Something along the lines of show application retrydial ?
[EMAIL PROTECTED] wrote:
I am looking for that feature to implement on Asterisk as well.
does anyone know how to implement it/
Thanks!
- Original Message - From: Jon Farmer [EMAIL PROTECTED]
To: Asterisk Users Mailing List -
John Novack wrote:
Mike Garey wrote:
well, the problem isn't that the card doesn't detect a disconnect,
it's that it doesn't detect it immediately (or at least within a short
period).
Odds are that is the telco, and not the Sangoma or Digium card. That is
quite normal for a 10-30 second
Are you in the USA or Canada?
Mike Garey wrote:
yes, I'm using kewlstart
On 4/24/06, Sean Cook [EMAIL PROTECTED] wrote:
On Mon, 2006-04-24 at 17:20 -0400, Mike Garey wrote:
As far as I can tell, after discussing this matter with other asterisk
users in my area, my telco _does_ provide
Benoit Panizzon wrote:
Hi all
I noticed that most caller are quite confused by the standard voicemail
announcement text. Especialy as the number read is the 'internal' number.
Callers often hang up because they think having called the wrong number when
they hear the announcement.
Is there
Davi-Ann wrote:
When I set asterisk to to sequence the lines as Ground Start the system
is not starting. It is giving the following error Invalid Argument 22
Do you have any ideas about this.
Any help or assistance appreciated.
I don't think Digium's analog cards support Ground Start..
--
how about, in sip.conf, [EMAIL PROTECTED] in
the [section] for that device?
Bill Gibbs wrote:
Put your voicemailbox number (usually extension) in the 1.subscribe field.
Bill
From: [EMAIL PROTECTED] on behalf of Kerry Garrison
Sent: Thu 4/20/2006 7:32 PM
Kristian Kielhofner wrote:
Steve Kennedy wrote:
Is there a list of sounds (base - as with Asterisk itself, and
additional) for the 1.2 release. As in a list with what the content of
each file is.
There's a list for 1.0.7 on the wiki, but that seems woefully out of
date.
Any help appreciated.
Jon-o Addleman wrote:
Is there a way to have an extension match on a sip address? I've tried
the obvious - [EMAIL PROTECTED] but it seems to behave just like _. which is no
good.
Is there a better way?
. stops a pattern match.
--
Now accepting new clients in Birmingham, Atlanta,
Roger Schreiter wrote:
Alex Mosburger schrieb:
...
It is not my end hearing or producing echo. My voice is heard correctly
without any echo, but the other side hears his OWN voice several msec
...
Yes, this is, what I meant.
The other's voice is fed back by your device and running
back to the
John Novack wrote:
Eric ManxPower Wieling wrote:
John Novack wrote:
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between
some SIP phones and asterisk, I have personal experience with
Polycom phones not being able to do a blind xfer using the feature key
Damon Estep wrote:
Is the current release different than what I am running, # transfer on
my systems are all blind, no attended option.
1.0.x only supported blind DTMF transfer hack. 1.2.x supports both
blind and supervised DTMF transfer hacks. See features.conf in 1.2.x
Andre Courchesne - Consultant wrote:
Ok,
Here is what I got working:
A call comes in from a Zap line. 5 SIP extension ring if nobody
picks up, the call is transfered to a cell phone number. That works.
I not want to add a playback of a file (Please waite while you are
being
John Novack wrote:
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between some
SIP phones and asterisk, I have personal experience with Polycom
phones not being able to do a blind xfer using the feature key.
Our receptionist does both blind and attended
What happens if you remove the r option? r is almost NEVER useful.
Steve Feinstein wrote:
I've been pulling my hair out over this one trying to understand it.
If you have a very simple extension:
exten = 1,n,Dial(IAX2/Steve|24|r)
Everything I've seen says this should tell the IAX phone (our
Damon Estep wrote:
There is some kind of issue with SIP transfer interaction between some
SIP phones and asterisk, I have personal experience with Polycom phones
not being able to do a blind xfer using the feature key.
