Maybe not, but I got it working good enough and time is scarce these
days so I didn't mess with it after that.
On 4/19/07, Stephen Bosch [EMAIL PROTECTED] wrote:
Brian McEntire wrote:
A follow-up with the solution in case anyone else is looking for this
answer:
I created two contexts in my
)
exten = s,n,Hangup
The r2 in the second context causes a distinctive ring. ALLPHONES is
defined above in my extensions.conf as ALLPHONES=ZAP/1ZAP/2
Hope that's useful to someone else. It works for me.
On 4/17/07, Brian McEntire [EMAIL PROTECTED] wrote:
Hello -
I have a TDM400P with 2 FXO and 2
the phone actually does ring).
Does anyone know why the r option might not be working? The
incoming channel is iax, and obviously the phone terminating the call
is Sip.
Thanks,
--Brian
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Hello -
I have a TDM400P with 2 FXO and 2 FXS modules.
Feeding the FXS modules are two VOIP lines which are terminated by
VOIP adapters and have regular RJ11 wires connecting to the FXS ports.
Since the two different VOIP lines have different phone numbers, and I
know and can tell asterisk
, Brian McEntire [EMAIL PROTECTED] wrote:
Hello -
I have a TDM400P with 2 FXO and 2 FXS modules.
Feeding the FXS modules are two VOIP lines which are terminated by
VOIP adapters and have regular RJ11 wires connecting to the FXS ports.
Since the two different VOIP lines have different phone
Hello,
I am using FastAGI and the AMI with Asterisk 1.2. Using the AMI I
originate a call to a person. I would like the person to be able to
press * to hangup or *9 to perform a certain function. I was thinking
I could make it hangup on * and then WAIT FOR DIGIT to see if 9 is
pressed.
Hello,
I am using FastAGI and the AMI with Asterisk 1.2. Using the AMI I
originate a call to a person. I would like the person to be able to
press * to hangup or *9 to perform a certain function. I was thinking
I could make it hangup on * and then WAIT FOR DIGIT to see if 9 is
pressed.
an option in
sip.conf to change how * handles not receiving an ACK? I know... it's
someone else's problem, but might help those of us stuck with buggy
implementations in production environments. :)
Brian.
On 4/12/07, Joao Pereira [EMAIL PROTECTED] wrote:
Hello
Thanks a lot for your reply.
Im now
and it would be nice to see
how many calls I have on each port.
Thanks for getting this going Brandon, I'll follow closely. We are heavy
Nagios and Cacti users here.
-Brian
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or
is confusing the Set line. I think the zap comes up Zap/2.
On 4/4/07, Bruce Reeves [EMAIL PROTECTED] wrote:
You have a syntax error.
exten = _#78,n,Set(DB(${DND/CALLERID
(num)})=1)
should read
exten = _#78,n,Set(DB(DND/ ${CALLERID
(num)})=1)
On 4/3/07, Brian McEntire [EMAIL PROTECTED] wrote
a vanilla install of the
latest stable asterisk 1.4.
On 4/4/07, Gordon Henderson [EMAIL PROTECTED] wrote:
On Wed, 4 Apr 2007, Brian McEntire wrote:
Don't think that was it unless I still have a typo. Here's my line
from extensions.conf:
exten = _#78,n,Set(DB(DND/${CALLERID(num)})=1
Hello -
I've read Asterisk should be able to activate a do not disturb feature
to turn off the ringers on extensions. I checked the wiki and can't
find documentation for how to do it.
Here's my attempt, added to extensions.conf:
[dnd-on]
exten = _#78,1,Answer
exten = _#78,n,Wait(1)
exten =
Ahh... got it now. Thanks for all the replies.
I was thinking that it was a function that was already built in, but
I see by setting a value and then testing it before ringing
extensions, it's easily added to the dialplan.
On 4/3/07, Philipp Kempgen [EMAIL PROTECTED] wrote:
Bruce Reeves
()
- - -
It appears to me that Set(DB ... as a function isn't working, isn't
built in, or needs more information.
