to transfer calls between networks
but I've never seen another CPE piece of equipment that can do this.
According to another IT magzaine, Divitas indeed uses Asterisk. But Divitas
does not seem to be a pure CPE solution. That may be why they could charge
a premium.
Yuan Liu
http
.
No special change in sip.conf required. I've transmitted SMS over local SIP
channel and it's be quire reliable - over LAN.
Yuan Liu
The SMSq stuff is for landline-type SMS, like those that never became
really popular here in Europe ;-) I do not know of any SIP hardphone
that supports them
of sip.conf and extensions.conf would be helpful if
you are not sure what to look.
Yuan Liu
What can I do to make sure I always send an error
sound and never again a busy signal?
thanks!
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this sequence?
Yuan Liu
Here are my files.
zapata.conf
context=incoming
switchtype=national
signalling=pri_cpe
group=1
channel=1-23
extension.conf
[incoming]
exten = _.,1,Dial(Zap/g1/19173995791)
# I have added this line in the dialplan is because I want it to
match the last 5 digit and simply
From: Mike [EMAIL PROTECTED]
Date: Fri, 11 May 2007 19:44:51 -0400
Yeah ok. That doesn't help.
What I mean is I want a call to go out on ProviderA, UNLESS it's down and
then go to ProviderB.
ChanIsAvail() is supposed to allow this.
Yuan Liu
I want it to ring 30 seconds and then Hangup
/index.php?page=Asterisk+Zap+channels
Yuan Liu
Is there a queues.conf option that I'm missing here?
Thanks for any advice,
Yaakov Menken
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the
requirements are.
Yuan Liu
Also, if ProviderA has a main server and a backup server, am I now forced
to
have 3 Dial commands, or can I setup ProviderA with host and backuphost in
the same SIP entry?
Mike
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From: Andre Wangler [EMAIL PROTECTED]
Date: Fri, 4 May 2007 07:35:38 +0200
Hello all
Could someone tell me what happens with running calls when reloading the
whole asterisk config files? I think SIP-calls are not
Nothing. All calls are maintained according to documentation.
Yuan Liu
in the way of a workaround. Does this
DTMF sequence absolutely have to be sent in the MIDDLE of the call or can it
be sent at the beginning, i.e., before any conversation starts?
Yuan Liu
channels. I'm trying to figure out if there's a way to then remove
asterisk from the RTP stream because
place and not
the right script.
Yuan Liu
Thanks,
James Texter
On Fri, 2007-05-04 at 18:44 +0200, Christian wrote:
Hi,
I have already done:
apt-get build-dep asterisk and then installed libpri, zaptel and
asterisk from the latest sources.
So what should i do then? New to Ubuntu.
many
From: Wilson Pickett [EMAIL PROTECTED]
Date: Thu, 3 May 2007 09:19:25 +0200
On 5/2/07, Yuan LIU [EMAIL PROTECTED] wrote:
From: Wilson Pickett [EMAIL PROTECTED]
Date: Wed, 2 May 2007 15:30:21 +0200
Is there a way to do the following scenario?
1) my asterisk box receives an incoming call from
? Is the originator a user in your system? Does the other URI
represent a peer? etc.
Yuan Liu
be broken (http://bugs.digium.com/bug_view_advanced_page.php?bug_id=9483).
Is there an alternative way to do this on Asterisk 1.2.18?
Regards
Cameron
. For some small sites, manually
setting up an AstDB family should suffice. This can even be semi automated.
Yuan Liu
At least that was what I found when working on the patch.
If anyone knows a way to lookup a peer/friend from the
dialplan and collect such details, it would be possible to
use
From: Steve Edwards [EMAIL PROTECTED]
Date: Tue, 1 May 2007 22:08:10 -0700 (PDT)
On Tue, 1 May 2007, Yuan LIU wrote:
From: Steve Edwards [EMAIL PROTECTED]
Date: Tue, 1 May 2007 21:10:40 -0700 (PDT)
On Tue, 1 May 2007, Jay Austad wrote:
I've got a directory under /var/lib/asterisk/sounds
to invoke this AGI from the
origination side can be very challenging. I can't think of a way right now.
Yuan Liu
Hangup(513)
etc, etc.
Anyone have any ideas?
