Chee Foong wrote:
Firewall shouldn't be a issue since the call works fine with ztdummy loaded.
I debug the chan_h323 and it uses the right codec G729 from digium.
H.323 does NOT deal with NAT or Firewalls without a smart edge device.
chan_h323 does not use ztdummy whatsoever, so that has no be
On 12-Aug-03 Dave Cotton wrote:
> What actually _read_ a manual, only wimps do that. Easier to ask someone
> else. :)
>
> He could try
>
> *CLI> help
>
> But then you'd have to read.
>
Hi Prakash -
As a slightly more sympathetic new * user... here are a few things I've
discovered in the
Thanks, I didn't realize that was the code for the codec. So, How do I know what codec
that code
translates to?
--
Paul
"WipeOut ." wrote:
> At the console while a call is in progress run "sip show channels" and look in the
> format column..
>
> > How can I tell what codec a SIP session is usin
Same thing. It will make sense to try
Register => @fwd.pulver.com:@fwdnat.pulver.com:5082
but in that case Asterisk sends
REGISTER sip:fwdnat.pulver.com SIP/2.0
which is not right. It should be sip:fwd.pulver.com but sent thru
fwdnat.pulver.com:5082
BR Borut
-Original Message-
Subjec
For PRI->*->fax over FXS
It's as simple as having the fax extension the the incoming context
associated with the PRI channels. With PRI channels we can hear the fax
before we even answer (in most cases)
regards
Martin
On Tue, 12 Aug 2003, Adams, Gavin wrote:
> > From: Martin Pycko [mailto:[EMAIL
I don't understand the reasoning here so could
somebody please help me out?
chan_h323 is causing a segmentation fault when
trying to connect a call.
I tracked the problem back to chan_h323.c in the
oh323_new() function.
the code is: tmp = ast_channel_alloc( 1
);
After this point, tmp->
Dears
I got one way audio using SIP, X-Ten and SIP ->
POTS integration. Does anybody knows some bug ?
Tks
Adelino Baena
and a little more on fax config support
- Original Message -
From: "Tilghman Lesher" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Monday, August 11, 2003 4:48 PM
Subject: Re: [Asterisk-Users] Fax Handled
> On Monday 11 August 2003 03:26 pm, Eduardo Goncalves wrote:
> > On Mon, 11 Au
> "Steve" == Steve Underwood <[EMAIL PROTECTED]>:
Steve> Kim C. Callis wrote:
>> I was reading on www.vovida.org/applications/downloads/G729A/ (home
>> of VOCAL) pages, and that there is a free license use for
>> non-commercial for G.729A. Is that usable under Asterisk or strictly
>> a Vov
Are the VoiceAge people generally unpleasant to work with and geniunely
uncaring, or do they just fail to respond?
Matt Hardeman
PaperSoft
- Original Message -
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 12, 2003 10:16 PM
Subject: Re: [Asterisk-
yes
On Wed, 13 Aug 2003, Chee Foong wrote:
> Hi,
>
> I manage to solve the problem. I just change the span configuration in
> zaptel.conf to E1 configuration. Unload zaptel driver and load it again. It
> seems to work fine.
>
> I would like to know if RFC2833 is equavalent to out of band DTMF?
>
Hi,
I am trying to think of the best way to manage the phone to extension to user
relationships in an Asterisk system so I am asking for any input on best practices
from thos out there who are running live systems.. the bigger the better...
It seems the common practice is to name the config for
I inquired to Grandstream about their resellers and they pointed me to
an establishment that never got back to me with a quote, even after
multiple reminders.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> James Sizemore
> Sent: Wednesday, Augus
Hi.
I'm trying to do some custom call logging, and I want to call an AGI
script from a hangup handler to log call durations and things. Although
the script executes, it isn't retrieving variables from the AGI
interface. Looking closer, I realised the variables are actually getting
unset before
If you can plug it into a regular analog phone line and have it work,
then it will work with Asterisk.
