?
Thanks,
Brent Davidson
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Julian Lyndon-Smith wrote:
I am trying to track a strange bug down, and need to ask a really stupid
question, just so I can eliminate the possibility ..
When a SIP channel is hung up, I import a variable called MEETMEROOM
from the BRIDGEPEER channel, and if it is set, jump to another part
Steve Murphy wrote:
On Thu, 2008-09-25 at 14:21 +, Tariq .. wrote:
Greetings,
i have two asterisk servers running on Centos with asterisk 1.4.21 and trixbox..
i tried to creat an SIP link between both servers and i discovered that one of my servers is not allowing the other to send calls
Manolet Gmail wrote:
Have, i want to create a sip extension to a context in my dialplan.
how i can do that?
___
Simple. Use a Goto:
[context1]
exten = 123,1,Goto (context2,456,1)
[context2]
exten = 456,1,Background(tt-monkeys)
Steve Totaro wrote:
On Tue, Aug 19, 2008 at 7:10 AM, Jim Boykin [EMAIL PROTECTED] wrote:
We run asterisk to handle incoming DIDs and we have observed
inefficient Codec Translation.
Here is the scenario
[DID Vendor] --- [Asterisk ]
External
Doug Lytle wrote:
Eric ManxPower Wieling wrote:
But what would you call it? It's not a card, so it can't be a NIC, right?
(N)etwork (I)nterface (C)ontraption
Doug
(N)etwork (I)nterface (C)onnector
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I have a question about click to dial. Each of my users is going to
have a VOIP phone with an assigned extension. Is there a simple way to
build a web-based speed-dial list that will allow them to put in their
extension, click on the number they want to dial, and have asterisk ring
their
So you basically want a call-interrupt feature that puts the interrupted
party on hold?
rachid wrote:
Hi,
I want to make an insertion in a communication; A et B are in
communication, an other C wants talk to A, how can i set B on
hold state and make a call to A?.
Thanks.
Rachid
Robert Goodyear wrote:
Yeah I'm thinking either homeland security or some other
identity-critical legislation might be on my side here.
On Thu, Jul 3, 2008 at 12:40 PM, randulo [EMAIL PROTECTED]
mailto:[EMAIL PROTECTED] wrote:
On Thu, Jul 3, 2008 at 9:26 PM, Robert Goodyear
[EMAIL
bkruse wrote:
Yes, probably, same basic error.
-brandon
Fidel Garcia wrote:
Great info! Thanks!
However, they do not mention the fact that when you create a new user you
cannot select the DialPlan. I wonder if the path fixes both issues. Any
idea?
Fidel Garcia
System Engineer
And there are people like me who still can't get PRI's for less than
$1100/month. (Granted, I doubt I'll ever need a pri for the business I
am with now, but I was with an ISP for a long time that still supported
dial-up and we had 8 PRI's with a bulk discount that got them for us at
I'm using the Rhino R4FXO cards to answer incoming voice calls and just
had this idea last night... Is there any type of module for asterisk
that will divert a call to a softmodem? For example, I call in on the
voice lines, get the main menu, instead of dialing an extension to get a
person I
Steve Totaro wrote:
On Wed, Jun 11, 2008 at 1:47 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Wed, Jun 11, 2008 at 11:53 AM, Raj Jain [EMAIL PROTECTED] wrote:
On Wed, Jun 11, 2008 at 9:17 AM, Brian J. Murrell [EMAIL PROTECTED] wrote:
I'm wondering if the SIP lines can start
Steve Totaro wrote:
On Wed, Jun 11, 2008 at 2:44 PM, Brian J. Murrell [EMAIL PROTECTED] wrote:
If you ever have problems with a call dropping after 30 seconds,
Answer() is usually the cause.
Thanks,
Steve T
This is the first I've heard of this. I've never actually had the drop
after
Philipp von Klitzing wrote:
Hi!
I am actually interested in the topic of this post. Ast 1.6 vs 1.4. We
are using asterisk 1.4.2 for a SIP only based configuration. [...] We
are planning to accomodate about 5,000 users on this server.
