Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI enterprise SIP with no password. Only thing i
changed is the part where i used regcontext.
on both boxes dundi.conf i have
[mapping]
priv
Hi all,
I would like to interconnect send and receive traffic to another SIP PBX.
The only thing I have is the IP of the other SIP PBX.
Actually, my Asterisk has an ISDN card so I can communicate with 3G
networks. I want to offer to the other PBX the ability to call a 3G devices
using my Asterisk
On Mon, 25 Aug 2008, Jonathan Disher wrote:
I am looking to replace the phone system at my father's shop with an
Asterisk box and some Cisco phones, but one piece of the
implementation is tripping me up. He has two buildings (the office,
and the shop proper), separated by about 3-400 yards.
Steve Alex thanks for your help. I've got it working perfectly now.
-Jon
- Original Message -
From: Alex Balashov [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Sunday, August 24, 2008 9:22 AM
Subject: Re:
Had an issue recently and it looked like there is a limit to the length of
a string... I built up a string dynamically and it seemed to get
truncated... Below is 2 lines of console output:
I've split this line to make sense:
-- Executing NoOp(IAX2/inco1-10969, About to dial
Ciao Loic,
Hello,
I am looking for a way to check if a call could be established with the
destination (SIP,IAX,ZAP).
So I thought about an application like DIAL but instead it should return
a variable and hangup immediately as soon as it gets something that
could lead to a valid
Jonathan Disher wrote:
He has two buildings (the office,
and the shop proper), separated by about 3-400 yards.
Your inter-building distance exceeds ethernet over copper limits, you
will need a fiber link.
paging intercom (to
page employees, etc) on a dedicated extension -
Easy to
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
A-Ha... That string is 256 characters long... Now there's a fishy number
if ever there was one.
So, if this a real limitation? This is 1.2.30 if that makes a
difference...
Did this limit go away in 1.4 ?
Yes, it did.
--
Tilghman
On Tue, Aug 26, 2008 at 9:01 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
A-Ha... That string is 256 characters long... Now there's a fishy number
if ever there was one.
So, if this a real limitation? This is 1.2.30 if that makes a
Jonathan Disher wrote:
I am looking to replace the phone system at my father's shop with an
Asterisk box and some Cisco phones, but one piece of the
implementation is tripping me up. He has two buildings (the office,
and the shop proper), separated by about 3-400 yards. Currently with
On 26/08/2008 Nhadie wrote:
is it possible to peer via IAX, then send SIP calls over the IAX peer?
yes indeed. The SIP calls would exist between the phones and the
Asterisk servers, and become IAX calls on the interconnect.
Tried insecure=very ?
Phil
On Tuesday 26 August 2008 08:06:10 Steve Totaro wrote:
On Tue, Aug 26, 2008 at 9:01 AM, Tilghman Lesher
[EMAIL PROTECTED] wrote:
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
A-Ha... That string is 256 characters long... Now there's a fishy number
if ever there was one.
On Tue, 26 Aug 2008, Tilghman Lesher wrote:
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
A-Ha... That string is 256 characters long... Now there's a fishy number
if ever there was one.
So, if this a real limitation? This is 1.2.30 if that makes a
difference...
Did this limit
On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote:
I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately,
the
shared_lastcall option is only in versions 1.6.0 and up.
Does anybody have a workaround for this in 1.4?
Or maybe a better question:
How stable is 1.6 for
On Tue, Aug 26, 2008 at 5:14 PM, Bob Pierce [EMAIL PROTECTED] wrote:
On Mon, 2008-08-25 at 17:47 -0500, Bob Pierce wrote:
I'm glad to hear that you're enjoying Asterisk 1.4! Unfortunately,
the
shared_lastcall option is only in versions 1.6.0 and up.