We have to use the asterisk # blind xfrer functionality for blind
transfers
Peter J Dean wrote:
I have an issue with trying to ensure that when dialling an extension
that it continues to ring up to the timeout value. But what I am finding
is that the timeout is all over the place. Sometimes half the timeout
value and other times within a few seconds of the timeout
Anton Krall wrote:
I will try that and see what happens...
This server is a supermicro one.. Anybody else had issues like this on
supermicro? Any hints on how to resolv them?
If I remember correctly, supermicro bios does let you assign irq to certain
pci ports right? Will that help?
Also, is
Many multi-line phones allow you to use the same username/password for
all lines. Then the phone only actually registers once using that
username and password, not once for each line.
What we do with the Polycoms is configure each line to register as a
different username/password (we use the
[EMAIL PROTECTED] wrote:
I'm having problems enabling stutter dialtone for users connected to
channel banks.
Half of our users are on iaxy's and the other half are connecting to
channel banks. The users on ixay's are getting the stutter dialtone
on new voicemails, but the ones on the channel
Michael Strelnikov wrote:
Hi,
Is it possible to force using codec depends on extension? For example,
voice codec is ILBC and with some prefix fax code should be ulaw.
[EMAIL PROTECTED] asterisk]# grep CODEC asterisk-1.2/doc/*
asterisk-1.2/doc/README.variables:${SIP_CODEC} Set the SIP codec
The only thing registration does is inform Asterisk about what IP the
device is at. It has nothing at all to do with Device - Asterisk
calls. Registration only affects Asterisk - Device calls. In a Device
- Asterisk call, Asterisk does not care what IP the device is at as
long as the
Julian Lyndon-Smith wrote:
Asterisk SVN-trunk-r7353M
I have a EuroISDN line. I am sometimes out of the office so I get my
extension to ring both my mobile and desk top (7960) phone at the same
time.
This all works just peachy. However, I have a question regarding
callerid. Is there any way
We reboot all our Asterisk servers once per week if they have a TDM400P
in them. If we don't do that, then the TDM400P modules stop working.
Lorentz Hinrichsen wrote:
at the risk of starting a flame war:
I stand corrected on the pricing, however I also stand behind my observation
about the
Bryan Mahin wrote:
Hello all,
I am looking for a way to restrict users from logging in two separate
phones with the same authorization name/password at the same time.
Meaning, I only want users to be able to place a call from one phone in
one location, but have the ability to move from computer
Andrew Kohlsmith wrote:
On Wednesday 05 April 2006 16:42, Jon Weisman wrote:
And in Extensions.conf
exten=_X.,1,Prefix(${ACCOUNTCODE})
exten=_X.,2,Dial,Zap/g1/${EXTEN}
That won't work for this case, as he needs to enter the access code *after*
dialing. Right offhand, I can't think of doing
All ITSPs suck. However, in my experience, Teliax seems to suck less
than most.
Lonnie Abelbeck wrote:
asterisk at anime.net writes:
On Thu, 30 Mar 2006, Giridhar Reddy Bandi wrote:
I am looking at purchasing some DID lines from Teliax to install it on my
asterisk.
i would like to know
Shad Mortazavi wrote:
What I would like to do is to redirect external SIP calls to our
external Asterisk server. e.g if I call sip:[EMAIL PROTECTED] I would
like the call to be routed from our Internal Asterisk server to our
External Asterisk server via IAX2 and for the external asterisk
Bjørn O wrote:
This is really the day for new experiences – sorry for the load on the
mailing list, but this will be the last issue I try to solve before I take
the night off;)
So I’ve got the incoming calls to work with a not-so-well solution (could
therefore still need some feedback on
It's ${EXTEN} NOT {EXTEN}
Mar 29 17:44:18 NOTICE[11502]: chan_iax2.c:6794 socket_read: Rejected
connect attempt from iax.providers.server.net, who was trying to reach
'{EXTEN}@'
Tx-Frame Retry[000] -- OSeqno: 000 ISeqno: 001 Type: IAX Subclass:
REJECT
Rene Nelson wrote:
Is this card supported in the Open source version of Asterisk? If so has
anyone had success implementing one? Any help will be greatly appreciated.