--
Kristian Kielhofner
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with reinvite. (I assume that
the original SIP caller is in fact the toll free provider.)
Yuan Liu
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they'd be represented as
33666
332266
247467
So if the user enters 2 we know they want bishop
if they enter 336 they want demon and 332 they want deacon.
There was a similar discussion in the forum,
http://forums.digium.com/viewtopic.php?t=14559. Don't seem to have a ready
answer.
Yuan Liu
) information. I saw
there is a zaptel configuration entry that sound pretty close to what I
need 'callprogress'.
Set progressinband to yes in sip.conf.
Yuan Liu
Has someone already solved this problem?
Knud
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From: bilal ghayyad [EMAIL PROTECTED]
Date: Tue, 1 May 2007 14:56:14 -0700 (PDT)
Hi Noah;
ut TDM11B contains physically 4 ports, if it supports
only 1 FXS and 1 FXO, then what shall we do in the
other two ports already existed?
You can populate two more interface modules.
Yuan Liu
Regards
it to select a file to play randomly. Is there any way
to do this?
I do this with an AGI.
In 1.4, there's also a dial plan function RAND().
Yuan Liu
Thanks in advance,
Steve Edwards [EMAIL PROTECTED] Voice: +1-760-468
] context in extensions.conf.
So either create a [default], or change contexts used by Zap and SIP to
something you have in extensions.conf.
Yuan Liu
[channels]
signalling=fxs_ks
usecallerid=yes
callwaiting=no
threewaycalling=no
transfer=yes
cancallforward=yes
; valores validos 256(32ms),512
Any way to use chan_bluetooth as FXS?
Yuan Liu
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= i,1,Hangup
Am I missing some functionality from WaitExten if I do not plan to do
anything special after timeout?
Yuan Liu
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against it. The
card you use also matters. Heavy echo could also interfere with DTMF.
If SIP, the symptom you described would happen only to inband DTMF. Try not
to use inband if you can help it.
Yuan Liu
looking forward to your opinions... I really start to like toying around
for call confirmation, may not require loop -
http://www.voip-info.org/wiki/view/Asterisk+ZAP+channels
Hope this helps.
Yuan Liu
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resolving 'pbx_dundi.so' (in 'so'?): 205.166.226.38#53
Apr 24 11:02:38 asterisk named[1072]:
lame server resolving 'pbx_spool.so' (in 'so'?): 205.166.226.38#53
Looks unrelated to Asterisk. More like one of DNS servers used by Asterisk.
Yuan Liu
Anyone else or am I looking at doing some serious
such provisioning info, or one that could be
reconfigured on demand (outside of itself). I'd like to know which one(s),
too. Wouldn't imagine pushing user credentials to end points.
Yuan Liu
The employee is required to re-authenticate at the start of each soft phone
session or after a timed interval when
it stopped working. If not, and especially if you
can restart and get it working again, I'd suspect some hardware failure.
(Assuming the problem is reproduceable - I had times when TDM card stopped
working with no trace of error.) Try installing on another box.
Yuan Liu
SIP/701 is a Grandstream
SIP applet count?
Yuan Liu
perhaps if more people apply pressure, it will become possible to
extend their current (quite useable) provisioning interface, but have
a user-configurable setting to determine where the configuration is
fetched from. At present the configuration server setting is fixed
that includes
extensions/ID's allowed. Hope this helps.
Yuan Liu
thanks
arun
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asterisk start
just for this purpose. Then you should be able to use dial plan logic to
take action. Still not out-of-box, but adds a little more flexibility than
cron (in the sense of less programming, not in ultimate control).
Yuan Liu
Or if there is an available solution even that involves
have an extra channel to spare with (seems you do), can also try to
set up a context to receive SMS so you know all your commands/dial plan are
working before testing against operator. (I always test via SIP channel to
simplify my debugging. You can do so, too.)
Yuan Liu
/Per Jessen, Zürich
this message make sense, not registered?
Yuan Liu
Call-ID: [EMAIL PROTECTED]
CSeq: 102 OPTIONS
From: asterisk sip:[EMAIL PROTECTED];tag=as01265eaf
To: sip:freephonie.net;tag=00-31057-001dc208-591e1ca81
Via: SIP/2.0/UDP 82.XXX.XXX.XXX:
5060;received=82.XXX.XXX.XXX;rport=5060;branch=z9hG4bK67c2df66
show user via asterisk -rx and parse the result.