Does anybody have a workaround for
On Tue, Aug 26, 2008 at 9:22 AM, Drew Gibson [EMAIL PROTECTED] wrote:
Jonathan Disher wrote:
I am looking to replace the phone system at my father's shop with an
Asterisk box and some Cisco phones, but one piece of the
implementation is tripping me up. He has two buildings (the office,
and
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote:
I'd say - go for backport instead. shared_lastcall is commited in
http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985
and it seems that there are no bugfixes for it since. So, backporting
should be fairly simple.
Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users
and it works very well only in an intranet environment (no connections
to the PSTN world).
But in the near future, we have to plan a telephone system that works in
the intranet (voip) and also it must be connected to the PSTN
hi phil,
yup i have tried insecure=very as well. but still get forbidden.
if i use an IAX trunk, how do i dial a SIP user?
e.g. if i define this on iax.conf
[asterisk-iax-1]
type=peer
host=10.10.10.10
can i simply dial like this:
exten = _1X,1,Dial(SIP/[EMAIL PROTECTED])
nhadie
Phil
On Tue, Aug 26, 2008 at 5:39 PM, Bob Pierce [EMAIL PROTECTED] wrote:
On Tue, 2008-08-26 at 17:30 +0300, Atis Lezdins wrote:
I'd say - go for backport instead. shared_lastcall is commited in
http://svn.digium.com/view/asterisk/trunk/apps/app_queue.c?r1=86820r2=86985
and it seems that there are
Hi Shariq -
I m facing problem with TDM2400P pstn card. When someone dials, the voice
quality is crappyInstead of hearing.
Echo cancel almost works, but the callee hear what they describe as a
'background crackle/buzz' coming back when they talk.
Crackling noise is usually caused by an
On Tuesday 26 August 2008 09:08:58 Gordon Henderson wrote:
On Tue, 26 Aug 2008, Tilghman Lesher wrote:
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
A-Ha... That string is 256 characters long... Now there's a fishy number
if ever there was one.
So, if this a real
I have a TE410P in a Dell 2650 running in production on an older redhat
distribution. The various packages have gotten old and I can no longer
been able to build asterisk on this machine. I have prepaired another
2650 running SUSE Enterprise 10.1 sp2 (my workplace standard) to replace
it with.
After I make a call o n the Zaptel Card X100P FXO moduleit remains offhook
state as shown here...
Signalling Type: FXS Kewlstart
Radio: 0re2uk*CLI
Owner: None*CLI
Real: Nonek*CLI
Callwait: NoneI
Threeway: NoneI
Confno: -12uk*CLI
Propagated Conference: -1
Real in conference: 0
DSP:
On Tue, 2008-08-26 at 17:53 +0300, Atis Lezdins wrote:
Are there any plans to back port this feature into upcoming 1.4
releases?
No, new features are added only in trunk, and released in next major
release (1.6).
So what would be involved in back porting this feature for our system?
Hi Alejandro -
Dear all, now I'm using an Asterisk 1.4.13 SIP server with 50 SIP users
and it works very well only in an intranet environment (no connections
to the PSTN world).
But in the near future, we have to plan a telephone system that works in
the intranet (voip) and also it must be
Hi Andrea -
I have two asterisk servers, an IAX trunk between and some SIP users
registered
to each server.
The scenario is this: user A, registered to PBX 1, calls user B, registered to
PBX 2. Then A wants to transfer the call using the features.conf method (in my
case, **), but is
Gordon Henderson wrote:
On Tue, 26 Aug 2008, Tilghman Lesher wrote:
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
A-Ha... That string is 256 characters long... Now there's a fishy number
if ever there was one.
So, if this a real limitation? This is 1.2.30 if that
Ron,
What does the peers section in dundi.conf look like?
On Tue, Aug 26, 2008 at 3:00 AM, ronald ramos [EMAIL PROTECTED] wrote:
Would like to try setting up dundi with 3-4 asterisk.
But for poc, i would like to try setting up dundi on between 2 asterisk.
I copied the config from DUNDI
Hi Bruce,
my apologies, but the error was because of the key.
i just run keys init on the CLI and it works,
question on regcontext though, i set it to sipregistrations, how often does an
extension be added to the context sipregistrations and for how long will it
stay there? i'm looking at
On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote:
Do you have some sort of IP connectivity between the sites? 400
yards is a
too long for copper cat5, but can be done with fibre, wireless or
free-space optics... (which I don't personally recommend!)