On Wed, 2003-08-13 at 10:42, Chris Hale wrote:
> Anyone know if the AT&T 964/954 series phones have any issues with
> Asterisk? We have 5 phones and would like to reuse them if possible.
> An
Last I looked, FWD is G711 only, unless you use the lite service, then
it is G729 only. No asterisk work will change FWD's setup.
On Wed, 2003-08-13 at 09:56, Jose Ildefonso Camargo Tolosa wrote:
> Hi!
>
> I'm trying an asterisk-FWD connection. I'm using X-Lite OR SIPPS as the
> IP phone. I c
unsubscribe
Dave Wilson wrote:
[...]
[default]
s,1,AGI(bash-scriptname.sh)
To call my script from asterisk?
That should work fine. You need to put the shell/perl script in the
agi-bin directory specified in /etc/asterisk/asterisk.conf (typically
/var/lib/asterisk/agi-bin/). Make sure you remember to chmod +
Hi all,
I'm about to start setting up my first asterisk/cti system in our test lab.
I've read through all the documentation I can find and relevant posts in the
list archives but can't seem to find anything explaining how to go about
initiating an http request upon an incoming call.
I basically w
Hi Steve
>Steve Underwood wrote:
> 06.10 isn't that great a codec,
> though. I don't think it is used very much on the GSM networks these
> days. Most of the time they use the enhanced full rate (EFR) or half
> rate codecs.
>
What do you mean by "isn't a great codec?
There is any major advanta
You can run SIP reliably behind nat with a SIP-aware adsl router. We have
set this up for a few customers, as well as for ourselves and it works fine.
http://www.telappliant.com/intertex_sip_aware.htm.
Tan
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tue
On Wed, 2003-08-13 at 16:15, Lee Goodman wrote:
>
> There is also a thread about how the XTEN softphone seems to
> have a DTMF problem. XTEN says they fixed it, but people
> (including myself) still see the problem. I wonder if there is
> a bug in Asterisk
IMHO for basic telephony I would say go for the Budgetone, At its price you can't go
far wrong and it works..
I have heard some comment that it is too light and so feels cheap but I have had a
Snom200 and a Budgetone side by side on the desk in front of me and they both felt the
same to use.. I
Being a relative Asterisk newbie, I may be wrong.. but as far as I can tell,
it doesn't. The standard queue/agent logic requires that you assign an
extension to a phone.
Someone correct me if I'm wrong, please. :)
- Devon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTEC
> From: George Lin [mailto:[EMAIL PROTECTED]
>
> I want to deploy multiple SIPs phone in our office. And we have
shutdown
> the
> firewall at our office router(with ip 211.x.x.x). we have deployed the
> asterisk with IP 218.x.x.x.
>
> All SIP phones have 192.x.x.x.
We have something similar Geor
The Cisco is from what I have heard a good phone but is VERY expenisve..
My suggestions would be to go with either a SNOM 200 or a Grandstream Bugetone..
All these have ben discussed many times in the list so search the archives and read up
on them..
> Hello,
>
> I would like to buy a SIP IP
How can I tell what codec a SIP session is using?
--
Paul
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Once the call has hungup the AGI functions stop working, so executing an
AGI from an h extension will not do what you expect. but you can still do
all kinds of perl stuff there, so you could stick the info in mysql or
somewhere else
James
On Wed, 13 Aug 2003, Alastair Maw wrote:
> Hi.
>
> I'm
How about when you compare the SNOM to the Budgetone, which one would you
recommend for basic telephony?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of WipeOut .
Sent: Tuesday, August 12, 2003 2:15 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IP ph
Big issues for sip: (Please note I use both Asterisk and Vocal between
the two you can have a fairly scalable sip environment with a fair
amount of call features.)
Pluses for Vocal:
For sip switching Vocal is much more scalable, You can have a cluster of
UserAgents and Gateways. It never termi
Hi
Has anyone ever been able to use webmin module that is in asterisk
directory. I will appreciate if someone quickly lays it out.