I saw something about GLOBAL variables, perhaps I can use those instead?
On 4/3/07, Doug Lytle [EMAIL PROTECTED] wrote:
Brian McEntire wrote:
Hello -
I've read Asterisk should
Jordan Novak wrote:
Okay, I get it. I still have a problem though. I have no way to wire 30%
of these end-points. P{hysically impossible. They do have cat3 twisted
pair to each phone. But of course they want IP. Are there any adpaters
that will give me just enough bandwidth to get it done. The
of work-around can be applied to digital cards but
it seemed to work for analog cards. However, using genzaptelconf was very
strait forward.
-Brian
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Mail list wrote:
Hello
Anyone here uses Vitelity as voip provider ? Their pplans looks good
but i need some feedback from existing customers if any here .
I would like to express an opposite opinion.
I have two accounts with them with lots of DIDs. Everything works fine,
and they
Hi all -
Been using Asterisk installed on Debian and love it. But it's time to
rearrange some lines and looking for a few features I didn't enable or
have in the dial plan the first time around and wondering if you would
recommend doing it through configs again or if one of the prepackaged
I would like to use AsteriskNow with a T1 PRI. I know that AsteriskNow will
not auto-detect the digital cards but is there a way to manually configure
AsteriskNow so it will be able to see and use the trunks?
Thanks,
-Brian
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need to somehow force the card into the configuration.
Is there a way to manually configure the system so that ports that the gui
normally could not detect would be available?
Thanks again,
-Brian
-Original Message-
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
Sent: Friday, March 23
Stephen Bosch wrote:
Bill Hackensack wrote:
On 3/22/07, [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]*
[EMAIL PROTECTED] mailto:[EMAIL PROTECTED] wrote:
Hello
P.S The program that I am using is open source, of course
( www.phpsurveyor.org http://www.phpsurveyor.org)!
What part of
shadowym wrote:
If I can't be confident enough in an important source of information
like this then I can't be confident enough to provide an Asterisk
solution to businesses. That's the way I see it. Yea, it's a wiki but
it's the best source of info out there.
Suggestion: switch to
Chris Mason (Lists) wrote:
Here's how I do it.
Buy complete fanless system flash card ready unit with four ethernet
interfaces:
http://www.ibt.ca/v2/items/fwa7204/index.html
It is very small, in an aluminium extrusion case, very robust.
What kind of money are those things? There doesn't
Davis Sylvester III wrote:
Is there a way to view the entire dialplan when using Realtime?
I use Realtime and MySQL connector.
If you mean the contents of .conf-file based merged with whatever the
Realtime engine is supplying, I don't think there's a way of seeing both
together.
But you
Davis Sylvester III wrote:
Brian Capouch wrote:
Davis Sylvester III wrote:
Is there a way to view the entire dialplan when using Realtime?
I use Realtime and MySQL connector.
If you mean the contents of .conf-file based merged with whatever the
Realtime engine is supplying, I don't
Mike Hammett wrote:
[Mar 7 14:12:37] DEBUG[4380]: res_config_mysql.c:623 mysql_reconnect: MySQL
RealTime: Successfully connected to database.
[Mar 7 14:12:37] NOTICE[4380]: config.c:1174 ast_config_engine_register:
Registered Config Engine mysql
[Mar 7 14:12:37] NOTICE[4380]: config.c:1174
Mike Hammett wrote:
Could someone provide some steps for troubleshooting Realtime? I can’t
see any signs that it’s working. I followed and double-checked a few
different guides around the net, but haven’t been able to figure it out.
You don't say which version you're running.
I *think* the
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with older version 1.2.3
Anyone out there running it against 1.4.0?
It built just fine for me, but then it crashes the server when I try to
run it.
Michiel van Baak wrote:
On 17:53, Sun 25 Feb 07, Brian Capouch wrote:
Supa wrote:
Try this:
http://www.bayhamsystems.com/asterisk.html
Works for me just fine, and it is very easy to get up and running, even
with older version 1.2.3
Anyone out there running it against 1.4.0?