The list of supported hardware is at http://www.asterisk.org/hardware
If your card is on the list you will need to call
Is there an account limit for voicemail? I have 80+ users in
the voicemail and I can only reach the 70-ieth user. If there is a
limit how can I increase it to hundred for example?
I've only seen this with something like Voicemail(123124125126)
(i.e. leaving the same voicemail in
It does? All this time I thought that permit= and deny= is what limited
access! Check the docs, host= is for OUTGOING, permit/deny is for INCOMING.
Mojo with Horan Company, LLC wrote:
meaning, when you put
host=dynamic
in sip.conf, it doesn't matter what ip the client comes from.
if you put
3Com is one of the few that lie about it. Many Cisco phones support
SIP, but not all of them. I think Nortel also lies about SIP on some of
their phones.
Daniel Hazelbaker wrote:
Drat, because the 3Com phones looked pretty good for the price. :) Is
there somewhere that has a compatibility
Nick Hoffman wrote:
On Sat March 25 2006 18:06, Nick Hoffman [EMAIL PROTECTED]
wrote:
Hi guys, I've been using a Polycom IP 301 for a couple of weeks now
and find that it's extremely slow for configuring. For instance, it
takes several minutes to boot up, apply any changes via the web
interface
Turn on message threading in your email client and you'll see just how
wrong this is.
Douglas Garstang wrote:
Well, actually I did... sort of... I picked a random post and hit reply
as I normally do. I forgot to clear the old text, but it's obvious that
in no way it detracted from the other
[EMAIL PROTECTED] named]# grep ANI /home/software/asterisk/asterisk-1.2/doc/*
/home/software/asterisk/asterisk-1.2/doc/README.variables:${CALLERANI}
* Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)})
/home/software/asterisk/asterisk-1.2/doc/README.variables:${CALLINGANI2}
I know the answer. The answer is NO! Asterisk does not support
changing the codec during a call. It also does not support changing the
codec on an INCOMING call. Of course, as you know by reading
README.variables, SIP_CODEC can force a specific codec on an OUTGOING call.
Wai Wu wrote:
setpriority(0, 0, 20);
This is for Perl, of course.
Benny Amorsen wrote:
RS == Roger Schreiter [EMAIL PROTECTED] writes:
RS Is there any mean to let AGI scripts run in a lower priority
RS (except starting a new shell from the a short initial AGI script)?
You can start the script with renice
For one thing, don't use the r option to dial. It can hide major
problems. If you don't hear ringing without using r then you have
massive problems.
Asterisk wrote:
Nope,
It's not a firewall problem.
I have a Juniper/Netscreen firewall with SIP NAT Traversal etc.
It replaces the inside IP
show applications in the Asterisk CLI will list the applications
availble. show application whatever will give detailed docs on that
application. Also look in the docs directory of your Asterisk source
code tree for much more documentation. Then if all else fails, look in
the Wiki and if
Asterisk only supports echocancel on PSTN (zap) interfaces. Echo needs
to be canceled out at the PSTN-VoIP interface. Calls that are all
VoIP can't have echo. If they have echo then you can, with confidence,
blame the SIP phone.
Here's another example:
Analog Phone - SIP ATA - Asterisk -
Derek Whitten wrote:
Charles Marcus wrote:
Whether or not a forum is a better idea isn't really depending on the
subject matter IMHO. Its success or failure depends on what the
prospective participants like better. I personally cannot stand
forums. That's a place where I have to expend energy
Bob McDowell wrote:
My PocketPC with ppcIAX and/or SJPhone behaves in exactly the same way.
The only resolution is to use an earbud... I'm guessing that the
server's echo cancelling is intended to cancel minor echo introduced by
the path, but doesn't handle 'real' echo caused by looping sound.
Kenige Ho wrote:
Dear All,
I am currently have this problem in which I am sending call out from the
Zaptel TE405 to a VoIP gateway. But the problem that the call over to the
VoIP Gateway will always have a fake ring tone. Can you please give some
pointer how to fix this problem?