What is the admin interface that anyone uses, which one is the best. I
have seen refrence to different managers like Astmanagerm and Gastman. Are
the useful, is the
Sorry, I miss typed , I meant
XLite not XTEN as the softclient I was talking about :)
- Original Message -
From:
Lee
Goodman
To: [EMAIL PROTECTED]
Sent: Wednesday, August 13, 2003 9:40
AM
Subject: Re: [Asterisk-Users] Weird DTMF
issue
Ok, I think I
/usr/src/usr/include/mysql/errmsg.h
The version of MySQL that I'm running is 3.23.57-1
Could you tell me where mysql/errmsg.h is located on your
distribution? We can update the Makefile to look there for that
header.
-Tilghman
_
Prot
Look for the line "usecallerid=yes" in the file"/etc/asterisk/zapata.conf".
Disabling caller id did resolve my two ring issue.
Jeff
[EMAIL PROTECTED] wrote on 08/13/2003 03:49:47 PM:
> On Tue, 2003-08-12 at 21:59, Steven Critchfield wrote:
> > To answer on the first ring turn off callerid s
My users also like the Budgetones better then the Cisco 7960 phones. The
Budgetones just handle echo better. (Cisco really need to get there echo
problem fixed!)
My only problem with the Budgetones is getting them. We would like to
place an order
for a 100 of them, and can't get Grandstream to
Sorry, rtfm issue!
Regards,
Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
[EMAIL PROTECTED]
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How do I disable faststart in Asterisk?
Regards,
Sean Langley, P.Eng
Firmware Engineer
General Dynamics Canada
(403)730-1482
[EMAIL PROTECTED]
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Title: RE: [Asterisk-Users] I can't get a two way conversation going?
This is going to sound like a stupid question but I am going to ask it anyways. I have been a victim of my own stupidity on this one before. Ok here goes... is the microphone muted?
-Original Message-
From: Leif Mad
Alastair Maw wrote:
log-call.pl
[...]
oops. I meant:
> $sql = "INSERT INTO call (call_from, call_to, add_date) values
> ('$clid', '$dnid', FROM_UNIXTIME($add_date));";
not:
$sql = "INSERT INTO call (call_from, call_to, add_date) values
('$myclid', '$mydnid', FROM_UNIXTIME($add_date));";
Also note
Ok, just been thinking about this and thought I would ask
before trying it out again.
What is the state of SIP transfers? By this I mean
transfers initiated via SIP messages, not via DTMF and
'#'.
Last time I tried, on X-Lite, clicking the transfer button
dropped the call.
Also, are/will
Hopefully these are all correct:
-No, a SIP phones cannot connect to Asterisk voicemail
using G.729 if you do not have a licence. You will need a
licence for at least one channel to do this.
-DTMF is not related to the codec itself or how it works,
however, inband DTMF is not recommended on co
Someone knows if there is any intention to build hardware IP phones or
strandalone interfaces (like ATA186) based on IAX and GSM?
Thanks,
Dan
Original Message -
From: "John Todd" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, August 12, 2003 11:49 PM
Subject: [Asterisk-User
> -Original Message-
> On Wed, 2003-08-13 at 15:45, Devon Henderson wrote:
> > Exactly, Steven. However, we also want the following logic to work:
> >
> > Agent 1 (ext. 1234) can login (either via the telephone, or
> preferably, a
> > web interface) from either of the below locations and h
> Could you tell me where mysql/errmsg.h is located on your
> distribution? We can update the Makefile to look there for that
> header.
