Hi,
I'm trying to set up a basic FXO SIP gateway. That is, I want calls
from my SIP phone to simply be dumped onto the POTS line. My (entire)
extensions.conf is:
[from-sip]
exten = _9NXX,1,Dial(ZAP/1/${EXTEN})
and my zaptel.conf is:
fxsks=1
loadzone=us
Does anyone know which signalling protocol works when connecting the
output of a NT8B90AL Norstar ATA2 analog terminal adapter to an FXO
card? (fxsks, fxsgs, fxsls?)
I'm trying to add an Asterisk branch to my Norstar PBX as outlined at
http://www.voip-info.org/wiki-Asterisk+Nortel.
- Mike
I'm not surprised. Asterisk is a PBX, not a key system or a hybrid
system. The kind of functionality that is being described here is one or
both of those 'other' beasts. Now I'm not saying that this wouldn't be
nice, or even a long term requirement if you really want to open the
entire SME
Hi all,
I'm curious if anyone is using something like the Iaxcomm softphone
(http://iaxclient.sourceforge.net/iaxcomm) with an IPP200 handset (or
something similar) in an office environment.
Is this just a neat (and cheap) parlour trick, or do people find it
usable for real work?
- Mike
Can someone point me to an agi script that will read back the asterisk
uptime, if such a thing exists?
- Mike
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What we tend to do with people who require out-of-area calling ability is
grab a toll free DID from a bit of a bigger or more stable provider. Here in
Ontario, Canada, we've had great success with Unlimitel for providing toll
free DIDs.
I have run across that name before as well - anyone
Rory Campbell-Lange wrote:
I'm trying to install wanpipe on my new 2.6.21-2-amd64 core 2 duo
machine (Debian's amd64 works with EMT64 too) to run Asterisk. I'm
getting compilation errors when trying to install the wanpipe utilities.
Sangoma says that 2.6.21/22 is not supported yet, just
In order to get subscriptions working and the Snom 360 lights turns
on, I have set everything just like all the pages in the net explain.
So, I get subsciption working. I can list subscription on the
asterisk and if I use the SIP trace function built in at the SNOM nad
see NOTIFY
Hi,
Is it possible to use the hint priority to allow call parking slots to
be monitored on (for example) Snom indicator lamps? How do you refer to
the slots (i.e., what is the channel) in the hint?
- Mike
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Hi,
I updated the firmware in my Snom 360 from 4.3 to version 5.3.6 (and
then back to 5.2), but I'm having a weird problem now:
After rebooting, I can make one outgoing call successfully. Subsequent
calls don't work - the 360 just seems to do nothing after pressing the
OK button (but I can
After rebooting, I can make one outgoing call successfully. Subsequent
calls don't work - the 360 just seems to do nothing after pressing the
OK button (but I can cancel the call, the phone isn't frozen). The
Asterisk console shows the first call going through, but nothing
appears for the
I'd like to be able to use my Snom 360 LEDs to view the status of
parking slots, so I'm trying to install the metermaid patch
(http://bugs.digium.com/view.php?id=5779). Can someone help an svn
newbie figure out how to install this patch? I've done the following:
svn checkout
nat=yes
qualify=yes
That works, but it works better if you use a NAT/firewall box that can
do VOIP transformations automatically. The Sonicwall TZ170 can do
this. It rewrites the packets auto-magically so things just work. The
above parameters can be set to no then.
It seems to work more
Hi,
Can someone recommend an IAX provider for US DIDs who will:
1) Accept Canadian credit cards (rules out Junction Networks!)
2) Can do local number porting (LNP)
3) Have great audio quality
I tried Teliax, but the IAX audio quality was terrible - pops and clicks
galore! The Teliax SIP
I find that DTMF does not work reliably if jitterbuffer=on for certain
IAX providers. For instance, DTMF tones are missed entirely about half
the time when I dial into an exgn.net account. However, it always works
fine for an unlimitel.ca account.
Someone else has seen this too:
Rich Adamson wrote:
I find that DTMF does not work reliably if jitterbuffer=on for certain
Can anyone suggest a workaround (other than jitterbuffer=off)?
Might try turning off trunking (assuming you have it turned on) and
test again. Seems a couple of parameters interact and probably has
Martin Joseph wrote:
snip
3) Have great audio quality
This is somewhat a meaningless question, as the route from you to the
call terminating service can make or break the quality.