Don't use
This is a known (but very old and fixed) bug. See
http://bugs.digium.com/view.php?id=4186
bails wrote:
Hi whatever I set the span line to in zaptel.conf
ie span=1,0,0,ccs,hdb3,crc4
span=1,1,0,ccs,hdb3,crc4
span=1,2,0,ccs,hdb3,crc4
zttool always shows
Sync Source: Internally
Steve Kennedy wrote:
On Mon, Mar 13, 2006 at 07:38:01PM -0500, Watkins, Bradley wrote:
That depends on what you mean by default. The supplied sample
extensions.conf contains the priorityjumping=no by default, but if this
parameter is absent then the default is to jump n+101.
OK, that
On Thu, Mar 02, 2006 at 04:14:34PM -0700, Douglas Garstang wrote:
The best way to achieve maximum manageability is to design a MySQL database and
develop AGI scripts (in your language of choice) that work to that design. I've
found that it has been far easier to develop complex routing logic
Dumpolid Exeplish wrote:
i am taking overr the administration of an existing production * PBX but i
cant seem to find out which version of * this is. When i use the 'show
version' coomandat the cli, i get this:
Asterisk CVS-HEAD built by [EMAIL PROTECTED] on a i686 running Linux on
2005-08-10
Michiel van Baak wrote:
On 13:02, Wed 01 Mar 06, Arne Morten Johansen wrote:
Hi there.
Is it possible to have different sip users have the same CallerId number
in sip.conf.
I need this because we got multiple companies on this Asterisk box.
Company A's internal numbers:
CID: User:
1000 -
Jesse Guardiani wrote:
Hello,
What is the best way to ignore a DID and not pick up the line?
I don't want to incur charges on the line (T1 PRI), so would
Hangup pick up the line, then hang up? Or can I use Hangup?
Use the Congestion application.
___
ADEGOKE ARUNA wrote:
I need your help
I have a sangoma A104D on my dell server; I got card status ok with no alarm
If I dialed the extension 6210006, it shows the output as stated below, but
there is no ringing from the pstn number nor the iax softphone am using on
my pc.
I will be glad
Anton Krall wrote:
Guys.
I have about 20 Polycom 301, some 501 and some 600 and I really like the
phones, but I have a question and maybe somebody else has seen this. Seems
sometimes when people talk a bit loud, Polycom phones have a tiny bit of
echo, can this be controled with some kind of
txgain for Asterisk.
Anton Krall wrote:
Anyway the phone can compensate? I don't think it works that way but worth
asking..
|-Original Message-
|From: [EMAIL PROTECTED]
|[mailto:[EMAIL PROTECTED] On Behalf Of
|Eric ManxPower Wieling
|Sent: Tuesday, February 28, 2006 7:37 PM
Paul C wrote:
I am running Asterisk 1.0.9 and have been running all my calls through a VSP over a IAX2 trunk however we have recently purchased and connected a TE110p to a PRI ( E1 with 16 voice channels ) through Optus. I can make outgoing calls via it fine, however incoming calls are dropped
Sam Tam wrote:
Hello Dan
I can assure you that our GSM Gateway quality is absolutely excellent and
this fact can be supported by hundred if not thousand of our users.
It is also very simple to use and even a newbie can set it up..
Does it provide Disconnect Supervision? If so, what method
mustardman29 wrote:
I have one question,
How does a large file transfer like your excel spreadsheet example, affect
communication between an Asterisk server and SIP phone? The only possible
configuration I can think of that would cause a problem is if the client PC
is sharing the same eternet
Chris Bagnall wrote:
...or if your
asterisk server is also a file server (which should never be
done)
I know I'm attracting flames for disagreeing, but sometimes when you're
dealing with small business customers there simply isn't the budget to have
separate machines for doing x, y and z, and
Douglas Garstang wrote:
It was trying to perform looping in the dialplan that made me seriously look at
AGI. Gee, I wonder what's easier.