Can't you use mysql-config to get the include and library paths? Granted,
you still need to make sure that mysql-config is in your $PATH, but it
keeps you from
Hi Dan,
Dan wrote:
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 13, 2003 9:49 AM
Subject: Re: [Asterisk-Users] Open G.729A codec
Steve Underwood wrote:
After writing this I got curious about how fast/slow the ITU ref
Welcome to the club... I can't get it working either.
bkw
On Wed, 13 Aug 2003, Eric Wieling wrote:
> Is there any way on the iaxtel.com web site to see if my asterisk is
> registering and what 700 number is associated with my iaxtel account? I
> registered many months ago but never used it. My
On Wed, 2003-08-13 at 15:45, Devon Henderson wrote:
> Exactly, Steven. However, we also want the following logic to work:
>
> Agent 1 (ext. 1234) can login (either via the telephone, or preferably, a
> web interface) from either of the below locations and have calls from one or
> more queues rout
At 11:28 AM +0300 8/14/03, Dan wrote:
Hi,
I have defined several extensions.
For some of them, the phone can be disconnected for a period of time (mobile
users).
When a call is initiated to that extension, if the user is not connected at
that moment in time the caller see this as a busy extension.
Hello,
I have got the CDR from asterisk, I know a
particular record is a conference call by looking at the dcontext fields.
I wonder how do I know what room that particular
caller was in based on the cdr record?
Do I need to dive into the meetme source and do
some modification to do t
Maybe you will need to use ALSA driver for i810 witch will unleash the
power of full-duplex from it. Maybe = for sure!
Cristian Vasiliu
AccessNET
mail to:<[EMAIL PROTECTED]>
web :
prakashmodak_74 wrote:
Hi all,
I'm using RH 7.1,and install Asterisk0.4.0 with I810 Fullduplex onboard Sound, i get
On Thursday 14 August 2003 16:42, Paul Cheng wrote:
> Does anyone know what this means?
>
> I suspect it has something to do with ztdummy as app_meetme no longer
> works--though it did a month or two ago.
No, it means that a built file has a timestamp older than the source
file. Usually this happ
Fixed it. Looking back a few emails, someone mentioned SIP natting.. so
this appears to have fixed my problem:
sip.conf:
[general]
register => X:[EMAIL PROTECTED]/1000
;; Free World Dialup Proxy
[fwd.pulver.com]
type=friend
host=fwd.p
You should instead update the appropriate Makefile to change:
-I/usr/local/mysql/include
to
-I/usr/src/usr/include/mysql
or you could just do:
cp -a /usr/src/usr/include/mysql /usr/include/mysql
I did that, and I still get the same error:
[root cdr]# make cdr_mysql
cc -fPIC -I/usr/local/mysql/incl
Hi,
I cannot use '#' to initiate transfers.
I have tried on different phones (7960, ATA, X-Lite).
When I press '#' during a call, nothing happen.
I have both T and t switches in Dial application.
The transfer function works with Flash key on ATA, but in a very strange
wayThe final destination
>
>I think this is a good idea but at least for FWD users can't they just use
>the FWD proxy that is
>designed to handle clients behind NAT with no special software on the
>client. The ones that
>allows even Windows Messenger to work behind NAT.
>
Sadly this doesn't work otherwise they'd all be
Does anyone know what this means?
I suspect it has something to do with ztdummy as app_meetme no longer
works--though it did a month or two ago.
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On Wed, 2003-08-13 at 18:07, Devon Henderson wrote:
> > -Original Message-
> > On Wed, 2003-08-13 at 15:45, Devon Henderson wrote:
> > > Exactly, Steven. However, we also want the following logic to work:
> > >
> > > Agent 1 (ext. 1234) can login (either via the telephone, or
> > preferabl
The highest quality codec is ulaw or alaw (otherwise know as G.711).. These are the
same as what comes in on your PSTN line..
If you want high Quality voice prompts your best bet os to record them on a PC with a
good quality mic and then copy then to your server.. You juast have to make sure tha
Hi,
How does *, in a particular context, determine which extension matches
and particular dialed number? Most specific to least like routing
protocols? In order? Local context first then included context?