Many people on this list will advise you to use a
Correct me if I'm wrong, but unless you pass specific options to the
dial command to have it override the ringing then when you dial out, you
hear the audio from whatever channel you're dialing on. So the tones
you are hearing are from the telco. The ring cadences defined in
indications.conf
that
is not an option.
Given this setup, is there any reason for me to switch to Asterisk 1.6
or should I stick with 1.4?
Thanks,
Brent Davidson
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Tzafrir Cohen wrote:
On Thu, Jun 05, 2008 at 03:00:01AM +1000, Rob Hillis wrote:
Brent Davidson wrote:
We're currently using Asterisk 1.4.19, Zaptel 1.4.10,
Oslec SVN, Rhino R4FXO-EC cards, and Snom 300 Phones.
Why on earth are you running two layers of echo cancellation
Matt Watson wrote:
Have you tuned rxgain txgain in Zapata.conf? shameless-self-plug
http://www.mattgwatson.ca/2008/05/howto-tune-zaptel-dahdi-fxo-interfaces-on-asterisk-pbx/
/plug
Also, have you used fxotune to tune each FXO interface?
I believe echo cancellation happens at the Zaptel /
Just an update. I tried updating to the newest Rhino Release firmware
1.15 and newest stable driver version 2.2.6. It works OK with
zaptel-1.4.9.2 and compiles OK with 1.4.10.1 but when compiled against
zaptel 1.4.10.1 Asterisk does not see any zap channels. I'm currently
running one branch
Another solution that works for me is to add Playback(silence/1) just
before whatever you are about to do. Something about the playback
command opens the channel up.
-Brent
Sherwood McGowan wrote:
Alan Lord wrote:
Sherwood McGowan wrote:
snip /
Hrm...I have
button as type DTMF with that key sequence
assigned to it (in this case *1). (There may be another way, but this
seems simplest.)
Let me know if you need any more help.
-Brent
Thermal Wetland wrote:
On Thu, May 8, 2008 at 5:20 AM, Brent Davidson
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED
Which phones are you using and what software revision. I've had a crash
course in Snom phone lately and can probably help with at least the park
orbits.
-Brent
Thermal Wetland wrote:
I would like to hire someone to help us tweak our asterisk system for
Snom phones.
We would like to
A friend of mine recently told me about a phone system his office was
considering that did not use any handsets. Instead of a phone, each
person was issued a BlueTooth headset and several bluetooth repeaters
were installed throughout the building. When a call came in, it would
be routed to
interrupting the call.
For smaller setups there is 3Com WXR100 that supports up to 3 MAPs (Managed
Access Points).
AttVinícius FontesDesenvolvimentoCanall Tecnologia em Comunicações Ltda.
- Brent Davidson [EMAIL PROTECTED] escreveu:
A friend of mine recently told me about a phone system his
Jerry Geis wrote:
The netstat show 0.0.0.0
netstat -anp | grep :69
udp0 0 0.0.0.0:69
0.0.0.0:* 4007/xinetd
--
cat /etc/xinetd.d/tftp
# default: off
# description: The tftp server serves files using the
Jerry Geis wrote:
Brent, below is the file. Looks good to me... Also Both networks start
at boot. Nothing is manual on this box at all.
--
# Simple configuration file for xinetd
#
# Some defaults, and include /etc/xinetd.d/
defaults
{
instances
John Signorello wrote:
excuse me...
But did you not just post
[asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New
Digium Boards Cheap X305 $199
Did you not provide a link to a COMMERICAL entity?
Wasn't your a post a unsolicited post, that is, not in response to a
Alex Balashov wrote:
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed in a Dell PowerEdge 2850
or 2650 (cannot recall):
00:00.0 Host bridge: Broadcom CMIC-WS
Tzafrir Cohen wrote:
On Wed, Apr 16, 2008 at 09:44:25AM -0500, Brent Davidson wrote:
Alex Balashov wrote:
Greetings,
This may have already been asked many times, but I cannot seem to find a
satisfactory and consistent answer anywhere.