This:
exten = s,1,Set(COUNT=0)
exten = s,2,Goto(loop,1);this is where we start the loop
exten = loop,1,GotoIf($[${COUNT} 5]?next,1);exit if more than 5
[EMAIL PROTECTED] wrote:
- Original Message -
From: Tomislav Parčina [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 20, 2006 10:46 PM
Subject: [Asterisk-Users] Re: Call centre - * hang's up
In
If the line is answered, you frequently need a
/etc/asterisk/indications.conf in order to get ringback.
Frederic Jean wrote:
Hi everybody,
I sent an e-mail this morning regarding SIP / IAX2
with no ring-back, I now succeeded to pin-point the
problem, here it is, if I dial a provider directly
Jayson Navitsky wrote:
See the problem is when I do
Dial(Local/[EMAIL PROTECTED]Local/[EMAIL PROTECTED]Local/[EMAIL
PROTECTED]Local/[EMAIL PROTECTED],30)
If someone is on the phone it returns Busy and then kills the incoming
call. ChanIsAvail would work great if I was going out to the PSTN
Reli Loin wrote:
hello,
I have one account i need using multiple sipura ata, for my account.
it's possible in asterisk.
No. Generally you never need multiple devices to use the same account
information. This has been talked about in the archives.
Personally I use the MAC address of the
Richard Perini wrote:
On Sun, Feb 12, 2006 at 03:35:56PM -0500, John Novack wrote:
That certainly is the way it SHOULD work. Blind and attended transfer
should be able to be initiated the same way. It certainly is the most
efficient logical way. Attended transfer should revert to blind simply
[EMAIL PROTECTED] wrote:
Hello All,
I am trying to figure out which way to go for a quad port fxo solution
with a good echo can on it. My options are the sangoma remora, a
mediatrix fxo, or something similar.
The issue is that I would need a good EC. This would be on about a 9000
foot loop,
Since your EC only needs to support a tail long enough to handle the
PSTN part of the call, I suspect even fairly short tails are fine.
Steve Underwood wrote:
I don't know about the Tellabs cancellers in particular, but I think any
echo canceller built in the 80s will be a fairly poor
James Harper wrote:
Just an enquiring mind wanting to know, but how is a hardware solution
different to a software solution? The echo cancellers in the Digium
hardware presumably just use the same sort of algorithms as the software
versions, so it is just that they are dedicated and perform
Koopmann, Jan-Peter wrote:
I would like to see if during a call a new voicemail was recorded. I want to
send a SMS to mobile phones if someone recorded a message on our voicemail
system.
I can use VMCOUNT to see if there are new messages in the Inbox but this will
result in new SMS being
Michael Graves wrote:
Surely there's something more to the truly SIP-aware device, such as
the Ingate IX66, that merits their use in some specific circumstances?
I know that I can stay with m0n0. The question still stands; are there
circumstances when something more is required? Would something
JCC wrote:
I don't get it. What is the advantage of using a GSM gateway? VOIP calls are
pretty inexpensive as they are now. Is the use of a gateway intended as a
backup incase a wired network connection goes down? I have being looking
around the net for information on this. Anyone out there
Steven wrote:
Just do:
exten = _12xx,2,Dial(${TRUNK}/0${EXTEN}|30,r) ; adding zero
exten = _012xx,2,Dial(${TRUNK}/${EXTEN}|30,r) ; not adding zero
The zero is added before ${EXTEN}.
I have only ever used the stable versions and have always done it this way.
Never trust anyone that tells you
Jason D. Wolfe wrote:
Hello,
If I use an IAX termination service to connect outgoing VoIP calls to a PSTN
will I have answer supervision so that my script won't initiate too early?
Correct. (At least it should be correct as any decent service provider
will be using PRIs)
Olle E Johansson wrote:
Mikael Magnusson wrote:
Olle E Johansson wrote / skrev:
Andreas Koch wrote:
Hello,
how is it possible to connect (register) more the one Phone to One
Sip-Acoount.
With, for example sipgate.de this is not a special feature, it is
common.
