May I ask : beside saving channels, what are the benefits of TBCT over
bridging calls inside Asterisk ?
What about caller ids ? I would say caller id should passed over to final
callee after bridging but that should need some kind of signaling update.
On Tuesday 14 April 2009, David Backeberg wrote:
With app_fax integrated into asterisk-1.6, you have an 'infinite'
modem pool that you control through the dial-plan. Using dialplan
variables you provide a filename to save the fax to, and you can use
other dialplan directives to describe what
I couldn't find any pointers for this online. Is it possible through Async AGI?
Let me know if anybody is aware of doing this.
From: Martin asteriskl...@callthem.info
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Hi,
I am about to integrate chan_misdn into asterisk version 1.4.22 with the
codec negotiation patch on an embedded platfrom. Has anyone ever integrated
chan_misdn with asterisk v1.4.22 + the codec negoiation patch, or any
version of asterisk + the codec negotiation patch? Is so, is there any
Hi
Last couple of days I received the subject WARNING message on a
home-based asterisk pbx.
Is someone spoofing a register method on port 5060? Or, is this warning
something random (sort of like sporadic alarms on an analog port)?
(This warning message is from chan_sip.c).
Am running asterisk
Hi,
I only run a home-based asterisk (v1.4.18), and have never
patched it, so I'm a unfamiliar with what time frame to
expect for patches being implimented.
I just downloaded (April 14) svn asterisk V1.6.1 r188415, on
a play machine and noticed that when I stop asterisk, the astcanary
module does
Sorry for the stupid question, but I think I'm not understanding
something. Why can't you use Fax for Asterisk with res_fax and
res_fax_digium?
Florian Hackenberger a écrit :
Thanks for the explanation! Sounds all good. There is one remaining
question however. As you mentioned T.30, is
2009/4/15 Olivier oza-4...@myamail.com
May I ask : beside saving channels, what are the benefits of TBCT over
bridging calls inside Asterisk ?
I'm not aware of anything apart from saving channels...
What about caller ids ? I would say caller id should passed over to final
callee after
2009/4/15 John covici cov...@ccs.covici.com
Its not there and the link you gave me says its for sip originating
rather than calls to a sip channel.
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
It's been around awhile. I've used it in 1.4 Check out this link for
-Original Message-
From: d...@tucny.com
Sent: Wed, 15 Apr 2009 19:44:23 +0800
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] 2B Channel Transfer on XO-based T1
2009/4/15 Olivier oza-4...@myamail.com
May I ask : beside saving channels, what are the benefits
Try voip-info.org. From a pseudo-code standpoint, here's how you would do
it:
1. start agi
2. agi creates fork for playing moh and dialing
3. when dialing reaches 180/183 status, moh fork is terminated
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
This will execute the macro, then dial the number. You will have to take
the hangups out of callback.
-Original Message-
From:
Call an AGI and return dnid to the dialplan. If it's not dnid, the value is
there, you'll just have to look a bit more (voip-info.org)
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ryan M.
Colbert
Sent: Tuesday, April
On Tuesday 14 April 2009 18:41, Jared Smith wrote:
Some time after the second leg of
the call has answered, Asterisk will send a facility message to the CO
switch saying Hey, mind bridging these two calls on your end, so I can
free up the channels on my end? If the switch says OK, you'll see
On Wednesday 15 April 2009 04:25:15 Gerald Harshany wrote:
Hi,
I only run a home-based asterisk (v1.4.18), and have never
patched it, so I'm a unfamiliar with what time frame to
expect for patches being implimented.
I just downloaded (April 14) svn asterisk V1.6.1 r188415, on
a play machine
Your box receives a 200 OK message as though it would have sent the
REGISTER sip message -
trying to register with a sip provider as a sip device.
Asterisk doesn't recognize it because:
1) the REGISTER was not sent from Asterisk
2) the 200 OK was sent too late
3) there's some other issue like
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
Here's how core show application dial says you should do it:
Change your dial to
exten = _X.,n,Dial(${DIALNUM},${ARG2},dtT,callback)
I'm not sure if this is correct. core show application dial says:
Ron Joffe schrieb:
On Tuesday 14 April 2009 18:41, Jared Smith wrote:
Some time after the second leg of
the call has answered, Asterisk will send a facility message to the CO
switch saying Hey, mind bridging these two calls on your end, so I can
free up the channels on my end? If the switch
pri debug span 1
should show you the ISDN messages for 2BCT if there are any
Also someone should have told you that the 2BCT code is by default not compiling
and you could enable it by editing chan_dahdi.c and adding
#define PRI_2BCT
Also since this flag is not present anywhere else in the
Hello all!, Ive running asterisk 1.4.23.1 and I need to get working pick up
from feature.conf. It does no work, only I can connect but cant send audio
over the phone. Is there a bug with this feature?. Thanks for any response!
