Hi Greyman,
Thanks for your reply. I have gone through that link, but none applies
in my case, as I strictly need an outbound proxy such that it works
with target endpoints that are not registered with my OpenSER.
Most of the help available in various links suggest -
exten =
If you search the Asterisk bug tracker with outboundproxy you'll get a
few hits. This is the latest one:
http://bugs.digium.com/view.php?id=12006
The way I use outboundproxy is via realtime and apply it to each
sipaccount. If you set the outboundproxy for each of your SIP accounts
in sip.conf
Thanks Greyman, for the quick response.
True, I think I had read it somewhere that outboundproxy only works when you
specify it on a per-sip-account basis.
This approach works well if one has a limited set of target endpoints, but
becomes a show-stopper, if we have to generically allow any
This approach works well if one has a limited set of target endpoints, but
becomes a show-stopper, if we have to generically allow any remote domain
user to be called via our local outbound. Right?
That's correct as far as I am aware, I have the same problem. To call
arbitrary endpoints I
I am working on Polycom IP601 console with expansion module.
I want to put on the BLF (busy lamp field) feature on all the
contact/speed dial names I put on the console but I could not get it to
work.
*CLI core show version
Asterisk 1.4.13 built by root @ hostname on a i686 running Linux on
Hi all,
In my understanding, we can use mssql as a database of asterisk
thro' unixodbc. And we can easy using mysql (realtime) to do the
same. Now, I want to keep 2 connections, one is mysql and one is
mssql. Because both database have information that needed to be read
from asterisk. Can I
Hello,
About 4 years ago there used to be a script floating around to generate
dynamic graphs/diagrams of extensions.conf (the asterisk dialplan).
It was using GraphViz to perform the graphing.
Does anyone have a copy of this script, or a better solution to generate a
flowchart of my dialplan?
Hi,
I'm not sure at this point who will be with us, but there's always
something to talk about on the conference, Fridays at 12 Noon Eastern
Time, 9 AM Pacific, 4PM GMT. I have invited Garrett Smith whose blog
you should consider following.
PSTN:
(724) 444-7444 Call ID: 22622
SIP:
exten = 1234,
Thanks for your reply, Anthony
So it is a known bug? Does this happen to others?
The asterisk code is full of fun things where it checks for things like
that in multiple places but doesn't always handle every instance of the
same check in the same way. This is getting resolved piecemeal and
I am working on Polycom IP601 console with expansion module.
I want to put on the BLF (busy lamp field) feature on all the
contact/speed dial names I put on the console but I could not
get it to work.
John,
If I understand correctly (and that's my experience) the BLF will only light
up
If I understand correctly (and that's my experience) the BLF will only
light
up when the phone is ringing/on a call. Asterisk doesn't support all
those
fancy status that you can select from the phone.
Mike, thanks for your response.
I think my test is worse than that. I pressed DND on one
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: Thursday, April 17, 2008 7:59 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] End to end call monitoring?
On Thu, Apr
I do not know if you manage more than one box.
But if are setting up multiple boxes, you should look at cogoblue.com
It is a fully graphical configuration tool, not just dial plans but
every from inbound to IVR's
Disclaimer, I work for ISPBX
John
Matthew Gibson wrote:
Hello,
About 4
Lee, Picking up the phone does not constitute 'making a call' Asterisk
is unaware of any Sip events until the phone sends it. Usually the
phone will not send Asterisk any information until it is ready to place
the call (ie you have dialed enough numbers to make a match on your
dialplan (local to
John,
Please I know the job of any salesperson is to promote and
push their product every chance you get. But please this is as it says
in the mailing list name
Asterisk Users Mailing List - Non-Commercial Discussion
You are more than welcome to advertise your
Anthony,
What bug report ID# would that be? Not being a dev I find it hard to know
which of the 4 chanspy bug I need a patch for, since none of them seem to
refer to a 1.4.19 bug.