Yuan Liu
C F wrote:
I use qualify in sip.conf and need to setup a trigger when asterisk
sees it as unreachable, so that I can either drop a call file, or send
an email, or both. How can I do that?
Thank you
--
Building Strong Relationships w
us how you used ${TIMESTAMP} in Monitor()?
Yuan Liu
I try to export this as a environment variable but nothing changes.
Any help is welcome, thanks.
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), not Swisscom SMS centers. Not
sure why your caller never complained about spam calls, if Asterisk indeed
made the calls.
A quick fix would be (untested)
exten = 0900900941,1,Goto(smsmotx,${EXTEN},1)
exten = 0794998990,1,Goto(smsmotx,${EXTEN},1)
Hope this helps.
Yuan Liu
What am I doing
the
other party press more buttons.
Hope this helps.
Yuan Liu
This might be double posted because I'm not sure if my first posting went
through. Sorry.
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From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Mon, 16 Apr 2007 08:45:38 +0300
On Sun, Apr 15, 2007 at 10:10:34PM -0700, Yuan LIU wrote:
(But if Zaptel and Hylafax can share an X100P driver ...)
Where can you find a modem driver for a X100P?
Kinda my question, too. Motorola used to have
plan using SIPHEADER.
Yuan Liu
Jesus Mogollon
On 2/22/07, Craig Guy [EMAIL PROTECTED] wrote:
Hi Richard,
there was a thread regarding this a while ago on the dev list which
resulted
in a patch being made to allow variable passing via IAX2 channels. See
http://bugs.digium.com/view.php?id=7619
forum. Yes, somebody posted positive results.
Yuan Liu
I know Fax is not officially supported on TDM400P cards but I did not
expect
not being able of sending one single Fax.
Actually when I try to send a Fax, the call
machine.
On same machine is a bit exaggerated, considering there is a Zaptel card on
it. (But if Zaptel and Hylafax can share an X100P driver ...)
Yuan Liu
-Stephen-
I could have sworn that is what I just said.
Thanks,
Steve
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to simply ignore these. Anyone experiencing similar?
Yuan Liu
# L(x[:y][:z]): Limit the call to 'x' ms, warning when 'y' ms are left,
repeated every 'z' ms) Only 'x' is required, 'y' and 'z' are optional. The
following special variables are optional for limit calls: (pasted from
app_dial.c
the call to [EMAIL PROTECTED] (at least that's what I'm using);
whether that user (more precisely, the server that hosts this user) accepts
the call is up to the server.
Yuan Liu
Will my asterisk bridge a SIP phone that dialed 101 to the SIP user:
[EMAIL PROTECTED] Do I need some think more
From: Ronaldo Zacarias Afonso [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 11:54:51 -0300
Hi all,
Is it possible to configure an extension number to dial a sip address?
Nothing prevents you from doing this.
Yuan Liu
For example:
exten = 101,1,Dial(SIP/sip:[EMAIL PROTECTED])
That way I can
,Dial(SIP/${EXTEN})
exten = _4[0-1]X,1,Dial(SIP/${EXTEN})
exten = 420,1,Dial(SIP/${EXTEN})
exten = 350,1,Dial(SIP/${EXTEN})
exten = i,1,Answer(); if
exten = i,n,Playback(nice-message)
exten = i,n,DISA(nopassword,incoming)
Hope this helps.
Yuan Liu
You can search for the word irc to see my comments
for the toll-free and toll numbers? Are you
receiving the call from VoIP?
Yuan Liu
When i dial ordinary(not Toll-Free)number everyting is OK.
Please help.
Regards!
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From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 09:18:46 +0300
On Wed, Apr 11, 2007 at 08:09:16PM -0700, Yuan LIU wrote:
From: Sanjay Rajdev [EMAIL PROTECTED]
Date: Thu, 12 Apr 2007 01:29:51 +0530 (IST)
[good stuff sniffed]
and downloaded zaptel 1.4.1, after that executed
*8# into. Are you referring
to the pickupexten feature (default set to *8)? What if you change
pickupexten = *8 to pickupexten = ** in features.conf?