Cheers!
Gustavo A. González
Dto. de Infraestructura
On Wed, Apr 15, 2009 at 4:47 PM, Christoph Fuerstaller
fuch_li...@kurtkrenn.com wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Danny Nicholas schrieb:
Here's how core show application dial says you should do it:
Change your dial to
exten =
Re Below:
This list is getting more and more useful! Build muscles somewhere besides
my finger tips. :-(
CF
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of VIAGRA R
Pfizer Inc.
Sent: Wednesday, April 15, 2009 8:56 AM
To:
Martin schrieb:
Also someone should have told you that the 2BCT code is by default not
compiling
and you could enable it by editing chan_dahdi.c and adding
#define PRI_2BCT
Could somebody shed some light on why PRI_2BCT is not enabled by
default? Is it an experimental feature?
I'd like
This is what you Really want; It should work with SIP or Zap
exten = _X.,1,Dial(${DIALNUM},${ARG2},tT)
exten = _X.-NOANSWER,1,background(press5tocallback)
exten = -X.-NOANSWER,2,waitexten(5)
exten = 5,1,goto(callback,s,1)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Gustavo A Gonzalez wrote:
Hello all!, I’ve running asterisk 1.4.23.1 and I need to get working
pick up from feature.conf. It does no work, only I can connect but cant
send audio over the phone. Is there a bug with this feature?. Thanks for
any response!
Cheers!
Yes there was.
Its not actually muscle as far as I know..
S
On 15 Apr 2009, at 15:07, Cary Fitch wrote:
This list is getting more and more useful! Build muscles somewhere
besides my finger tips. L
___
-- Bandwidth and Colocation Provided by
Philipp Kempgen wrote:
Could somebody shed some light on why PRI_2BCT is not enabled by
default? Is it an experimental feature?
I'd like to compile stuff without patching defines. :-)
It's not enabled by default because when it is used the Asterisk server
loses control of the call and the
Cary Fitch escribió:
Re Below:
This list is getting more and more useful! Build muscles somewhere
besides my finger tips. L
CF
I rarely see this type of spam messages on my list folder. Ask your
e-mail administrator for a decent spam filter. ;)
On Wed, 2009-04-15 at 09:58 -0500, Kevin P. Fleming wrote:
It's not enabled by default because when it is used the Asterisk server
loses control of the call and the CDR becomes incomplete. Not everyone
wants that behavior.
But since many people *would* like that behavior, wouldn't it make more
Hi,
On Mon, Apr 13, 2009 at 05:32:45PM -0400, John covici wrote:
Hi. I have a SIP provider which wants RFC2833 for the dtmfmode,
however I would like to increase the duration of the tone, its pretty
short and some IVR's are unhappy or don't detect it. I did poke
around, but it looks like
Well, this solution seemed not to work for me, maybe because I did not
set the minimum and also if I am using a sip phone or ATA, the
solution would not apply -- correct me if I am wrong on either of
these.
on Wednesday 04/15/2009 Mark G. Thomas(m...@misty.com) wrote
Hi,
On Mon, Apr 13,
Hi Moy,
You are right. I failed applying the patch. In fact, I applied it but I didn't
make install so I started a wrong asterisk. I apologize, it was my mistake.
This time I made sure twice before getting the logs and this time the log
message you said appears, but it doesn't work either as
Hi,
Setting the minimum was necessary in my case, and did affect
tones to and from SIP devices as well as the SIP provider, though
this was some time ago and you may have different results with
your setup than I did with mine.