Mike
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Anthony
DND does not do anything for me BLF-wise either (shame). Simply picking up
the handset won't do, at that point the phone is giving you a dialtone but
nothing is sent to the server. You actually have dial out. Try actually
calling somebody, the state should change to InUse.
At first glance your
On Fri, Apr 18, 2008 at 09:04:43AM -0400, John Signorello wrote:
I do not know if you manage more than one box.
But if are setting up multiple boxes, you should look at
[$MY_CONFIG_TOOL]
And that answers the question exactly how?
That specific tool does provide visualization of the
On Fri, Apr 18, 2008 at 04:28:29AM -0400, Matthew Gibson wrote:
Hello,
About 4 years ago there used to be a script floating around to generate
dynamic graphs/diagrams of extensions.conf (the asterisk dialplan).
It was using GraphViz to perform the graphing.
Does anyone have a copy of
Silly newbie question. Is the license only required for the Digium
product? Or is it also required for the "unofficial, unsupported (other
than yourself, and those who want to help)"?
Thank you for clarification!
es
Moises Silva wrote:
Required by all (with the exception of academic work possibly?)
Thanks,
Steve Totaro
On Fri, Apr 18, 2008 at 9:55 AM, Eugen Soare [EMAIL PROTECTED] wrote:
Silly newbie question. Is the license only required for the Digium product?
Or is it also required for the unofficial, unsupported (other
On Fri, Apr 18, 2008 at 8:07 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Thu, Apr 17, 2008 at 5:14 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Raúl Gómez C. wrote:
Hi list,
snip
I think this is a very common scenario so, how are you doing to
handle this
Hi Steve and Mojo,
Thank you very much for your input, I will try this approach and I'll let
you know for the results...
Thanks again...
--
Nacho
Linux Counter #156439
On Fri, Apr 18, 2008 at 8:07 PM, Steve Totaro
[EMAIL PROTECTED] wrote:
On Thu, Apr 17, 2008 at 5:14 PM, Mojo with Horan
Just an FYI on a great server for cheap if anyone is looking for a
solid Asterisk platform. I don't work for these guys or have any
affiliation, I just saw this deal and used to use these and the X306s
for Asterisk deployments because they were trouble free as far as
hardware, IRQs, and OS. They
Another silly question,
In the first Digium link posted before there is a line that said *The G.729
codec works with all Digium cards*, but this license will work with a
Sangoma Remora Card??? Or do I need to buy it from Sangoma??? (I don't know
if the are selling G729 licenses)
Thanks...
--
Hey Alex,
Take a chill pill dude - he was suggesting it in reply as a solution to
someone else's problem.
If I ask a question (eg an Outlook dialer) and someone replies with
Snapanumber then that's an appropriate response.
Yes I've posted about Mexuar last year when I worked for them but
Even if you guys want to try to make a gray area out of what is black
and white to some of us, it wasn't what the OP wanted or asked for at
all... See Tzafrir's post if you cannot comprehend why the solution
solicited does not fit Matthew Gibson's request (there is no hammer).
Thanks,
Steve
Hi Jessi,
It seems that I have the same problem. I'm using an Asterisk with a
TE120P card with R2 cas signalling. The channels are working ok but the
PBX on the other side shows MFAS alarm. I have dig the code of the
zaptel drivers but I have not found anything where you can change the
bit pattern
Im sorry but I dont think he answered the users question by trying to hawk
commercial products, Id have to agree with alex here
even I find grave disdain for people doing this on the non-commercial lists.
the guy was asking for a simple script that was at one
time available, not a full blow
On Thu, 17 Apr 2008 13:25:19 +0100
Alan Lord [EMAIL PROTECTED] wrote:
[cut]
I bought an X100p card from ..
I have a similar card (X101P Tiger Jet) but seems does not
recognize dmtf: external pstn callers cannot select from menu
options, that is asterisk doesn't route the ext user selection
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
question???