I have an X100P card (from x100p.com) installed
this problem or have any ideas how to
eliminate it?
Thanks,
Brent Davidson
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Brian J. Murrell wrote:
On Wed, 2008-04-09 at 17:21 -0500, Brent Davidson wrote:
Does anyone know if Asterisk will convert an inband DTMF from one sip
channel to an info or rfc2833 DTMF on another (i.e. bridged) SIP
channel?
You might also try canreinvite=no for both your phone
Jon Pounder wrote:
I had the phantom rings for years, once a day same time roughly every
day, finally just got annoyed enough one day I trapped the telco on
the phone with me till I finally got to talk to the right person. The
right person knew instantly what I was talking about after
Brian J. Murrell wrote:
On Thu, 2008-04-10 at 11:13 -0500, Brent Davidson wrote:
You might also try canreinvite=no for both your phone and the sip
peer.
Yeah, there is definitely no re-inviting going on. Both Asterisk and
the local handset are in a local network behind NAT
I have an operator queue that is supposed to ring 2 phones, extension 10
and 11. Everything is working correctly but I keep seeing these
messages in my log: The device state of this queue member, Sip/10, is
still 'Not in Use'. Everything I've been able to find on this message
so far points
Lee Jenkins wrote:
Brent,
I had a similar problem and I feel for you, its frustrating.
Are you using polycom phones by chance? Here is the problem that I had, not
sure if your problem is related.
Specs:
- 6 Polycom 301 phones.
- CentOS 4 Server with Asterisk 1.2.x
- Sangoma A200 card
Asterisk Development Team wrote:
The Asterisk.org development team has announced the release of Zaptel
versions 1.2.25 and 1.4.10. These releases contain many bug fixes as
well as performance enhancements.
A couple of the more major changes include: modifications to the
wctdm24xxp and
Jason Parker wrote:
Brent Davidson wrote:
Do they mean 1.4.20 instead of 1.4.10? If not, then this message was
seriously delayed :-D
-Brent
Zaptel, not Asterisk. :)
1.4.10 is correct.
Doh! My bad.Was looking at the wrong version numbers. As many
times as I've
Have you tried the using the SIPDtmfMode function in your dial plan?
It can be used to change the DTMF mode between two points in a call.
The problem, I would think, would be if your phones are set up to ONLY
send inband audio then you have to find someway to get audio to
transcode the DTMF
You could also, conceivably, handle this outside of asterisk by using a
more complex MOH stream source. For instance, use a shoutcast client as
the MOH source, run your own shoutcast server streaming your music and
have a script set up to periodically interrupt the stream being served
to the
Greg Woods wrote:
Shaun Ruffell wrote:
svn co http://svn.digium.com/svn/zaptel/branches/[EMAIL PROTECTED]
zaptel-1.4-4122
Thank you, I will try that tonight when I get home and report back.
--Greg
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the MOH to ringing and the
majority of the complains stopped. (The remaining complaints are
related to DTMF detection problems.
Just food for thought.
Good luck,
Brent
Atis Lezdins wrote:
On Wed, Apr 2, 2008 at 11:05 PM, Brent Davidson
[EMAIL PROTECTED] wrote:
You could also, conceivably
With canreinvite=no you are forcing asterisk to remain in the call
path. As long as Asterisk is in the call path, it is supposed to be
transcoding the calls, so it doesn't care what the compatible codecs are
between then endpoints. Each leg of the call is phone-asterisk so
asterisk
audio examples to anyone that is interested.
Thanks,
Brent Davidson
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LOL
Reminds me of that old Ray Stevens Song - Jeremiah Peabody's
Polyunsaturated Quick Dissolving Fast Acting Pleasant Tasting Green and
Purple Pills
Oh Yeah Binary System = Pyramid Scheme
BJ Weschke wrote:
I'll give you an A+ for originality after I get done laughing and then
to further relax the
DTMF detection?
Thanks,
Brent Davidson
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system, and it is not limited to any one caller.