We have users, what like
BJ Weschke wrote:
On 1/3/06, Kerry Garrison [EMAIL PROTECTED] wrote:
The magic setting is callprogress=yes, however, we have this working
properly in the lab but not at this particular client location right now.
Strange, but true.
-Kerry
You're going to have very unpredictable results with
Frank Liu wrote:
Within asterisk, is it possible to detect that an incoming call is a
direct dialing, or forwarded via another place? When a call is being
forwarded via a 3rd party (say, SBC), will it have some indication in
the call packet?
You mean something like as is documented in
Code Lover wrote:
Hi friends,
How i can enable and disable RBT in asterisk for SIP users.
We have linksys IP Phones but its give ring to the caller before
ringing the called phone.
Don't use the r option to Dial
___
--Bandwidth and Colocation
Joseph Rothstein wrote:
I am setting up 10 SNOM 320s for a customer, and there seems to be a problem
with call-limit and hints.
Here is my sip config for one phone:
[944]
type=friend
context=x
language=de
accountcode=x
notifyringing=yes
host=dynamic
dtmfmode=rfc2833
[EMAIL PROTECTED]
])
(using TLSv1 with cipher DHE-RSA-AES256-SHA (256/256 bits))
(No client certificate requested)
by bourbon.fnords.org (Postfix) with ESMTP id D5E5D88;
Mon, 2 Jan 2006 23:32:08 -0600 (CST)
Message-ID: [EMAIL PROTECTED]
Date: Mon, 02 Jan 2006 23:30:25 -0600
From: Eric \ManxPower\ Wieling [EMAIL
Rene Nelson wrote:
I have an SPA connected to my * box which seems to be configured
properly because it works 'mostly' but whenever I hang up a call, I will
get a phantom call immediately. If I don't answer it, it will ring a
couple of times then stop. If I answer it, I get no dial tone, no
Alyed Tzompa wrote:
sip.conf
[general]
port=5060
externip = www.theip.net
localnet = 192.168.1.0
localmask = 255.255.255.0
allow=all
Don't use allow=all. Use disallow=all and then allow= line for the
specific codec you want to use.
___
--Bandwidth
Leonard Burton wrote:
HI All,
I have an extension 300 (which is a remote sip connection) and when I
call it internally it will ring the other side but if I am on the DID
and dial 300 it will not ring the extension but It sounds like it is
ringing.
When I call on the DID to any other extension
Adam Moffett wrote:
I set up an IVR awhile back.
press 1 for sales, press 2 for support etc etc.
Everything works fine except when you enter your option there is a 7 or
8 second pause before the next step is taken in the dial plan. I assume
it's waiting to see if I'm going to dial more
Imagine this:
[fnord-context]
exten = 1,1,Noop(Selection 1)
exten = 2,1,Noop(Selection 2)
exten = 3,1,Noop(Selection 3)
exten = 4,1,Noop(Selection 4)
exten = _XXX,1,Noop(Wants to call ${EXTEN})
When you dial option 2 how does Asterisk know you don't want to call
extension 200? In the
*sigh* Analog Zap FXO ports consider the call answered as soon as
it's finished throwing the DTMF at the telco. This is because a Zap
port CAN'T tell when an analog call has been answered.
Andrew Kohlsmith wrote:
On Saturday 17 December 2005 15:23, chawki hammoud wrote:
[tele]
exten =
Kristof Hardy wrote:
The problem has been consistent from 1.0 through CVS to 1.2, and
across different machines and distributions. Does anyone have any
suggestions on how I can deal with this? I have had echo cancellation
happening, but half-duplex speech is not acceptable.
You're not using
The phone's built in dialplan is prolly blocking the call. Check the
docs for your SIP device. Remember SIP devices collect all digits, then
pass them on to Asterisk as one packet.
Also what Zap port is your analog phone connected to? What card are you
using?
Robert La Ferla wrote:
Doug
Douglas Garstang wrote:
What exactly do you mean by 'documented not to be implemented'? If you are referring to the fact it isn't implemented, yes I realise that. That's why I'm trying to get an idea for when these features will be. This isn't whining.