Sure, but some carriers have problems inside their own networks. I can
optimize the routing to the provider
Hi all,
The metermaid patch allows you to use the programmable buttons and
LEDs on phones (like the Snom 2xx or 3xx) to view the status of parking
slots and transfer to them. This should be really useful for
small-office environments.
Anyway, the patch seems to work with Snom phones (and
I use Snoms. I know there's the feature. I just don't know how to use
it, and there's so little documentation on the web.. Anyway, with see
I meant that the secretary's phone would have one of the function keys
on whenever the chef is on the phone (also when he picks it up,
right before
With Asterisk 1.4 I could use commands like:
/usr/sbin/asterisk -rx sip notify reboot-snom mjc_home
to reboot a snom phone. Now, with 1.6, when I try that, I get:
Unable to find notify type 'reboot-snom'
Command 'sip notify reboot-snom mjc_home' failed.
Do I need to add some magic to
With Asterisk 1.4 I could use commands like:
/usr/sbin/asterisk -rx sip notify reboot-snom mjc_home
to reboot a snom phone. Now, with 1.6, when I try that, I get:
Unable to find notify type 'reboot-snom'
Command 'sip notify reboot-snom mjc_home' failed.
Do I need to add some magic to
Hi,
I use hints to drive the LEDs on my snom phones, something like:
exten =
601,1,Dial(SIP/mjc_officeSIP/mjc_homeSIP/mjc_labSIP/mjc_serverSIP/mjc_library,20,trj)
exten = 601,2,Voicemail([EMAIL PROTECTED],u)
exten = 601,102,Voicemail([EMAIL PROTECTED],u)
exten =
601,hint,SIP/mjc_officeSIP/mjc_homeSIP/mjc_labSIP/mjc_serverSIP/mjc_library
Sometimes asterisk gets confused, though, and reports my extension as
in-use, even though no channels are active. Dialing something makes the
hint report inactive - the states are inverted, in other words.
How can
Philipp Kempgen wrote:
Hmm, I'll see if that gives me any clues...
Or you could try 'sip show inuse'.
Thanks, Philipp! I never noticed that command; I'm sure it will be very
handy for debugging.
- Mike
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Craig Van Ham wrote:
I had weird issues when using a Sonicwall, gave up.
Same here, avoid them! I use the SnapGear SG560 now.
- Mike
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I've had a good ongoing experience using http://www.unlimitel.ca. They
are responsive and reliable.
Ditto here - Unlimitel is small but reliable and supportive.
- Mike
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How can I play wav49 or gsm voicemail files on FC6? Nothing seems to
play them (hxplay, xine, mplayer, etc). I think I have all the normal
codec packages installed.
I can play regular wav files, but they're too big.
- Mike
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Gordon Henderson wrote:
On Mon, 5 Feb 2007, Dr. Michael J. Chudobiak wrote:
How can I play wav49 or gsm voicemail files on FC6? Nothing seems to
play them (hxplay, xine, mplayer, etc). I think I have all the normal
codec packages installed.
Have you got 'sox' installed? It comes
Derek Whitten wrote:
switch voicemail to .ogg format
voicemail.conf:
format=ogg
but you can't actually do that, can you?
WARNING[9933]: file.c:984 ast_writefile: No such format 'ogg'
mp3 would be better, but it doesn't work either.
WARNING[9879]: file.c:984 ast_writefile: No such format
OCG Technical Support wrote:
After a variety of connectivity problems, my itsp (Unlimitel.ca) blamed
the problem on the IAX protocol. They told me that as of Asterisk 1.4
the IAX protocol went downhill and many carriers (like VoicePulse) are
discontinuing support for IAX.
Is this
The choice of router/NAT is critical though. Unlimitel recommended the
SnapGear 560 to me, and it eliminated all the issues I was having with
IAX going through my Sonicwall devices.
Just another datapoint for you...
Just curious.
Since IAX only uses ONE port, do you have any idea what the
The sangoma has hardware echo cancel ?
If so it makes sence, because the settings in zapata.conf
are for the software echo cancel, and that should be
disabled for all interfaces that have hardware echo can.
No, that is incorrect. From
http://wiki.sangoma.com/wanpipe-asterisk-configure:
The
Mike wrote:
Thanks Tony. Its possible that the phone stops sending RTP stream (but it
certainly is receiving some!). How do I get Asterisk to stop caring whether
it receives RTP or not?
Yes there is a NAT between the phone the the Internet. The Asterisk server
doesn't have NAT though.
My
Kohler, Jeffrey wrote:
I am encountering an intermittent issue where some of my calls are being
dropped. Most of the calls that are made are successful. However, some
calls will be dropped after having been connected for some time.