I have a context my SIP phones are in that uses _X. for local extension
dialing (to change t
--On Wednesday, August 13, 2003 1:58 pm -0500 James Sizemore
<[EMAIL PROTECTED]> wrote:
If you add "t" to you out-going trunk "Dial" lines:
exten => _NXX,1,Dial(SIP/[EMAIL PROTECTED]||t)
exten => _NXX,2,Congestion
so that you can still use park to park a call or transfer
the phones, Yo
Well I wasn't sure :) But if it works than it works :)
Martin
On Thu, 14 Aug 2003, Brian West wrote:
> Accually it will work with any codec if you use rfc2833. G711 is only
> needed if you are passing DTMF inband.
>
> bkw
>
> On Thu, 14 Aug 2003, Martin Pycko wrote:
>
> > It works only with G71
Try "cvs update -dA" in asterisk
On Thu, 14 Aug 2003, Jerk Face wrote:
> >You should instead update the appropriate Makefile to change:
> >-I/usr/local/mysql/include
> >to
> >-I/usr/src/usr/include/mysql
> >or you could just do:
> >cp -a /usr/src/usr/include/mysql /usr/include/mysql
>
> I did tha
try using DigitTimeout application
regards
Martin
On Thu, 14 Aug 2003, Manoj K Gupta wrote:
> Hi list,
>
> Can anyone give me a hint on how i can introduce some delay in dialing an
> extension and thus preventing timeout, when i press # to transfer a call.
> And also in which context should i w
Since nobody else took the hint, I submitted it as a feature request for SIP.
http://bugs.digium.com/bug_view_page.php?bug_id=104
Personally, this is not high on my "I'd love to see this fixed" list.
However, many others here are less fortunate from a network
perspective, and you're stuck b
I updated asterisk this morning "cvs update -dA"
When I try to run Asterisk (asterisk -vvvc), I get the following error:
[cdr_mysql.so]WARNING[16384]: File loader.c, Line 226 (ast_load_resource):
/usr/lib/asterisk/modules/cdr_mysql.so: undefined symbol: mysql_init
WARNING[16384]: File loader.c, Li
Only audio playbacks or on voice calls too ?
Make sure the video card is in a text mode. What spec is the machine ?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Asterisk -
linux - JVB
Sent: Thursday, August 14, 2003 2:22 PM
To: [EMAIL PROTECTED]
Subje
On Thu, 14 Aug 2003, Eduardo Goncalves wrote:
> I'm using G.711alaw.
> My extensions.conf:
>
> ===
> [globals]
> TRUNK=Zap/g1
> [sip]
> exten => s,1,Background(demo-moreinfo)
> exten => fax,1,Dial(${TRUNK}/${EXTEN})
> exten => _0.,1,Dial(${TRUNK}/${EXTEN})
> exten => _9.,1,D
Hi!
I decided to apply Chris's patch for the rtp problem, it is working just
fine now. Thanks Chris!.
I think that Mark should submit it to the CVS.
Ildefonso.
[EMAIL PROTECTED]
Pete,
Try this patch below... I noticed that eStara's softphone has the same
problem as xten's softphone when it c
mark wrote:
>
> Can you try iax2?
>
We tried that, but couldnt seem to get the peering to work on IAX2. We being
myself and a third party gateway provider in UK
Dave
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Dear all,
I like to know if the DTMF option is related to the codec or not. Can a SIP
phone with g729 codec to access asterisk voicemail2 in case the asterisk
does not have g729 license ?? If yes, what is the DTMF option inband or
outband ??? Is there any successful experience ???
Regards,
Geor
--On Thursday, August 14, 2003 12:58 pm +0200 Dave Cotton
<[EMAIL PROTECTED]> wrote:
Last night I posted showing that the problem is repeatable and only
occurs in one certain circumstance. I think it is within voicemail.c. If
the caller exits voicemail by pressing # the line is dropped correctl
It works only with G711 (ulaw/alaw)
regards
Martin
On Thu, 14 Aug 2003, Dan wrote:
> Hi,
>
> I cannot use '#' to initiate transfers.
> I have tried on different phones (7960, ATA, X-Lite).