There seems to be two
Ex Vito wrote:
On Wed, Apr 16, 2008 at 7:18 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
Ex Vito wrote:
Tested with no 4K stack kernel and stackcleanup svn branch
zaptel version. Correct, the kernel no longer complains about
the soft hangup.
However the system still hangs (console
On Fri, Apr 18, 2008 at 10:15 AM, giuliano curti [EMAIL PROTECTED] wrote:
On Thu, 17 Apr 2008 13:25:19 +0100
Alan Lord [EMAIL PROTECTED] wrote:
[cut]
I bought an X100p card from ..
I have a similar card (X101P Tiger Jet) but seems does not
recognize dmtf: external pstn callers
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
On Fri, Apr 18, 2008 at 04:15:32PM +0200, giuliano curti wrote:
On Thu, 17 Apr 2008 13:25:19 +0100
Alan Lord [EMAIL PROTECTED] wrote:
[cut]
I bought an X100p card from ..
I have a similar card (X101P Tiger Jet) but seems does not
recognize dmtf: external pstn callers cannot select
Apparently, there is a SIP(diversionheader) field that fixes the problem
below, but I cannot find any docs or examples of how to use it in my
dialplan. Any help would be appreciated. We have a Cisco CallManager
where users forward their numbers, so PSTN-PSTN calls get this error...
-Greg
---
On Fri, Apr 18, 2008 at 11:09 AM, John Signorello [EMAIL PROTECTED] 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
No steve it was a good post. $199 for a 1ru makes me look at my fat but
quiet 3ru's and begin to think.h :) how much noisier can they
be.
I think everyone should just take a chill pill.
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642
Quoting Steve Totaro [EMAIL PROTECTED]:
You may have a point although I was more doing a favor or looking
out rather than trying to push my wares on someone where they did not
fit.
Thanks,
Steve Totaro
I for one found your post about the cheap machines very useful and would
appreciate them
I believe I am close to fixing my problems with my 1.2 to 1.4.19 upgrade. I
have one question: to limit my customers to the number of channels they have
paid for, I use the GROUP feature. I also regularly check in the CLI what`s
going on using group show channels.
Basically, my system is
Raúl Gómez C. wrote:
Another silly question,
In the first Digium link posted before there is a line that said *The G.729
codec works with all Digium cards*, but this license will work with a
Sangoma Remora Card??? Or do I need to buy it from Sangoma??? (I don't know
if the are selling G729
On Fri, Apr 18, 2008 at 12:04:45PM -0400, Steve Totaro wrote:
I certainly was clear in the title that it was a server that cost $199
even if someone did not know that OT was short for Off Topic
I missed the post. Link?
-- j
--
Jay R. Ashworth Baylink
Brent Davidson wrote:
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
Try GROUP()=internal-...
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Mike
Sent: Friday, April 18, 2008 11:30 AM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [asterisk-users] Question on groups
I believe I am close to fixing my problems with my 1.2 to
On Fri, Apr 18, 2008 at 12:39 PM, Jay R. Ashworth [EMAIL PROTECTED] wrote:
On Fri, Apr 18, 2008 at 12:04:45PM -0400, Steve Totaro wrote:
I certainly was clear in the title that it was a server that cost $199
even if someone did not know that OT was short for Off Topic
I missed the post.
LOL!!! Thanks Mojo!
On Sat, Apr 19, 2008 at 12:07 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
Raúl Gómez C. wrote:
Another silly question,
In the first Digium link posted before there is a line that said *The
G.729
codec works with all Digium cards*, but this license
Wow that was easy. Thanks! My wrong syntax must have worked in 1.2 by pure
chance, or I must have erased the brackets by mistake while I was tweaking
the config files following the upgrade.
Thanks alot, you saved me alot of time and grief.
Regards,
Mike
_
From: [EMAIL PROTECTED]
On Fri, Apr 18, 2008 at 12:44:41PM -0400, Steve Totaro wrote:
By request
http://www.surpluscomputers.com/store/main.aspx?p=ItemDetailitem=COM10775
Sorry; I thought I'd taken that offlist. nice, but not SCSI capable
from what I can see. We run SCSI RAID-1 on all our diallers.