Each time a call gets dropped, I get output similar to the
[EMAIL PROTECTED] wrote:
Just upgraded my * box to 1.2 and don't seem to be able to get MWI working.
Worked with my previous installation. My conf files are the same ( except
for a few 1.2 changes ). I've tried:
In sip.conf
fromuser=Anyname
fromdomain=my * ip
vmexten=7000
Are you missing
Barry Fawthrop wrote:
Hi all
Anyone using a sonicwall firewall ?
I have been and then suddenly it drops UDP packets because SIP is no
longer on port 5060 but some random assigned port ?
Why ?
Which Sonicwall model? Some (like the TZ170) have special VOIP settings,
like Enable consistent
Hi all,
I currently have an Asterisk test server behind a TZ170 Sonicwall
firewall / NAT box, with several DIDs.
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly documented by
I've found that inbound IAX2 calls don't work reliably (i.e., I get a
busy tone) unless I enable Use Consistent NAT in the Sonicwall. This
feature is poorly documented by Sonicwall, so I thought I'd pass it along.
I've used the iaxcomm softphone and a snom 200 behind serveral different
Hi all,
I want to link three incoming Bell Canada centrex pstn lines (which
currently go to an old norstar pbx) into asterisk.
Can anyone suggest the most painless (i.e., just works) way to do
this? Has anyone used the D-link DVG-3004S four-port FXO-to-sip adapter,
or the twice-as-costly
OK apart of my beleive that sonicwall is a piece of crap (personal), try
to do a port forwarding for the IAX port (4569)
Saul,
Why do you consider Sonicwalls to be crap? Aside from this odd issue
(which is fixed by using an obscure setting) they've been rock solid for
me, for years.
- Mike
Hi all,
I can't figure out why my TDM400P (with one FXO plugin) won't answer any
calls. There are no messages in the Asterisk console when a call is
placed to the FXO line from the PSTN. Any suggestions would be most
appreciated.
The wctdm and zaptel modules are loaded:
[EMAIL PROTECTED]
[EMAIL PROTECTED] wrote:
If so, is there a way to detect the hangup?
Check out
http://www.asteriskguru.com/tutorials/resolving_hangup_detection_problems_fxo_tdm_voicemail.html
for some possible clues.
- Mike
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Hadley Rich wrote:
Hi all,
I have hit a wall configuring a TDM400, I have set these up before without
issue but today I just can't seem to figure out what I am doing wrong.
I couldn't make TDM400/FXO work on my 1.2.5 Asterisk either. It wouldn't
answer calls, for unknown reasons. I gave up
D-Link has a 4 port FXO device on their site.
http://www.dlink.com/products/?sec=2pid=451
Apparently it hasn't shipped yet and costs $500.00
I've been testing a AudioCodes MP104-FXO-C3S (around $1000) 4-port FXO
box. It works, but the number of configuration options are staggering,
complex,
Hi Mike,
may I ask where you purchsded your A200 card from?
I managed to get one of the pre-production cards from Sangoma back in November,
however there are some bugs with it and I am unable to flash the
firmware or run latest drivers with it.
Sure, I got it at:
Hi all,
I have several locations, each connected by a Sonicwall VPN through
PPPOE DSL, with Snom 360 phones.
I've found that I have to tweak the Asterisk server MTU (inside one of
the firewalls) to get everything to work just right. Set the server
MTU too low, and the Snom phones don't
Can anyone suggest a good source of simple-but-distinctive wav ringtones
for a business environment, to use on Snom phones? The built-in Bellcore
tones are hard to distinguish, to my ear.
I want variations of ring, ring, not Madonna or Eminem :-)
- Mike
TWV wrote:
By now, every Snom fan should have installed the 6.0 (beta) firmware :-)
See http://www.snom.com/wiki/index.php/Beta_Firmware
I had to revert back to 5.5, because 6.0 kept garbling my LCD screen
(the screen would become unreadable). You might want to wait for 6.0.1 :-)
- Mike
Hi all,
One of my users reports frequently hearing echo on her Snom 360 phone,
even while talking to other Snom phones (via Asterisk) on the same LAN
(i.e., all-digital low-latency connection). I can never reproduce it
though, and swapping the phone didn't help.
Has anyone else seen mystery
Hi all,
I've been having problems with my A20002D lately - callers from the PSTN
don't hear me when I answer, but I hear them. Disabling echo
cancellation in zapata.conf brings the audio (and echo) back. This used
to work fine, until two days ago.