> When I press '#' during a call, nothing happen.
> I have both T and t switches in Dial application.
> The
I would imagine that using some sort of variable temp structure might
help you. However, the trick is keying the database tree so that in
the event of multiple calls at the same time, you're looking at the
"right" tree during your post-call processing. Hmm... is ${CHANNEL}
getting erased in t
On Thursday 14 August 2003 08:30, Eduardo Goncalves wrote:
> On Wed, 13 Aug 2003 17:56:46 -0500
>
> Tilghman Lesher <[EMAIL PROTECTED]> wrote:
> > The CNG tones are sent by the sending fax machine, not the
> > receiving fax machine. Those tones are sent from the moment that
> > the fax machines di
Hi Brain,
Strange, but changing the parameter ConnectMode from 0x0060400 to 0x00460400
solve this issue.
But even more strange is that I have tried to change it back and it still
works..?!?!
Now I have another problem. Using '#' and trying to transfer the call to
701 (parking) it is transfered, t
As far as I remember it is only in playbacks - voicecalls ok. Videocard
in text mode? (does it have a special reason? Need to check how to do so)
::... . :: Specs ..::.. . .:;
Pentium 4 - 2.4 GHz - 533 FSB
512 MB RAM
Soundcard - soundblaster PCI16 (chip ES1373)
Video - nVidia Gforce4 -
Brian West wrote:
Welcome to the club... I can't get it working either.
bkw
On Wed, 13 Aug 2003, Eric Wieling wrote:
Is there any way on the iaxtel.com web site to see if my asterisk is
registering and what 700 number is associated with my iaxtel account? I
registered many months ago but never
To answer on the first ring turn off callerid support. If you need
callerid support and answering on the first ring, then you must leave
analog phone signaling.
On Tue, 2003-08-12 at 16:22, [EMAIL PROTECTED] wrote:
> It appears that my X100P card is only answering after two rings. Ideally,
> I'd
Assuming this is on incoming calls, the most usual source of the problem is
that the telco exchange either doesn't send a disconnect pulse or the wcfxo
driver doesn't recognise the format used. I've unfortunately forgotten the
exact situation but, when a call finishes, a telco exchange in the U
The hangup stuff is a known problem in many networks. But can't say much about
it since we don't use analogue lines for a long time now. Th e400p works fine
on close to any ETSI PRI lines so should work fine in Belgium as well. But
for the sake of performance and flexibility I would anyway rathe
I've never actually placed a call with the budgetone, but it just looks and
feels cheap. The Snom is much nicer, and I tend to find that the Cisco
7960s/7940s are quite nice as well.
Matt Hardeman
PaperSoft
- Original Message -
From: "Uriel Carrasquilla" <[EMAIL PROTECTED]>
To: <[EMAIL
Anyone know if the AT&T 964/954 series phones have any issues with
Asterisk? We have 5 phones and would like to reuse them if possible.
Any restrictions or clunky workarounds needed?
Thanks in advance,
Chris
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Forgot to mention that we have specified the nat=yes for all sip entries in
sip.conf.
Regards
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of George Lin
Sent: Wednesday, August 13, 2003 10:32 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SIP NAT questio
I have tried both G711u and GSM codecs, and I get the same problem with
both. The asterisk computer is running a TD20B card with two phones
attached. I call from my laptop with a microphone to the asterisk box.
Phone rings, I answer and the call doesn't drop. I can talk into the
phone and hear my
Brian, what the person was asking for was to have a user with extension
1234 sit down at a desk and login to their pbx and have extension 1234
ring the phone the login was done from. What you described below would
tell the switch that user is sitting at extension , but not that
1234 should ring
- Original Message -
From: "Steve Underwood" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, August 13, 2003 9:49 AM
Subject: Re: [Asterisk-Users] Open G.729A codec
> Steve Underwood wrote:
> After writing this I got curious about how fast/slow the ITU reference
> code real
If I had to venture a guess, I would say that the protection scheme is in
place in the hopes that everyone will use their implementation rather than
reinvent the wheel. If this is indeed the case, their protection scheme is
useful in helping to protect the patent license as well as their code. So
We're still in the planning stages of our Asterisk implementation, but we
have a requirement that the extension be mapped to a user, with the phone as
a variable (we have hot seats in our contact center, and we also have agents
that work both from remote locations and our contact center).
