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The order matters in asterisk so if you want GSM to take priority over
G729,
simply put that ahead of the G729... so your settings should be:
Allow=all
Allow=gsm
Allow=g729
Allow=ulaw
Allow=alaw
Try that and see if it works.
Regards,
Mohammed Salim
EZZI Telecom, Inc.
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Asterisk 1.0.3
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to read certain digits(tones).
When had it configured to use SIP for incoming calls, it never failed.
- Original Message - From: Mark Eissler [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com; Brian Dingman
[EMAIL PROTECTED
.
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doesn't work at all.
So what does this say about * and DTMF recognition over IAX? Or the
service providers?
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Houston TX to Cambridge UK. Dead
reliable.
I'm dropping my IAXTel registration when next I get around to such
things.
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because I've found two problems with them so
far while using IAX2:
1) The fax problem mentioned above.
2) Inbound DTMF is quite broken. (They are working on a fix and said it
would be at least 30 days...but then in December they said it would
take 2 weeks...). What a drag.
-mark
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!
http://www.YourOwnISP.com
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connects you a hear a brief CNG tone and then the line goes silent.
Maybe they have T.38 turned on.
-mark
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