The only weird thing in the logs is this:
Hi all,
I've been testing various codecs to eliminate choppiness that I
sometimes get on my Asterisk IAX2 DSL provider (Exgn) connections,
and Speex seems to work the best, so far - but Speex seems oddly unpopular.
Can anyone share their experiences with Speex (good and bad)? Is anyone
Last time I had this problem was following a unclean powerdown and the
solution was:
- Kill Asterisk
- Stop wanpipe
- cd /etc/wanpipe/wan_ec
- In there there should be 2 files:
wan_ec_pid
wan_ec_socket=
- Delete those files
- Perform a reboot of your
I've been having problems with my A20002D lately - callers from the PSTN
don't hear me when I answer, but I hear them. Disabling echo
cancellation in zapata.conf brings the audio (and echo) back. This used
to work fine, until two days ago.
Well, just to complete my own thread, this seems to
Remco Barende wrote:
Most people seem quite positive about Snom phones, I cannot share this
opinion.
The displays are dying quite often, and firmware is buggy. I have tried
every firmware from 4.5 up to 5.x and 6.04 but keep having problems with
phones locking up or rebooting during an
Old English saying A bad workman always blames his tools
I don't think that's fair... these are very complicated phones, made in
China for very low prices. Problems do occur with them.
Some Snom LCDs do have problems.
There are firmware glitches, though I've only run into minor ones.
Steven wrote:
Make extensions that can hold a call. (like a 701)
Make this extension hintable for use in button programming.
If I am on a call and hit a non-lit button, it parks it there.
If I am not on a call and push the lit button, I connect to the park.
I suppose that if you are on a call
Just after some info on the Snom 320 before I got out an buy some...
I'm looking to use the shared line feature and hints with * so that i
can monitor the activity of other users, but I'm not sure If this also
turns the programmable buttons into a speed dial for quick transfers etc
(or if it
I'm getting a slight echo...sometimes...it varies from call-to-call,
but the biggest problem I have is a constant hiss in the background.
Again, this varies from call-to-call. I know my SIP phones are fine
as SIP-to-SIP calls on my LAN work perfectly. I only have problems
going out to the PSTN.
Derek Lee-Wo wrote:
With this card, would you say your audio quality is identical to that
of an analog phone connected directly to the PSTN? I'm trying to
understand if I should expect some audio degradation when going
through Asterisk.
In my experience, this card provides the sames quality
While I agree that the Sangoma cards are good, your statement that software
echo cancellation doesn't really work is ... incorrect.
Software echo cancel works very well if it's done correctly, if your audio
levels are where the canceller's sweet spot is, and the tail is not longer
than the
I've been having problems with incoming IAX2 calls - some work, but a
large fraction are answered with dead air or disconnects from my IAX
provider.
Disabling the jitterbuffer seems to eliminate the problem (so far)! Has
anyone else seen this? I'm using 1.2.6, but I'm not sure what my
Dr. Michael J. Chudobiak wrote:
Disabling the jitterbuffer seems to eliminate the problem (so far)! Has
anyone else seen this? I'm using 1.2.6, but I'm not sure what my
provider is using.
Oops, the problem still happens without the jitterbuffer - so something
else is causing it. Any ideas
I looked long and hard at the LAN and it was basically narrowed down to the
switches. In this smaller install, several cheapo Dlink ($30) switches
de-aggregate a Cisco Catalyst switch. What I noticed was that any phone
plugged direcly into the Catalyst did *not* lock up or reboot. Any phone
Blaming the 3com switch is very likely to be the wrong root cause. High
probability the 3com was not configured properly for the phone.
Just curious - what configuration issues did you have in mind?
- Mike
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Hi all,
I'm having trouble with incoming IAX2 calls on 1.2.7.1 - mostly they
work, but sometimes the caller just gets dead air or disconnects. IAX2
debugs show HANGUP and INVALID codes in these cases, rather than a
proper RINGING transaction.
My firewall is doing NAT, and changing the
If memory serves me properly what you are showing looks correct. You
server is registering to your provider on port 4569 as it should. Their
server is seeing you register from 64.26.155.62 and using the prt 14353
which is the port that your firewall has given that outgoing connection.
There isn't quite enough info in that log to tell what is going on.
What you have above is part of 2 separate conversations.
You have the tail end of a successful registration with 70.87.18.51
and the HANGUP of a call with 64.26.157.230 which your asterisk seems
to be confused about.