So, I am
Has anyone had the opportunity to use a PingTel phone with Asterisk?
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of WipeOut .
> Sent: Wednesday, August 13, 2003 2:01 AM
> To: [EMAIL PROTECTED]
> Subject: RE: [Asterisk-Users] IP phone recommendation
On Wed, 2003-08-13 at 19:41, Steve Meyers wrote:
> Well, I'm in the US, and I still have the problem, so I'm assuming the
> problem isn't some European-only problem. Mine is sporadic, however -
> if you're getting the same thing consistently, then maybe your problem
> is worse.
OK Getting a bit f
I'm not sure what to call what I want to do; I'm not sure how to do it;
I'm not sure it can even be done. . . .
I have googled and read the docs, but can't find anything close to this:
I have an asterisk box with an X100P that is attached to a PBX line.
What we want to do is to have the PBX att
How about something like this?
exten => _7,1,Ringing
exten => _7,2,Wait(4)
exten => _7,3,Voicemail2(u${EXTEN:1})
exten => _7,4,Hangup
Just be sure you get the extension right :)
Mark
On Wed, 13 Aug 2003, Brian Capouch wrote:
> I'm not sure what to call what I want to do; I'm no
Hi,
I have defined several extensions.
For some of them, the phone can be disconnected for a period of time (mobile
users).
When a call is initiated to that extension, if the user is not connected at
that moment in time the caller see this as a busy extension.
How can I detect from the extensions.
This is the exact opposite of my experience. I've used a budgetone for about
4 months and I constantly have echo and ndp problems. They have made
frequent updates to their firmware - which I appreciate - but every other
version breaks something.
We ended up standardizing on Cisco 7960 phones and b
*This message was transferred with a trial version of CommuniGate(tm) Pro*
Ooh, can i answer this one? Please??
RTFM! :)
http://www.digium.com/handbook-draft.pdf
hehe...
Regards,
Jamie Carl
Jazz-Inc.
Email: [EMAIL PROTECTED]
Phone: +61 414 365 466
Jabber: [EMAIL PROTECTED]
-Origina
Yes, you can run Asterisk behind a NAT.
NO, you CAN'T (reliably, easily) run SIP behind a NAT.
For FWD think about using their behindnat and fwdproxy addresses.
Maybe a STUN would help. Also, test your setup infront of NAT also, make
sure they work, before you head behind a NAT.
--
wasim
This m
Hi George,
Do you have qualify=yes set in sip.conf for your phones?
When you check sip show peers, does it give you an OK (X ms) or does it
say UNREACHABLE or UNMONITORED?
If you enable qualify=yes or qualify=[some number] then Asterisk will
poll the SIP UA every once in a while to make sure i
gdb asterisk core.
then after its done loading type 'bt' (without the quotes) and send the
gory details.
(use -g to force a core drop)
Jeremy McNamara
Jim Friedeck wrote:
Just cvs'ed about 40 minutes ago (10:15 CST 8-8-03). Segfaults when I
use a queue app in many different scenarios. W
I don't think SIP should be bashed just of the NAT problems you guys are
having. Not that IAX isn't an -exellent- protocoll for its uses, but SIP
it a bit more than just p2p trunk VoIP.
F
On Wed, 2003-08-06 at 21:48, Tilghman Lesher wrote:
> On Wednesday 06 August 2003 02:27 pm, William Flanagan
Hi,
As far as I know docs are still under the construction and what is available
on the sites is it!
Senad
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of CallTrex
Personal Assistant
Sent: 08 August 2003 20:35
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
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