Yuan Liu
THANKS
Saludos, Lukassky.
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first as a selection
may have been made for you when zaptel wasn't loaded.
Yuan Liu
Can someone please help.
Regards,
Sanjay Rajdev
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polling event either internally or externally.
You are right. It shouldn't be hard to just require the primary server to
register with the backup, monitor this registration from backup; when
Asterisk on primary fails, run a script to request primary to shutdown and
take over.
Yuan Liu
You
thought for a
second they had solid ground to stand on, they would disclose which
patents they're referencing so the public could decide.
I bet you can access court records under some public information access
laws.
Yuan Liu
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announcements working properly, though.
Yuan Liu
thanks
arun
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. My reading has
led me to believe that manipulating gain on an IP PBX is neither necessary
nor practical in VoIP channels, so Asterisk does not devise such settings.
Yuan Liu
Thanks for any insight.
--
Bob Smither [EMAIL PROTECTED]
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and what is the result?) Also, you need to clarify the
settings on two servers more clearly - your sip-calls context seems to
suggest that server B uses IAX with its users, but uses SIP to connect to
server A? The iax.conf seems to suggest SIP rather than IAX.
Yuan Liu
i want to connecte
From: Mark Price [EMAIL PROTECTED]
Date: Wed, 4 Apr 2007 10:07:31 -0400
Is there a way to cause asterisk to accept all calls without any
authentication?
Mark
Yes - not to set up a user/peer section in sip.conf. The context in
[general] section will be used.
Yuan Liu
lsmod to confirm that zaptel is indeed installed. I'm not familiar with
CentOS or yum, but I assume you installed a binary package, so chan_zap.so
is probably included. Hope this helps.
Yuan Liu
On 4/4/07, Eric ManxPower Wieling [EMAIL PROTECTED] wrote:
Devraj Mukherjee wrote:
Hi Everyone
of 4003 being transposed for 4002 or vice versa and was not able to find
any.
What additional information is necessary to provide to trace down and
resolve this issue?
Corresponding entries in sip.conf may help.
Yuan Liu
AFAICT, the server is using Asterisk 1.2.x and beyond the 7960 phones
before list completes
exten = _Z.,n,Set(group=$[${group}=~${AVAILORIGCHAN}*(.*)])
exten = _Z.,n,Endwhile
exten = _Z.,check+101,Congestion; or however way you want to handle no
channel available
You may need to tweak a bit to get it working but that's the spirit. Hope
this helps.
Yuan Liu
a simple plan for testing :-
There is a simpler way, by using label.
[inbound-sip]
exten = uxbod(ntest),1,Dial(sip/1001,20,t)
exten = uxbod,n,PlayBack(uxbod)
exten = uxbod,n,Hangup()
exten = uxbod,ntest+101,PlayBack(uxbod)
exten = uxbod,n,VoiceMail([EMAIL PROTECTED],s)
exten = uxbod,n,Hangup()
Yuan
key. You guessed it: Flash stands alone. (I do appreciate the
corner location of the Mute/Del key. But Del is really not that useful to
qualify for this premier location.)
Yuan Liu
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From: Delca [EMAIL PROTECTED]
Date: Thu, 29 Mar 2007 18:39:37 -0300
How do I set rxgain per channel on zapata.conf? I've a TDM400 with 2 FXS.
Does FXS even use rxgain? To set rxgain for an FXO channel, simply put the
entry before saying channel =.
Hope this helps.
Yuan Liu
Thank you
CPU-intensive CODECs, the
number will drop sharply. It also varies with types of channels, i.e.,
whether you use PSTN, IAX, SIP, H.323. But still, I don't think 120 is any
limit.
Yuan Liu
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the channel chain instead of down.
I feel the need to propagate a variable up the chain from time to time. But
I still don't understand why this is necessary in your case, much less how
this relates to the need to unset. Maybe you can give more specifics, even
pseudo code.
Yuan Liu
From: Pezhman Lali [EMAIL PROTECTED]
Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT)
hello
is any web based sip phone?
The easy answer is yes. Search for Java SIP phone. Some of them can be
deployed on the Web.
Yuan Liu
for example:
a user after logining in, view a configured sip phone
at least up to 1.2.16.)