Could you
Mike Garey wrote:
It turns out that the Sangoma card had suddently decided to stop
answering on channels 2,3 and 4, so if someone was using channel 1,
then no other calls would be picked up. We could, however, make
outgoing calls. I tried restarting Asterisk and it didn't make a
difference.
Brian Swan wrote:
I've spent the last week or so troubleshooting echo problems at my
Wife's business, and I've been able to clear up about 99% of the echo,
but there is still a little residual echo that I can't seem to tweak
out. The users describe it as buzzing or crackling, but what it
Mimmus wrote:
Hi,
I'm trying my new Snom 360 phone (6.2 firmware) and I'm seeing that it
doesn't register with the Asterisk 1.2.9.1 server after a reboot. I need to
click Re-register in the web interface.
I think that was fixed in 6.2.1. See
http://www.snom.com/wiki/index.php/Beta_Firmware
I found my error: the TDM01B (1-port FXO TDM400P bundle) ships with
the single FXO module in position 4, not position 1. Thus using fxsks=4
in zaptel.conf and channel = 4 in zapata.conf fixed things.
- Mike
*CLI -- Executing Dial(SIP/555-83ee, ZAP/1/92262802) in new stack
Jul 23 13:50:24
Hi,
Does anyone know how to tell Asterisk to transmit a switchhook flash? I
have Asterisk attached to the ATA adapter of my legacy PBX, and some
features of the ATA/PBX are supposed to be accessed using an analog
telephone's switchhook flash or link button. I want to emulate this.
- Mike
Has anyone used the programmable buttons on the SNOM 200 with Asterisk
call parking? Do they work nicely together - i.e., do the LEDs show
which parking spots are in use, and does the press-to-park button
function work?
- Mike
___
Asterisk-Users
Hi,
I'm trying to set up my SNOM 200 with extensions, with different
ringtones - but it doesn't seem to work.
I've defined two extensions for it in Asterisk and in the SNOM 200
configuration. In the SNOM homesettingsSIPLines config page, I have
set the ringer for the first extension to
The distinctive rings still fail to work after upgrading to 3.35.
(However, the message-waiting indicator is much more reliable now!)
- Mike
I'm trying to set up my SNOM 200 with extensions, with different
ringtones - but it doesn't seem to work.
You may want to try the newest version 3.35.
Steve,
Yes, I can set distinctive-ringing-by-contact (SetupPreferences) just
fine, but I would prefer to use distinctive-ringing-by-line
(SetupLine2Ringtone, for instance). I far as I can tell, the
per-contact ringing works and the per-line ringtone settings don't
actually do anything.
I
I'm trying to get call parking working with the lighted buttons on the
SNOM 200. I have set the 5 buttons to Park Orbit, for extensions 700-704.
Pressing the first button (x700) does park the call. However, the
remaining buttons (x701-704) don't allow me to pick up parked calls, or
show
Anyone had any luck using the programmable keys for anything but
transfering/calling sip url's?
Nope. See this page:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20snom%20call%20park
- Mike
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[EMAIL
Actually, I should clarify my last reply:
I can park calls using x700 (programming one button as park orbit,
[EMAIL PROTECTED]), but I can't pick up parked calls using programmed
buttons.
I'm using v3.35 firmware.
- Mike
Steve Woolley wrote:
I would like to use one of my Snom 200's 5
Koopmann, Jan-Peter wrote:
Hi,
Has anybody experience with Snom360 and Firmware 6.X with Asterisk 1.2.X? I
am currently using Firmware 5.5 without serious problems but wanted to make
sure 6.X will work as well (including subscription etc.)
Use the very latest - 6.2.1. It seems quite good.
Is it possible to set up some sort of call-quality statistics
reporting/logging for IAX2 calls? Something that can keep track of
dropped packet / jitter trends?
(I know iax2 show channels shows this info for active calls.)
Suggestions appreciated!
- Mike
Von L. wrote:
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.
They each have NAT routers, and I have triple checked that they have
opened/forwarded the
Von L. wrote:
plugged in. They work immediately after being plugged in, but they lose
the ability shortly thereafter. They can always make outbound calls, but
only to real phone numbers, not extensions.
They each have NAT routers, and I have triple checked that they have
opened/forwarded the
shadowym wrote:
Yes, I am using the 1.2.7.1 patch on 1.2.9.1. It seemed to work fine.