It built
Pavel Jezek wrote:
it can be also usefull to use 'limitonpeer' option in sip.conf
with this, it would not be needed to define separate type=user and
type=peer for each phone,
instead define one type=friend and apply limitonpeers=yes
;limitonpeers=no; Apply all call limits
Tim Connolly wrote:
You can buy smartnet on a single phone for something like $8 a year.
This will get you in legally.
Any idea about how specifically to get such a contract? It is rumored
to be pretty tricky.
B.
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Hi list,
We are looking for an Asterisk consultant for a 3 months mission in Paris. If
you are interested, please contact us at [EMAIL PROTECTED]
Best regards,
Brian
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Stephen Bosch wrote:
Have you tried calling ATT and asking for call disconnect supervision?
I realise that this can be a thankless and tedious endeavour, but it IS
worth trying. There are almost no commercial switches that don't support
this; it's a matter of activating it for the specific
Stephen Bosch wrote:
And then he hung up on me.
...wow.
This society is doomed.
Actually, it isn't so much society as the legacy telcos.
But unfortunately, they've been pretty smart about using the billions
that they've stolen from us over the years: they use a lot of it to line
the
Stephen Bosch wrote:
And use a different Wiki engine! Augh! (Mediawiki, anyone?)
Who runs voip-info.org?
I'll bet if you volunteered to take it over, the folks who run it would
gladly let you have it
And I'd further bet they'd gladly let you run whichever Wiki software
you want!!
Yuan Liu wrote:
My multiple postings to this list this morning got garbled in
http://lists.digium.com/pipermail/asterisk-users/, and don't come back from
list. (e.g.,
http://lists.digium.com/pipermail/asterisk-users/2007-February/179315.html) I
thought it was Hotmail, so I saved one
?
Thanks,
Brian Arlinghaus
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somewhere but I can't find it.
I am curious to learn more about your solution. If you post more information
I might be able to help you with your RD. In any event thanks for posting
up and in advance for keeping us posted on your progress.
-Brian
_
From: [EMAIL PROTECTED]
[mailto
Ack. That should be I am using analog for the proof of concept but plan to
use PRI for the actual system.
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Brian K.
Alexander,Jr. (Vision Point Systems)
Sent: Tuesday, February 06, 2007 8:44 AM
To: 'Asterisk Users Mailing
I haven't quite gotten this working yet but I am going to update the thread
with what I have learned. Maybe this will help the next guy who tries to
figure this out.
The trick to using the DIALSTATUS seems to be to put it in the handler for
the h (hang-up extension).
[outdialer]
not get it to work. I posted a topic (Using
Local Channels with originate) to this list yesterday with the details about
what I had tried. Maybe you will see what I missed.
-Brian
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael
Collins
Sent: Thursday
Benko,
You can put multiple files in the MOH directory giving your listener a good
chance of getting a new piece of music each time he is on hold. Asterisk
picks one of your files randomly.
Regards,
Brian
- Original Message -
From: Benko [EMAIL PROTECTED]
To: asterisk-users
The notdeadyet.py script never runs. The ${DIALSTATUS} passed into
showstatus.py is empty. I don't understand what I did wrong.
Thanks in advance for your help. I am stumped by this.'
-Brian
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Digium.
Brian
On 1/30/07, Mark Spencer [EMAIL PROTECTED] wrote:
Many of you may have seen the recent announcement about Danny Windham
coming on as the new CEO of Digium. This is one of the most exciting
things to happen to Digium and to Asterisk at large. When Danny comes on
board, I
Julian Lyndon-Smith wrote:
I cannot seem to find any reference to labels in realtime extensions -
using 1.4.
I've googled until my eyes have bled, and also scoured voip-info.org.
Is there anything that helps me here ?
You have to have numbered priorities with realtime.