The current plan is wireless bridge +
On Aug 26, 2008, at 5:34 AM, Chris Mason (Lists) wrote:
Jonathan Disher wrote:
He has two buildings (the office,
and the shop proper), separated by about 3-400 yards.
Your inter-building distance exceeds ethernet over copper limits, you
will need a fiber link.
Fiber would be great, if I
It is added when a phone registers, or re-registers. Depending on the
timing of the registrations and any restarts on the asterisk process
it may take some time for phones to re-register.
On Tue, Aug 26, 2008 at 2:10 PM, ronald ramos [EMAIL PROTECTED] wrote:
Hi Bruce,
my apologies, but the
On Tue, 26 Aug 2008, Jonathan Disher wrote:
On Aug 26, 2008, at 2:27 AM, Gordon Henderson wrote:
Do you have some sort of IP connectivity between the sites? 400
yards is a
too long for copper cat5, but can be done with fibre, wireless or
free-space optics... (which I don't personally
Some one already touched on this, but my guess is the Nortel system is
sending the page signal out to an actual paging system and the
speakers are in the remote building or the page port on the Nortel is
running over cat 3 copper to the other building. in either case tie it
in to the Asterisk
On Tue, 26 Aug 2008, Drew Gibson wrote:
Gordon Henderson wrote:
On Tue, 26 Aug 2008, Tilghman Lesher wrote:
On Tuesday 26 August 2008 06:49:54 Gordon Henderson wrote:
A-Ha... That string is 256 characters long... Now there's a fishy number
if ever there was one.
So, if this a real
On 26 Aug 2008, at 18:33, Drew Gibson wrote:
Is there a maximum string length for use with the legacy interface
chan_string?
Does it depend on the type of cup used? Does styrofoam give better
range
than paper?
regards,
Drew
DTMF modes include: as audio, tugging on the string correct
AAUG Meeting - Tuesday, August 26th at 7PM EDT August 26th, 2008
Here’s how to participate on Tuesday, August 26th at 7PM EDT.
The audio portion of the program will be available via a conference
bridge at 1-404-492-8060.
The shared desktop is available using a Java enabled browser at
On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote:
The shared desktop is available using a Java enabled browser at
???http://callin.xelatec.com/vnc??? with a password of ???aretta???.
Of course you must first have Zoiper installed and then add a new Zoiper
IAX account with Account
We've done a similar thing at a metal worker here by running xlite on
a pc set to auto answer and with the speaker out of the pc connected
to an amplifier which runs to the speakers. One way paging though.
Sorry for the top post, doesn't let me comment inline.
On 8/27/08, Bruce Reeves [EMAIL
Asterisks greatest strength is that it's a highly flexible platform that
let's you pretty much do anything.
It's downside, is that it's a highly flexible platform that let's you
pretty much do anything.
In other words, the quality of what you are trying to do depends on the
quality and volume
Hello everyone!
Sorry, if the whole task is silly, I'm new to this.
I have my newly installed asterisk (1.6.0-beta9) and my AVM Fritz a1 card. I
have a simple German isdn line and I have a microphone, headphones and a
running JACKd (JACK Aduio Connection Kit).
The question: Can I
Nhadie
Can you copy and paste your sip.conf settings for those two servers?? i think
there is a problem with your settings..
regards
Tarek Sawah
Date: Tue, 26 Aug 2008 09:00:52 +0800 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip peering
Does any have some good experience with the various freetds variants?
Is 0.64 better or worse than 0.82? I know that to use 0.82 you have to
use ODBC, since libtds.a is not long installed. Which is more stable?
I plan on using it for CDR, realtime and func_odbc. I'm connecting to
SQL
Tariq .. schrieb:
i think there is a problem with your settings..