Yuan Liu
thanks
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Asterisk to generate random SIP traffic. Or maybe you
really mean SIP+RTP traffic. Either way, Asterisk can do it, just like you
can program C or Perl to do so. The real question is: what is
unsatisfactory about SIP traffic generators you have tried that you hope
Asterisk to help?
Yuan Liu
like on old style PBX
phones or multi-line phones. Short of having custom made phones, you can
play with dial plan and use, for example, a special prefix or postfix to
indicate which personality you want to present when outgoing. Is this
practical?
Yuan Liu
regards,
Drew
--
Drew Gibson
to the available zap
and original zap that receive it will be freed to receive another call.
It can only be used when all 2 lines ares used.
Thanks.
Lito
Telco calls this line rollover. No it cannot be done with Asterisk or any
PBX. It can only be configured on the telco side.
Yuan Liu
application.
Yuan Liu
best Mani
--- Yuan LIU [EMAIL PROTECTED] wrote:
From: Pezhman Lali [EMAIL PROTECTED]
Date: Fri, 30 Mar 2007 02:05:35 -0700 (PDT)
hello
is any web based sip phone?
The easy answer is yes. Search for Java SIP phone.
Some of them can be
deployed on the Web.
Yuan Liu
/user,20)
Getting more confused about what inbound call you did not see after reading
the sample conf. Did you put a context title before brads stuff? What is
your sip.conf/user.conf if you expect incoming call from SIP?
Ah. Feels good to teach grandma cook milk:-)
Yuan Liu
Thanks to all
? Is this extension dexter as your config
suggested? You can get much better response if you can help others
understand what your problem is.
Yuan Liu
My configurations
sip.config
[general]
context=default
register = raja:[EMAIL PROTECTED]/1234
bindport=5060 ; UDP Port to bind
application dial
Yuan Liu
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just the sip number and delete the remainder
'@server.com'.
Ideally I'd like to use 'SayDigits($([EMAIL PROTECTED])'
You can certainy use CUT(), or regular expression.
Yuan Liu
All replies greatfully accepted.
Phil
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is to have different personalities, some soft phones can have
multiple personalities as well so you don't have to run multiple soft
phones.
Yuan Liu
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serveur les
sache. Vous pouvez poser des questions plus spécifiques à l'égard
d'Asterisk. Mais je ne sais rien de carte VOIBE (Google ne donne pas plus
que des millions de millions de documents ruisses), donc je ne peux pas vous
en aider.
Au regard.
Yuan Liu
card.
Yuan Liu
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===Centos4.4
2.6.9-34.0.2.ELzaptel 1.4.1asterisk 1.4.2iksemel
1.2Dmitri Smirnoff
msn: [EMAIL PROTECTED]: 613 693 1299 ext 120
Rerun make menuselect?
Yuan Liu
or does it
need to be to see the card?
I had a similar situation. What I found was: the CMOS setup program had an
option to turn PCI 2.2 on or off - default was off. Later motherboards no
longer have this.
Yuan Liu
Thanks, Bart
Here's what I know:
Processors 1
Model Pentium III (Katmai
L() flag in when dialing the physical end point.
Yuan Liu
I think you can say something like:
AbsoluteTimeout (or in 1.2x, Set(TIMEOUT(absolute) = seconds) )
See: http://www.voip-info.org/wiki/view/Asterisk+cmd+AbsoluteTimeout
Thank you,
I will try later today, but I think this is what I
, add data network status (and possibly quality).
Yuan Liu
- config files backup (compare live and saved configuration files, if files
differ, notifies the administration team)
- systems variables (disk and CPU)
- log files (trigger an alarm for every ERROR or NOTICE message in full
logs)
What do
, a simple script to kick off some call files
should suffice. Won't take a week. (Search for call file.) But having to
deal with answering machines is always tricky for any automation.
Yuan Liu
Cory Andrews
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
. A user or friend will use the context to call
others through your Asterisk, therefore needs a context.
Yuan Liu
username=bob
host=dynamic
secret=nothing
context=BOB_SIP
qualify=yes
canreinvite=yes
callingpres=allowed_passed_screen
So what am I doing wrong? What do I have to change in order to get
From: Eric \ManxPower\ Wieling [EMAIL PROTECTED]
Date: Mon, 19 Mar 2007 11:26:56 -0500
Yuan LIU wrote:
Just noticed that no matter what the line condition is, zttool always
reports OK, so it's pretty useless. (In contrast, I'd get Red alert if
I unplug the line connecting to an X100P.)