This is because (as
Jon Pounder wrote:
Buzzwrong answer! Don't answer on things you have no idea. and
stop providing bad information.
you should take your own advice - an acre is 200ft x 200ft - what idiot
would
pay a consultant $7000 to tell them they need one access point in the
middle.
This is
Al Bochter wrote:
Matthew
I agree. I only know what I have told by others so I do need this input
I have been told that Digum G729 is a big pain the the butt to get
working with Asterisk
and it is very hard on the CPU
Keep in mind I have never used any Ver. of G 729
So tell me what you
a 24port switch or not.
-Brian
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Steve Underwood wrote:
Paul Hales wrote:
When you built Asterisk, it must have refused to build the ilbc codec -
I have never seen an Asterisk box that could not transcode ilbc, in over
3 years of working with Asterisk.
Most versions of embedded Asterisk will choke unto unusable if they are
Folks, with all due respect: this thread is now wy off topic, as it
has nothing to do with Asterisk whatsoever.
Please take it offline, or to ~biz.
thx.
B.
--
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dangerous content by MailScanner, and is
believed to be clean.
to grab a compiled copy off of another box.
Only needs vmail.cgi and a couple of supporting graphics.
-Brian
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at the other Polycom Phones and they
still show me as Online. (Yes, I have bw set to 1.)
Does that mean that DND doesn't show up as a hint either?
-Brian
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Lacy Moore - Aspendora wrote:
I think that's the problem with the Asterisk community right now.
Anytime something is suggested, the response is either write it yourself
or deal with what is there.
Do you have experience with other big, complex Open Source projects? Do
you know of any
You need to make sure that you install the asterisk-config package as well.
--Brian
On 11/16/06, blackwater dev [EMAIL PROTECTED] wrote:
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got
You need to make sure that you install the asterisk-config package as well.
--Brian
On 11/16/06, blackwater dev [EMAIL PROTECTED] wrote:
I have an Ubuntu system and went into Synaptic and checked asterisk for
installation. Once installed, I started it with /usr/sbin/asterisk -vvvgc
and got
option worked fine.
Hope this helps,
-Brian
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There are many none free area codes in the 8xx space. Just google area
codes.
.Brian
Doug Crompton wrote:
Ok so ONLY 900 numbers are pay.
Next question 18XX numbers. are they all toll free? Is there any
space in 8xx that is used otherwise?
Doug
On Tue, 14 Nov 2006
On Mon, Nov 13, 2006 at 12:46:14PM +0100, nik600 wrote:
Hi
i have an application developed with bayonne.
Recentely i'm experiencing some problems and i am planning to migrate
to asterisk.
I would like to know if i can do these things whit asterisk:
- IVR integration with database
You can just seperate multiple phones with in the Dial command,
as the voip-info wiki page shows:
http://www.voip-info.org/wiki/view/Asterisk+cmd+Dial
On Tue, Oct 31, 2006 at 10:28:32AM -0700, Stephen Bosch wrote:
Hi, folks:
I need to be able to have a single DID ring multiple remote (IP
After John Todd's talk at Astricon about the ISN project, I spent much
of the weekend playing around with it.
I have discovered that the default dialplans on my Sipura gear, as well
as my Grandstream phones, intercept the * key that is a required part
of ISN numbers and interpret it as a
and sub-contexts to clean this up. An example is
shown below. Each new local user you add just needs a single entry under the
[extensions] section, plus an entry in their specific channel (e.g.
sip.conf). This is reasonably easy to manage.
HTH,
Brian.
- extensions.conf
want to
allow outbound calls. For example:
[internal]
exten = _9.,1,Dial(SIP/${EXTEN:[EMAIL PROTECTED],15,r)
exten = _9.,2,Congestion()
exten = _9.,102,Congestion()
This will match all numbers which begin with 9, and route them via sipgate,
stripping off the leading 9.
Regards,
Brian.