Date: Tue, 26 Aug 2008 09:00:52 +0800 From: [EMAIL PROTECTED] To:
asterisk-users@lists.digium.com Subject: Re: [asterisk-users] sip peering
between 2 asterisk Hi Tariq, Tnx for your reply. Tried adding the
deny/permit
Hi All,
I received a request for a special application and need some guidance.
Cust has there own Asterisk PBX with SIP phones, pretty standard
setup.
They want an after hours application that checks inbound caller ID
numbers and matches them to a list, say 5 to 10 numbers of special VIP
Hey JR,
Is this a one VIP to one cell number match? Or is it on VIP to multiple cells?
On Tue, Aug 26, 2008 at 7:28 PM, JR Richardson [EMAIL PROTECTED] wrote:
Hi All,
I received a request for a special application and need some guidance.
Cust has there own Asterisk PBX with SIP phones,
Hi
as maximum link capacity could be calculated using codecs and channel types
so , regarding the CPU and processors load , Is there any formula or (any
relations could help ) that can give the maximum CPU load (mainly processor
and RAM ) or scalability average using asterisk channels ,
Jay R. Ashworth wrote:
On Tue, Aug 26, 2008 at 05:10:35PM -0400, Asterisk wrote:
The shared desktop is available using a Java enabled browser at
???http://callin.xelatec.com/vnc??? with a password of ???aretta???.
Of course you must first have Zoiper installed and then add a new Zoiper
On Tuesday 26 August 2008 19:28:17 JR Richardson wrote:
I received a request for a special application and need some guidance.
Cust has there own Asterisk PBX with SIP phones, pretty standard
setup.
They want an after hours application that checks inbound caller ID
numbers and matches them
Hi JR,
This may help you - we were using it to route calls from friends through the
IVR so they hit us directly. You'll have to modify it to suit your dialplan,
but it should be a good starting point.
http://www.voipphreak.ca/2006/11/26/asterisk-14-php-rolodex-howto-script/
Thanks,
Matt G
:
Is there a maximum string length for use with the legacy interface
chan_string?
Does it depend on the type of cup used? Does styrofoam give better range
than paper?
regards,
Drew
A lighter material for the cup will give better dynamic range than a
heavier one, at the
Asterisks greatest strength is that it's a highly flexible platform that
let's you pretty much do anything.
It's downside, is that it's a highly flexible platform that let's you
pretty much do anything.
In other words, the quality of what you are trying to do depends on the
quality and
Any pointers on this one?
--- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote:
From: Jay Ray [EMAIL PROTECTED]
Subject: [asterisk-users] X100P Card in OFFHOOK state
To: asterisk-users@lists.digium.com
Date: Tuesday, August 26, 2008, 12:24 PM
After I make a call o n the Zaptel Card X100P FXO
El mar, 26-08-2008 a las 19:46 -0700, Jay Ray escribió:
Any pointers on this one?
--- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote:
From: Jay Ray [EMAIL PROTECTED]
Subject: [asterisk-users] X100P Card in OFFHOOK state
To: asterisk-users@lists.digium.com
It would be clearer if it said Hookstate (FXS ports only): Offhook
i.e. the state information is not valid for FXO ports.
Jay Ray wrote:
Any pointers on this one?
--- On Tue, 8/26/08, Jay Ray [EMAIL PROTECTED] wrote:
From: Jay Ray [EMAIL PROTECTED]
Subject: [asterisk-users] X100P Card in
Hi ,
I am new user of asterisk.
here is my environment which is setup on Suse linux 10.0.
zaptel-1.4.11
libpri-1.4.7
asterisk-1.4.21.2
E1 Line.
and i have configured extension.conf,zapata.conf and able to make the
outgoing call from call files and originate command and incoming call also
I'll be that none of the other coffee makers can handle anywhere NEAR 60
voice channels, and don't get me started about HPEC!
http://www1.shopzilla.com/8N_-_cat_id--13050802__oid--680459759
___
-- Bandwidth and Colocation Provided by
58 matches
Mail list logo