I'm
Just noticed that no matter what the line condition is, zttool always
reports OK, so it's pretty useless. (In contrast, I'd get Red alert if I
unplug the line connecting to an X100P.)
I'm using zaptel 1.2.15 on Linux 2.6.15-28 (also tested on 2.6.10).
Yuan Liu
header before
really dialing boss' phone.
. Keep boss_xfer a top secret.
Yuan Liu
I've also tried to do it using different contexts, but it still doesn't
work. I've done like this:
[default]
exten = secretary_extension,1,Dial(SIP/secretary_extension)
exten = boss_extension,1,Dial(SIP
than that in PSTN, you can't deny advantages of a caller-pays system.
Yuan Liu
With my VoIP terminators, I can call most of the world's
landline's for a price so low I think of it as free,
with one exception -- the damn caller-pays cell phones
which cost over an order of mangitude more because
in such a way you could make a welcome message, for example, appear
on you contact phone screen.
Cheers
There was a thread indicating that you can do that with SendText() with
capable hard phones.
Yuan Liu
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above? If the PBX is
configured to take DNIS as DTMF string, D() flag could be used.
Yuan Liu
exten = 888111,n,Hangup()
The PBX just get the number 2 as it's DNIS when i change it to ZAP/1 or
ZAP/g1 the PBX get the number 1. What should i add to send the extension
number as DNID/DNIS
From: Wilson Pickett [EMAIL PROTECTED]
Date: Wed, 14 Mar 2007 15:18:35 +0100
I'm used to seeing the same versioning (maybe I've been gone too long)
Is zaptel 1.2.15 the right one for asterisk 1.2.16 ?
It works. I've tried some other mixes and they also work.
Yuan Liu
From: Yuan LIU [EMAIL PROTECTED]
Date: Tue, 06 Mar 2007 13:58:00 -0800
How to compile smsq in 1.2? It is compile in 1.4 by default. It is
included in 1.2.13, but not compiled. Any rule or method to make it?
Problem solved after upgrading to 1.2.16. Two points:
1) There was indeed a rule
is a bit more complicated, if you want to count for
local toll and toll-free numbers, etc.
Yuan Liu
Any help would be greatfull
Thomas Patterson
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). It interprets all extensions upon
reading extensions.conf.
Hope this helps.
Yuan Liu
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How to compile smsq in 1.2? It is compile in 1.4 by default. It is
included in 1.2.13, but not compiled. Any rule or method to make it?
Yuan Liu
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leg and hope the remote party picks up soon enough.
I thought call file extension will start execution only when the outgoing
leg is answered. Or is there some way to detect this?
Yuan Liu
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From: Doug Lytle [EMAIL PROTECTED]
Date: Sun, 04 Mar 2007 13:56:35 -0500
Yuan LIU wrote:
I noticed that [EMAIL PROTECTED] started to execute regardless of the state of
the outgoing call. Is this supposed to be? So far I can only set a
Wait() in the local leg and hope the remote party picks
Does application Read() return a status? Console displays stuff, but show
application read doesn't mention any status variable.
Yuan Liu
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of similar names. AGI is definitely not the choice in
Asteriskland.
Yuan Liu
Where can I find good Howto (with good explanation)?
--
#Joseph
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.
Apparently this AGISTATUS is a 1.4 thing, and probably still very
simplistic. Just wonder why all AGI commands carry sophisticated return
codes.
Yuan Liu
On Sat, 03 Mar 2007 06:28:23 +0100, Yuan LIU [EMAIL PROTECTED] wrote:
I mean how do I set failure condition in AGI? My script exits
problem with some regular phone devices, too.)
Yuan Liu
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From: Tzafrir Cohen [EMAIL PROTECTED]
Date: Fri, 2 Mar 2007 22:14:09 +0200
On Fri, Mar 02, 2007 at 09:36:28AM -0800, Yuan LIU wrote:
With one IVR payment system, I noticed quite a difference in DTMF
transmission between these two cards. The IVR missed nearly all digits
from X100P, while
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