P.S. All
I have installed 1.2.12.1 http://1.2.12.1 in FC5 with libpri.1.2.4
when i start
iam getting the following error and it quits
== Registered channel type 'Local' (Local Proxy Channel Driver)
[chan_zap.so]Oct 23 16:16:07 WARNING[11084]: loader.c:325
__load_resource: libpri.so.1.0:
is Asterisk: The Future of Telephony,
downloadable for free from www.asteriskdocs.org. The chapters on dialplans
(ch 5 and 6) should tell you what you need.
Regards,
Brian.
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On Wed, Oct 25, 2006 at 11:37:35AM -0700, Tielin Xu wrote:
I have been testing Xlite 2.0 and 3.0. The Xlite 2.0 is slow on
initiate time, but I can add second sip proxy account, which is very
critical to my testing. I installed Xlite 3.0, which I could not add
second account on SIP account
Andrew Joakimsen wrote:
Another thing is my understanding of the peer, user and friend. I
thought that a peer can only receive calls from either a user or a
friend, a user sends calls to a peer or friend and a friend is both a
peer and a user, however in my production machine I have the
nothing to stop Digium licensing this code
commercially from him, if it adds value to the product.
Regards,
Brian.
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Bloggs. Many SIP phones will display the
value given by the far end when an incoming call arrives.
HTH,
Brian.
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On Mon, Oct 23, 2006 at 02:32:55PM +0300, Giedrius Augys wrote:
Hi,
I have problem installing spandsp-0.0.3pre24 on FreeBSD 6.1. I get
error: configure: error: Can't build without libtiff . But I have
installed tiff from port tiff-3.8.2. I understand that the problem is
On 10/23/06, Unmetered Pipe [EMAIL PROTECTED] wrote:
I don't mean to be a troll in any way shape or form. I was on IRC last night and I observed the following convo. below. What do you guys make of it ?
That you are a troll?
-Brian
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which
Asterisk routes outwards via a POTS card to be prefixed with 55, you can do
that in your dialplan (extensions.conf) by modifying the rule for placing
outbound calls.
[internal]
exten = _1.,1,Dial(Zap/4/55${EXTEN})
^^
Brian
On Fri, Oct 20, 2006 at 04:46:52PM +0300, [EMAIL PROTECTED] wrote:
Let me put is clear. I'm using TRIXBOX which many out here feel - its
for kids- but i do like it an use it for sometime.
As all i do have extensions and trunks configured on it. I want one of
my extensions to use a
Kristian Kielhofner wrote:
Administrator TOOTAI wrote:
Cory Andrews wrote:
I caught a thread the other day concerning Astricon and users embedding
Asterisk on a Linksys or Netgear broadband router. I lost track of the
email thread, if anyone is presently working with this scenario
please
Just an FYI to anyone out there who will be attending Astricon and who
would like to play around with embedded Asterisk on the Netgear WGT634U
platform.
If you want to bring your own to the show, I'll be bringing all the
appropriate stuff to flash them there with my latest openWGT/Asterisk
think # here is just a shorthand for number, and * is short for
asterisk
I can see how using those symbols in this context is pretty confusing -
perhaps you could correct the entry on the Wiki.
Regards,
Brian.
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, tip or cu will let you talk to it. If it's asterisk, then use the
asterisk console (asterisk -r)
HTH,
Brian.
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for setting up chan_sip
Brian.
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On Wed, Oct 18, 2006 at 09:11:15AM -0400, Matt wrote:
In the case of you example the IAX2 registration came in from the source
port on the far device of 1207.
Connections don't just move between ports.
I understand all this. However, here is my question.
MY on 4569 OTHER SIDE 1027.
Kristian Kielhofner wrote:
Do you have the necessary components for a serial cable for these
little guys? I would like to play with the loader and get a serial
console...
If you don't have one perhaps we can work on getting the parts
before then.
I have one, and also will be
172.31.131.189.62505 69.16.138.164.21 ESTABLISHED
^ ^^
Regards,
Brian.
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the mobile network.
Would I need to load some sort of app onto the phone?
Thanks,
Brian.
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or response.
