Let me start by saying no, this is not the normal stupid echo question :)
I currently have a SIP device and an X101P, and have had the usual echo
issues and have played around with the various solutions, none of which
are quite perfect (understandably).. however, my girlfriend is a little
more
On Fri, 2 Jan 2004, Steven Critchfield wrote:
On Fri, 2004-01-02 at 12:34, Gary Gapinski wrote:
Does anyone have recommendations for (or against) mini-ITX platforms to
be used with Wildcard X100P and TDM400P cards?
I am considering the use of systems using VIA EPIA CL and Epia M as
Tilghman,
What happens if someone needs the new signalling routines *and* working
fax detection? I'm personally not in this boat, but it's only a matter of
time before someone is.
Is this a temporary fix? If not, this should be documented somewhere as it
seems to be a problem for enough
This was a problem that got introduced some time around (I believe)
12/15/2003 -- It might have been fixed by now, you should do a fresh cvs
checkout and try again. If not, do a cvs checkout from around 12/08/2003,
that version worked for me.
(btw, the issue is some updates that were applied to
I know this issue has been covered with at least 2 different patches, and
probably a dozen different discussions, however I'm a bit unclear as to what
my options are.
I have a DSL line coming in with 8 IP addresses going to an OpenBSD firewall
doing 1:1 NAT for machines behind the firewall. My
At one point I was tasked with investigating some sort of data link between
New York City and Cebu, Philippines. We were interested in doing voice over
this link, and quickly learned that voice over satellite links is *not* very
usable. We ended up with an international fractional T1 (and I was
All,
I upgraded my asterisk setup from CVS on or about 12/15. Suddenly, *all*
of my incoming calls are coming up as FAXes. I had to disable my fax
extension because every call to my POTS line was getting redirected to my
FAX machine. After removing the FAX extension, if I call my POTS
FYI: Doing an rm -rf zaptel asterisk in /usr/src, then doing a checkout from
12/08/2003, I no longer have this problem.. so it's something with the newer
code?
Thanks!
PAt
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Patrick
Cantwell
Sent: Wednesday
: Wednesday, December 17, 2003 4:25 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ALL incoming Zap channel calls are getting
picked up as FAX calls!
On Wednesday 17 December 2003 14:43, Patrick Cantwell wrote:
I upgraded my asterisk setup from CVS on or about 12/15.
Suddenly, *all
PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Tilghman Lesher
Sent: 17 December 2003 21:25
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] ALL incoming Zap channel calls are getting
picked up as FAX calls!
On Wednesday 17 December 2003 14:43, Patrick Cantwell wrote:
I upgraded my asterisk
That's because when you do a make, it compiles the binaries with today's
date.
you could be running a version from 5 months ago, but it'd have today's date
when you compile :)
-Pat
- Original Message -
From: Kevin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, December 18, 2003
Just a thought: Out of everyone having problems, how many are connecting
through NAT to their asterisk server? (client - nat - ... - asterisk).
NAT firewalls time ports out after a certain amount of inactivity. I was
having the same problem with my SIP analog adapter. I could fire it up and
FAX over VoIP works fine, but it is bandwidth-hungry. I have no problems
here picking up a FAX off the PSTN and routing it via VoIP to my SPA2000,
and out the second port to a box running hylafax (no different than an
analog FAX machine). As long as you use a high bitrate codec and can afford
to
Of Patrick
Cantwell
Sent: Sunday, December 14, 2003 9:26 AM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] voip-info.org DNS seems broken
Ollie,
It's been my experience that you can never have too many mirrors and/or at
least secondary DNS.
If you'd like me to do either for you, I'd
Yes, Most likely.
Also, be sure your hardware is up to spec. I'm running into the same issue,
however I'm running on old hardware which has already been covered on the
list :)
Shut down X, make sure your hardware is beefy enough, and try it again..
-Pat
- Original Message -
From:
All,
If you currently own a Sipura SPA2000, avoid going to the sipura website
and upgrading the firmware. I upgraded my SPA2k a couple of days ago from
1.0.9 (what it came with) to 1.0.18 off the site, and I am having issues
with my SPA rebooting itself every 3-10 minutes for no apparent
Why would you need one of those? Those are designed for plugging a voip
line into a POTS line or such so that you can dial a number, hit the
converter, and pass your call out the VoIP line. All you need to do is plug
the ata186 into your X100P. No extra hardware required.
-Pat
-Original
Are the Aastra PTXXX phones ADSI? How compatable are they with asterisk? I
see some PT450s on eBay for reasonable prices, and I just may be tempted to
pick one up for fun. Is it worthwhile?
Thanks!
Pat
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of
I actually have a small echo problem on my x100p that I haven't gotten
around to fixing yet, however my FAX works fine, even at 14.4k.
Unless you have a huge echo problem, it should be negligable.
-Pat
- Original Message -
From: Andrew Joakimsen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Are you logging in on the right page?
Voicepulse offers both 'individual' accounts and IAX accounts. If you
signed up for an IAX account you have to login at
http://connect.voicepulse.com (or scroll down to the very bottom of the
page at www.voicepulse.com and look for the text Are you a VoIP
Hi..
I've got an asterisk setup with
an X100P card installed.. I'm noticing that upon hangup, it takes a good 3 to 5
seconds before asterisk realizes the line has been hung up and drops the call..
this causes my SIP phone to continue ringing, and occassional phantom voice mail
messages to
OK everybody, I have solved my problem!
The issue lies in how you handle the incoming call. I actually stumbled
across this while trying to find a better way of doing fax autodetection.
The trick is you *must* use Answer to pick up the call. The *WRONG* way of
doing things is:
exten =
All,
Here's a cool one.. I was attempting to call a retarded conferencing
service, and was having problems with it picking up my DTMF.. after trying
all the settings my Sipura SPA2000 offers, I found inband actually works..
unfortunately, I can't get anything else to pick up my inband
Have you recompiled all of the other libraries asterisk links in as well?
I'm pretty sure when you do a glibc rebuild/upgrade, other libraries must
be rebuilt/relinked as well as they depend on parts of glibc.
Pat
On Sat, 29 Nov 2003 [EMAIL PROTECTED] wrote:
I've been running an asterisk
blue, so if you have it, please let me know.
Also, I've been watching the list for a couple of weeks now, and it seems
the grandstream phone plays very well with Asterisk. Any particular
gotchyas I should look for?
Thank You,
Patrick Cantwell
Ok,
I've googled for 15+ minutes, and have yet to find a usable answer, so I'm
going to annoy everyone and ask here.
I have, in my posession, a creative VoIP blaster. I have installed the
fobbit LKM and I can see the device. Can I use it with asterisk in any
meaningful way, shape, or
I am having the same problem.
I'm running it from the command line out of c:\iaxcomm, and it loads, but
only shows up in task manager.
This is a WinXP Pro box.
Thanks,
Pat
--
C:\cd iaxcomm
C:\iaxcommdir
Volume in drive C has no label.
Volume Serial Number is A8A6-3CB0
Directory of
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