Still curious if anyone has this working on an Aastra phone? I can't get it
to work but someone in the bug.digium.com list said they had it working on
an Aastra phone. Maybe I am missing something. I tried
Alex Robar wrote:
Hi all,
I have a Sangoma A200 card with hardware echo cancellation. The card has
12 ports (10 of which are active; All FXO). Twice on this particular
card I've seen all ports simply stop receiving incoming calls. There is
no other indication of this, however. I am able to
- Ability for the phone to ring when the receptionist is on one call
and a second or third call is incoming. (this has been the biggest
frustration up to now. When a second call comes, there is no tone
that heard on the IP500. Perhaps I am missing a setting?)
The Snom 360 can
David Gagnon wrote:
Finally, in the trunk all the states of my device are broken. If I
downgrade to 1.2.10, everything is fine. The device get busy and
ringing. But in the current trunk Asterisk SVN-trunk-r40632M none of my
hints works.
Anyone could confim this bugs ?
David,
I haven't
Hi all,
I'm using 1.2.9.1, with the metermaid patches to show parking spot
status on Snom BLF lights.
I see from http://www.asterisk.org/node/97 that the metermaid code has
changed substantially since 1.2.9.1.
Is anyone successfully using the new metermaid functionality in 1.4.x?
I'd like
I'm using 1.2.9.1, with the metermaid patches to show parking spot
status on Snom BLF lights.
I see from http://www.asterisk.org/node/97 that the metermaid code has
changed substantially since 1.2.9.1.
Is anyone successfully using the new metermaid functionality in 1.4.x?
Did anyone get
J. Oquendo wrote:
Andrew Latham wrote:
you are asking about Shared line apperance or hints. Look at this
http://www.voip-info.org/wiki/view/snom+360
Been there done that page. Nothing worth noting in there.
Do the line appearances work on the 12 non-sidecar buttons?
- Mike
randulo wrote:
On Nov 30, 2007 1:40 PM, Steve Totaro [EMAIL PROTECTED] wrote:
solved these issues. I think trunking (one of the main selling points
of IAX due to less overhead) may be a common denominator.
That does tend to explain why I've never experienced (or at least
noticed) problems.
Phil Knighton wrote:
Been through lots of stuff in the forums, and as far as I can tell I
have got the hints setup correctly and everything *should* be working
fine. There must be something different within 1.4 that I'm missing?
Yes, the metermaid format changed slightly. See the Parking
Brent Davidson wrote:
I thought I had the echo out of the system, but it keeps coming back...
What I'm being told is that when the users call out from their snom
phones they hear their own voice. There's no delay, but it's extremely
Does it happen on all-digital calls (e.g., intercom
On 10/14/2009 01:29 PM, David Wathen wrote:
Hi Myles,
Thanks to you and everyone else that has responded. I've really learned a
lot. pFSense and IPCop sounds let best so far for LINUX based firewalls.
I'm also wondering if anyone has any suggestions for a standalone firewall
appliance like
Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason
a0 on CPU 0.
Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely
on the PCI bus.
Nov 12 08:54:27 steerpike kernel: Dazed and confused, but trying to continue
Would my Digium TDM410P cause an NMI, or
On 11/12/2009 09:42 AM, Francesco Peeters wrote:
Dr. Michael J. Chudobiak wrote:
Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason
a0 on CPU 0.
Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely
on the PCI bus.
Nov 12 08:54:27 steerpike kernel
On 11/12/2009 09:31 AM, Dr. Michael J. Chudobiak wrote:
Nov 12 08:54:27 steerpike kernel: Uhhuh. NMI received for unknown reason
a0 on CPU 0.
Nov 12 08:54:27 steerpike kernel: You have some hardware problem, likely
on the PCI bus.
Nov 12 08:54:27 steerpike kernel: Dazed and confused
Hi all,
I'm running 1.6.2.0-rc6, and I'm running into a problem: sometimes the
audio vanishes in the middle of listening to an IVR background prompt.
This happens with both analog (Digium card) and IAX2 incoming calls.
The prompts are stored in ulaw format (and the IAX2 calls use ulaw).
The
Hi all,
Does anyone else use the SG560 firewall with Asterisk? I do, and it
normally works great, except when it randomly reboots. Has anyone else
experienced this annoyance? Did you fix it?
- Mike
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On 11/24/2009 02:14 PM, David Backeberg wrote:
The asterisk console claims that the IVR prompts are proceeding in the
expected fashion, but I can't hear anything.
Are you playing with the system clock?
...
dramatic ntp changes?
No, that shouldn't be happening. But I'll keep it in mind while
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