Brian GreulTexas Shirt CompanySolutions To Promote
Youwww.txshirts.com713-802-0369 / 713-861-6261
(fax)ASI/343253
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michelle
DupuisSent: Tuesday, October 17, 2006 11:07 AMTo:
[EMAIL PROTECTED]; 'Asterisk Users Mailing
state. The
original socket sockfd is unaffected by this call.
Regards,
Brian.
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Scott Higginbotham wrote:
I'm interesting in testing this.
OFF LIST PLEASE, FOLKS!!
The list has enough traffic without the 10,000 me too mails that are
likely to follow if nobody points out that it's bad netiquette.
B.
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want to route to your internal
extensions using this number, or by stripping off the first 9 digits, so
that everyone gets their own DDI for free!
Regards,
Brian.
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= freedestinations
Check the wiki:
http://www.voip-info.org/wiki/index.php?page=Asterisk+config+extensions.conf+sorting
Regards,
Brian.
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On Mon, Oct 16, 2006 at 04:47:31PM +0200, Giorgio Incantalupo wrote:
Hi,
every second I get on the console:
Remote UNIX connection
Remote UNIX disconnected
which gives no problem but makes console unusable.
Is there anybody who has encountered the same problem? How did you solve it?
Have
Jay R. Ashworth wrote:
On Mon, Oct 16, 2006 at 10:21:29PM -0400, C F wrote:
On 10/16/06, VaibhaV Sharma [EMAIL PROTECTED] wrote:
I don't think this is a problem because of the snow storm.
Yo, all; please take this thread where it belongs, which is the -biz list.
It is not relevant to the
Matt wrote:
Ok I understand all that... Just wanted to confirm that A) it was the
remote router mangeling the port and B) that it wouldn't cause an
issue (I wasn't 100% sure if it would.. since only the 4569 port is
open on the firewall).
Could this cause an issue? If only 4569 is open on the
authentication.
Regards,
Brian.
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On Sat, Oct 14, 2006 at 09:34:33AM +1000, Paul Hales wrote:
The Zulty's 4x5 does (or did) fwiw.
Thanks. voipon.co.uk has them for GBP 299 or $557.64 (gulp)
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To
address.
Regards,
Brian.
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contained within the SIP/SDP
packets and uses the source IP address it sees (x.x.x.x) instead.
HTH,
Brian.
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they no longer stock it. (Not sure if that
means it has been discontinued, or it is unpopular). They also said they
have no other phone which is both VoIP and PSTN.
Regards,
Brian.
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. Enter ekiga into google. That gives you www.gnomemeeting.org as the
web site.
2. Click the downloads link.
3. Scroll down the page to sources and click there.
This then gives you:
ekiga-2.0.3.tar.gz
opal-2.2.3.tar.gz
pwlib-1.10.2.tar.gz
HTH,
Brian
Attempting native bridge makes me think of this second
possibility. If so, you could try setting nat=yes and/or canreinvite=no on
the channel to the SIP provider, so that Asterisk proxies the RTP data.
Regards,
Brian.
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Douglas Garstang wrote:
I don't get it. The clients are ok with their phone systems being down anywhere
from minutes to hours?
Try googling for cost benefit. I got 135 million hits.
Your brain has some very odd twists in the way it works.
Or you're a troll.
B.
/$FILE/SIPNATtraversal.pdf
* http://siprouter.onsip.org/doc/gettingstarted/ch04s05.html
HTH,
Brian.
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it
a chance to unregister itself, it will eventually be unregistered due to the
timeout.
(Additionally, some half-baked SIP NAT solutions require you to set a
ludicrously short registration timeout, e.g. 20 seconds, just to keep UDP
state open on the firewall)
Regards,
Brian
On Wed, Oct 11, 2006 at 04:06:06AM -0400, Ahmed Ndaula wrote:
I am absolutely new to asterisk for the Voice Over IP. I have set up my
own server using asterisk, successfully connected and be in position to
test the voice over IP by connecting to the digium server and testing
the echo system
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