Nice pricing
Raúl Gómez C. wrote:
LOL!!! Thanks Mojo!
On Sat, Apr 19, 2008 at 12:07 PM, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
The codec in use for a specific channel doesn't even care if that
channel exists over zapata analog or digital cards, sip channels, iax[2]
channels, smoke
My post was made b/c John Signorello has done this before and I thought
that a friendly reminder of the proper places to post his 'offers'
should be posted.
This is the one that came to mind when I composed the email reply:
The codec in use for a specific channel doesn't even care if that
channel exists over zapata analog or digital cards, sip channels, iax[2]
channels, smoke signals, etc. If you care to use ping pong balls
and the atlantic ocean as your medium, you should be able to interface
with the
What's the situation with configuring Asterisk to handle
videoconferencing over a community wireless intranet? I found a radio
at www.ubnt.com that is called, Nanostation2. It has a range up to 15
kilometers, and sells for a mere $80.
Seems like a really low-cost community wireless network
Lol - you really do hate anyone doing anything commercial with asterisk
huh :)
Regards,
Dean Collins
Cognation Pty Ltd
[EMAIL PROTECTED]
+1-212-203-4357 Ph
+61-2-9016-5642 (Sydney in-dial).
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On
Atis Lezdins wrote:
The codec in use for a specific channel doesn't even care if that
channel exists over zapata analog or digital cards, sip channels, iax[2]
channels, smoke signals, etc. If you care to use ping pong balls
and the atlantic ocean as your medium, you should be able to
Where you calling my question silly? :)
Well here is another one..question that is.
I was not aware of the licensing of G.729. Is it that way with other
codecs? Some that I have heard are G.711 and SPEEKS/SPEAKS...
Perhaps pointing towards a resource that would have what ones require a
On Fri, Apr 18, 2008 at 01:45:36PM -0400, Dean Collins wrote:
Lol - you really do hate anyone doing anything commercial with asterisk
huh :)
No. But there are rules. And he has not really followed them despite
repeated reminders.
BTW: nobody really did answer the OP.
--
This is the first of several questions I have but to keep them thread
friendly I'll post each separately.
From reading the docs I see Asterisk will attempt to bind to all
addresses. What about all interfaces? I have a box with two NICs that
I'd like to connect to both the WAN (Internet) and
NO, I just feel that if you are going to make money, go ahead and make
it, but if you use the resources that people have built and sweat over
and given to you for free, the least you can do is show respect and
follow the rules.
Trust me, I like to make money, with a newborn on the way I NEED to
robert boardman wrote:
Hi,
I have a load of files recorded with MixMonitor that are out of sync ie
one leg of the call is 2-3 seconds behind the other,
is this a bug in Asterisk 1.4.18, or am I possibly doing something wrong
Is it possible to edit the file and re sync the a/b leg?
The outbound proxy support in 1.2 and 1.4 is buggy and basically
totally wrong.
That's why I have a branch in my svn repository with a rewrite of the
outbound proxy support for 1.4.
This is part of Asterisk 1.6.0 and future releases, but not of 1.4.
Enjoy!
Steve Totaro wrote:
On Fri, Apr 18, 2008 at 11:09 AM, John Signorello [EMAIL PROTECTED] 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?
Second questions.
Well possibly three questions.
Can I create in a context a priority that skips a chunk. The example in
Paul Mahler's book indicates so but I'd like to confirm, without/before
testing, my code.