If you are however, stating they are
Philipp von Klitzing wrote:
Asterisk is not perfect, there is a lot of work in progress (sometimes
too much), but it is the only one of its kind, it works, and it gets
better day-by-day. And if you find a way to help with that (and
preferably a way that doesn't step on people's collective
Robert La Ferla wrote:
Derek Whitten wrote:
[incoming]
exten = s,1,Dial(SIP/myextSIP/myext1SIP/myext2,25,t,r)
exten = s,2,Voicemail(myext)
exten = s,3,Hangup()
Thanks. This will call/ring multiple extensions but what about waiting
for X rings before going to voicemail? How do I do that?
show application dial is your friend. Love it, hold it, buy it
flowers, but READ it.
Robert La Ferla wrote:
I realize that it's a timeout but what's implicit in that is that
Asterisk can't detect # of rings just the amount of time spent ringing?
I have been looking at the reference manual
This information is outdated or wrong for some versions of Asterisk. Be
smart, read the output of show application dial in the asterisk server
you are using and see what it tells you.
Time Bandit wrote:
I have been looking at the reference manual on asteriskguru.com. They
say it's a timeout
T/t/H/h and other options to Dial require Asterisk to stay in the RTP
stream.
George Pajari wrote:
We are trying to use Asterisk to set up a call between two SIP devices
and then step out of the path.
- all systems have public IP addresses (no firewalls, no NAT).
- sip.conf has
Douglas Garstang wrote:
Can someone tell me when SRV lookups are going to be fully supported in
Asterisk? I see we just had a new release, 1.2.1. Considering this lack of
functionality is a huge gaping hole for reliability, I would have thought 1.2.1
would have been a good time to implement
The device that interfaces to the PSTN is the interface that must cancel
echo. If I read your post correctly, that is the SAP-3000 and the
Audiocodes boxes in your case.
Jeff Busch wrote:
Update on this... And it is still not solved.
This is actually fairly interesting. I have two
Sounds like you have an allow=all somewhere.
Ryan Courtnage wrote:
Hi all,
I'm finding with Asterisk 1.2.1 (and 1.2.0) that when connecting over an
unauthenticated IAX2 connection (ie: as [guest] in iax.conf), a codec
will always fail to be negotiated (see trace snippet below).
The problem
I seem to recall a similar issue where the guest section HAD to be the
last section of iax.conf. It's been that way for years.
Ryan Courtnage wrote:
On Wed, 2005-07-12 at 16:41 -0600, Eric ManxPower Wieling wrote:
Sounds like you have an allow=all somewhere.
Thanks for the response
You have the contact set to the extension, you need the contact set to
whatever you dial to retrieve your voicemail. i.e. the one that runs
voicemailmain.
Brent Bloodworth wrote:
Actually I think that is how it is setup now. I configured the phone
through the web interface. Callback mode is
funny guy wrote:
Just wondering, is the echo canceller in the TE411P capable of
cancelling the echo caused by the delay over satellite link (i.e. approx
400 ms delay)?
Does anyone have any success story to share?
I'm kinda stuck with a really2 annoying echo... adjusting the gain
Colin Anderson wrote:
Don't want to point out the obvious, but seems to me that the lowest common
denominator here is to dial out the PRI if there's no extension match,
correct? If this is the case, then you can use the 's' extension. The 's'
extension is a 'match-none' extension and is invoked
Damon Estep wrote:
I have some phones that perform better with rfc2833 for DTMF, but a
termination provider that only supports INBAND.
Is this possible;
Yes. It happens automagically. BTW, pretty much all phones will have
more reliable DTMF if you are using RFC2833.
Michiel van Baak wrote:
On 14:42, Mon 05 Dec 05, snacktime wrote:
On 12/5/05, calvis [EMAIL PROTECTED] wrote:
I need to replace my switch. Does anyone have any recommendations for a
switch that is VoIP friendly? I want it to be a managed gigabyte switch.
There are lots of brands out there,
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