Thanks,
Brent Davidson
Brent Davidson wrote:
Still grasping at straws trying to solve DTMF detection issues with one
of my asterisk servers. This particular server is now running Asterisk
1.4.18.1 and Zaptel 1.4.9.2 in runlevel 3 (console
Does anyone know of a way to make a Snom 300 phone monitor the parking
lot extensions and allow one-button pickup with the programmable buttons?
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To
I've been reading most of the day and can't seem to find a clear
definition of the syntax for parking lot hints in AEL2. I have tried
all of the following and they either do not light up the line button on
my Snom 300 or give syntax errors:
hint(park/701) 701 = {
ParkedCall(701);
}
I seem to be having similar problems at one of my branch offices. See
my message on Intermittent DTMF issues for some of the standard
replies. Have you tried the RelaxDTMF tag in zapata.conf? I don't
think Gain calibration applies to T1 cards or I would recommend that as
well.
Thanks,
Do you have canreinvite=no in the sip client configuration? If not then
the two sip phones are probably issuing a reinvite command and taking
asterisk out of the call path. If that happens and the phones can't
reach consensus on a codec then you run into audio problems. If you're
not a
about 1 call in 5. I'm wondering if
it's just because they call us more than any of our other customers or
if there is some peculiarity with their phone system. Anybody have any
ideas what to try next?
Thanks,
Brent Davidson
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an extension about 1 call in 5.
I'm wondering if
it's just because they call us more than any of our
other customers or
if there is some peculiarity with their phone system.
Anybody have any
ideas what to try next?
Thanks,
Brent Davidson
because they call us more than any of our
other customers or
if there is some peculiarity with their phone system.
Anybody have any
ideas what to try next?
Thanks,
Brent Davidson
Looking for last
PROTECTED]
[mailto:[EMAIL PROTECTED] *On Behalf Of *Brent
Davidson
*Sent:* 15 February 2008 00:30
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] ISDN PRIs and taking a server down
formaintenance - blocking issue
Correct me if I'm wrong, but as I
. Hope this helps someone else later on.
Thanks,
Brent Davidson
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I considered doing just that, but since I didn't have my scope with me
and it's an hour's drive away it didn't seem worth it at this point. If
we have trouble again I may take the scope down there and test it.
-Brent
Steve Edwards wrote:
On Thu, 14 Feb 2008, Brent Davidson wrote
. The only other way I'd know would be to hack the code for the
dial or answer command and build another command that simply takes the
channel off-hook and leaves it there.
Good luck,
Brent Davidson
Lyle Giese wrote:
If you take Asterisk down, the PRI should go down as the D channel is
down
and connect
to each of the branch office servers as needed never seems to have the
echo/sidetone problem. I need some help figuring this out.
Thanks,
Brent Davidson
Brent Davidson wrote:
We're deploying an asterisk-based phone system at all of our branch
offices in an effort to eliminate long
the OSLEC drivers and that doesn't
seem to have had any effect either. What am I missing?
Thanks,
Brent Davidson
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identity on the phone. With sip
set debug peer 15 on the Asterisk CLI I don't see any Notify messages
go by. I've been searching google for about 2 days on this and haven't
come up with anything useful yet. Everything I see indicates it should
just work.
Thanks,
Brent Davidson
Nevermind... I found the appropriate mojo. The key was putting
[EMAIL PROTECTED],password and removing the subscribemwi=yes. (I
think that's all it required. Among the other 1500 things I've already
tried, there may have been some residual.)
Thanks,
Brent
.
Brent Davidson wrote:
I think I
from/to a Zap channel, the calls are perfectly clear.
I am completely lost at this point. Any ideas?
Thanks,
Brent Davidson
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.
Thanks for the suggestions.
Brent Davidson wrote:
I'm having a strange problem with transfers on IAX phones. I have two
IAX phones behind my firewall that are extensions from my office phone
system. Both phones can receive calls, but only one of the extensions
can do blind transfers
I'm having a strange problem with transfers on IAX phones. I have two
IAX phones behind my firewall that are extensions from my office phone
system. Both phones can receive calls, but only one of the extensions
can do blind transfers by pressing the # key. I have a similar problem
at the
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