This is desired so I can add/remove/augment dialplans/contexts that
have a
On Fri, Apr 18, 2008 at 4:15 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I just realized where this is coming from. I was attempting to patch
this from a different angle, but as soon as you mentioned the drastic
difference in load time I realized what had happened. I'm going to make
Anyone use the LDAP feature yet on the polycom phones? If so how well
does it work? How are you using it in your environment?
http://polycom.com/usa/en/products/voice/desktop/soundpoint_ip/applicati
ons/corporate_directory_access.html
Roy Anciso
Director of Technology
Manistee Intermediate
Is it possible to set an outboundproxy for REGISTER from Asterisk?
register = xxx:[EMAIL PROTECTED]
[foobar]
type=peer
host=sip99.foobar.com
disallow=all
allow=g729
canreinvite=no
secret=yyy
fromuser=xxx
port=5099
outboundproxy=xxx.42.149.69
However, SIP REGISTER still goes directly to
On Fri, Apr 18, 2008 at 08:35:26PM +0300, Atis Lezdins wrote:
The codec in use for a specific channel doesn't even care if that
channel exists over zapata analog or digital cards, sip channels, iax[2]
channels, smoke signals, etc. If you care to use ping pong balls
and the
On Fri, Apr 18, 2008 at 2:47 PM, BJ Weschke [EMAIL PROTECTED] wrote:
Steve Totaro wrote:
On Fri, Apr 18, 2008 at 11:09 AM, John Signorello [EMAIL PROTECTED]
wrote:
excuse me...
But did you not just post
[asterisk-users] OT Nice IBM 1U Server Gets Along w/Old and New
Digium
We've just upgraded to asterisk 1.4 and we have changed the way we
handle our calls a bit. This seems to be giving us a bit of an issue.
We now allow the phones to reinvite. In the rtp.conf file, i've set the
range from 1-2. However, when the phones begin talking to one
another, they
On Fri, Apr 18, 2008 at 9:51 AM, Tzafrir Cohen [EMAIL PROTECTED]
wrote:
On Fri, Apr 18, 2008 at 04:28:29AM -0400, Matthew Gibson wrote:
Hello,
About 4 years ago there used to be a script floating around to generate
dynamic graphs/diagrams of extensions.conf (the asterisk dialplan).
On Friday 18 April 2008 13:21:01 Roderick A. Anderson wrote:
This is the first of several questions I have but to keep them thread
friendly I'll post each separately.
From reading the docs I see Asterisk will attempt to bind to all
addresses. What about all interfaces? I have a box with
So this is what it has all come down to? Spamming the mailing lists...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dean Collins
Sent: Friday, April 18, 2008 1:46 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Ex Vito wrote:
On Fri, Apr 18, 2008 at 4:15 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
I just realized where this is coming from. I was attempting to patch
this from a different angle, but as soon as you mentioned the drastic
difference in load time I realized what had happened.
On Friday 18 April 2008 13:48:04 Roderick A. Anderson wrote:
Second questions.
Well possibly three questions.
Can I create in a context a priority that skips a chunk. The example in
Paul Mahler's book indicates so but I'd like to confirm, without/before
testing, my code.
This is
On Fri, Apr 18, 2008 at 1:17 PM, Rob Schall [EMAIL PROTECTED] wrote:
If not, is there a way to configure this range on the phones? (They are
polycom 501s).
You're right, how could the phone read your asterisk rtp.conf setup?
Download the SIP Administrator Guide for the 501 from the Polycom
Tilghman Lesher wrote:
On Friday 18 April 2008 13:48:04 Roderick A. Anderson wrote:
Second questions.
Well possibly three questions.
Can I create in a context a priority that skips a chunk. The example in
Paul Mahler's book indicates so but I'd like to confirm, without/before
testing,
Oops, I got that wrong... should have been
register = xxx:[EMAIL PROTECTED]
Doug.
- Original Message
From: Douglas Garstang [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Friday, April 18, 2008 11:56:27 AM
Subject: [asterisk-users] REGISTER Outboundproxy
Is it possible to
On Fri, 18 Apr 2008 08:37:32 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
If you care to use ping pong balls and the atlantic ocean as your medium,
you should be able to interface with the g729 codec if you still needed
to :D
I've heard that RFC1149-compliant devices work
Tilghman Lesher wrote:
On Friday 18 April 2008 13:48:04 Roderick A. Anderson wrote:
Second questions.
Well possibly three questions.
Can I create in a context a priority that skips a chunk. The example in
Paul Mahler's book indicates so but I'd like to confirm, without/before
testing, my
On Fri, Apr 18, 2008 at 3:53 PM, Godwin Stewart Horwich IT Services
[EMAIL PROTECTED] wrote:
On Fri, 18 Apr 2008 08:37:32 -0800, Mojo with Horan Company, LLC
[EMAIL PROTECTED] wrote:
If you care to use ping pong balls and the atlantic ocean as your medium,
you should be able to
On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED] wrote:
I just updated the branch. Wait about 5-10 minutes in case for the
changes to get mirrored, and then try updating and doing it again.
Looks better, no more soft lockup and ztcfg time is comparable to
On Fri, Apr 18, 2008 at 9:36 PM, Ex Vito [EMAIL PROTECTED] wrote:
On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
I just updated the branch. Wait about 5-10 minutes in case for the
changes to get mirrored, and then try updating and doing it again.
There's a constant discussion of commercial/biz hardware for Asterisk
here (we can still discuss Digium and Sangoma hardware, right?).
I think I would draw the -biz distinction on whether the OP was (a) on
topic and (b) solicited. If it is neither, then I'd be all for calling
it spam on a
Ex Vito wrote:
On Fri, Apr 18, 2008 at 9:36 PM, Ex Vito [EMAIL PROTECTED] wrote:
On Fri, Apr 18, 2008 at 8:20 PM, Matthew Fredrickson [EMAIL PROTECTED]
wrote:
I just updated the branch. Wait about 5-10 minutes in case for the
changes to get mirrored, and then try updating and doing
Wow Steve,
Don't ever stop posting links like this
It's stuff like this that makes mailing lists worthwhile
Femi
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve Totaro
Sent: 18 April 2008 17:45
To: Asterisk Users Mailing List - Non-Commercial
In article [EMAIL PROTECTED],
Roderick A. Anderson [EMAIL PROTECTED] wrote:
Just to clarify (and tie into Moj's response) if my last 'n' works out
to a priority of 20 I _do not_ need priorities 21 through 32.
That's true. Just remember the dialplan won't fall through a gap.
The only way to
I havent tried it. I have quite a few polycoms and didnt even know
polycom had this feature! :)
This is obviously a separate peice of software that must be purchased
and installed on the phones. Looks amazing though- any idea on
pricing?.
On Fri, 2008-04-18 at 14:53 -0400, Anciso, Roy wrote:
I actually just ordered 50 licenses to give this and the other
applications a try. I'll post my results to the list once I get them
and have had a chance to play around.
Regards,
- Brad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of faraz
Sent:
please do!. how much did the 50 cost you?
On Fri, 2008-04-18 at 18:22 -0400, Watkins, Bradley wrote:
I actually just ordered 50 licenses to give this and the other
applications a try. I'll post my results to the list once I get them
and have had a chance to play around.
Regards,
- Brad
Hi all.
Please, how can I configure an Asterisk PBX using an outbound proxy (that
resolve NAT Traversal)
I'm trying using the outboundproxy and outboundproxyport values in sip.conf but
the PBX don't get registered on the outbound proxy side.
I'm using SER + Asterisk with Jasomi outbound
Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Roderick A. Anderson [EMAIL PROTECTED] wrote:
Just to clarify (and tie into Moj's response) if my last 'n' works out
to a priority of 20 I _do not_ need priorities 21 through 32.
That's true. Just remember the dialplan won't fall
On Wed, 2008-04-16 at 19:44 -0400, Brian J. Murrell wrote:
I'm wondering if anyone has some extensions.conf dialplan using
Dial(..., L(...)) and the astdb to do lightweight prepaid service. I
only need to meter a handful of users.
Since I asked and nobody else answered (although I know you
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