Mark
On Wed, Apr 15, 2009 at 11:31:06AM -0400, John covici wrote:
i create inbound confi my confi is:
[incoming]
exten= 1246463,,1,Dial(SIP/8003,60,rT)
exten= 6463,1,Dial(SIP/8003,60,rT)
exten= 1246463,,n,Wait(5)
exten= 1246463,,n,Hangup
but y calling and send this error in my CLI:
[Apr 15 09:58:48] NOTICE[26985]: chan_sip.c:14383
On Wed, 2009-04-15 at 09:59 -0600, Bayardo Sanchez wrote:
i create inbound confi my confi is:
[incoming]
exten= 1246463,,1,Dial(SIP/8003,60,rT)
exten= 6463,1,Dial(SIP/8003,60,rT)
exten= 1246463,,n,Wait(5)
exten= 1246463,,n,Hangup
but y calling and send this error in my CLI:
On Wed, Apr 15, 2009 at 3:29 AM, Florian Hackenberger
f.hackenber...@chello.at wrote:
Thanks for the explanation! Sounds all good. There is one remaining
question however. As you mentioned T.30, is app_fax capable of
terminating T.38?
Yes although I'm speaking about 1.6. I can't say for
nothing send the error:
[Apr 15 10:30:54] NOTICE[26985]: chan_sip.c:14383 handle_request_invite:
Call from '101396_procall' to extension '246463' rejected because
extension not found
On Wed, Apr 15, 2009 at 10:11 AM, Jared Smith jsm...@digium.com wrote:
On Wed, 2009-04-15 at 09:59 -0600,
Ok, that makes more sense. Try this new patch and let me know how it
goes, once you confirm it works I will post it in my blog with a
better name.
http://moythreads.com/testasync.diff
Moy
On Wed, Apr 15, 2009 at 11:52 AM, cyr2...@gmail.com wrote:
Hi Moy,
You are right. I failed applying the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Danny,
Thanks for your replay. Jep, that would be a possibility. But then the user has
to wait until my dialtime is over. If he/she is that inpatient, then with my
solution he/she can end the dialing whenever needed. But, I'll try your
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
Atis Lezdins schrieb:
I think the limitation could be by analogous Zap phones, as they
probably don't support sending DTMF on unanswered channel. You could
try it opposite way - Dial from SIP phone to Zap.
Noop, it's not a Zap problem. I
Someone referred to a facility message when the TBCT call is torn down.
There are actually two messages--when the PSTN switch takes back the calls
and completes the transfer, it sends a facility message including a unique
ID. Then, when one of the parties disconnects, the switch sends another
You call call to extension '246463' will not match 'exten =
1246463'.
On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez
bayardo.sanc...@gmail.comwrote:
i create inbound confi my confi is:
[incoming]
exten= 1246463,,1,Dial(SIP/8003,60,rT)
exten= 6463,1,Dial(SIP/8003,60,rT)
exten=
i call my tollfree number and send the call to my extension 8003
On Wed, Apr 15, 2009 at 10:51 AM, Brandon B. bran...@brellsystems.comwrote:
You call call to extension '246463' will not match 'exten =
1246463'.
On Wed, Apr 15, 2009 at 9:59 AM, Bayardo Sanchez
Try this:
[incoming]
exten= 246463,1,Dial(SIP/8003,60,rT)
exten= 246463,n,Wait(5)
exten= 246463,n,Hangup
exten= 6463,1,Dial(SIP/8003,60,rT)
On Wed, Apr 15, 2009 at 11:00 AM, Bayardo Sanchez bayardo.sanc...@gmail.com
wrote:
i call my tollfree number and send the call to my
I can do it as a paid bounty if there's noone volunteering.
Would need access to the box with the live circuit including TBCT enabled.
PM me if interested
Martin
On Wed, Apr 15, 2009 at 10:45 AM, Don Kelly d...@donkelly.biz wrote:
Someone referred to a facility message when the TBCT call is
nothing
[Apr 15 11:24:15] NOTICE[26985]: chan_sip.c:14383 handle_request_invite:
Call from '101396_procall' to extension '246463' rejected because
extension not found.
On Wed, Apr 15, 2009 at 11:07 AM, Brandon B. bran...@brellsystems.comwrote:
Try this:
[incoming]
exten=
Is your system configured to send the dialed calls to the [incoming]
context?
On Wed, Apr 15, 2009 at 11:22 AM, Bayardo Sanchez bayardo.sanc...@gmail.com
wrote:
nothing
[Apr 15 11:24:15] NOTICE[26985]: chan_sip.c:14383 handle_request_invite:
Call from '101396_procall' to extension
tollfree calls was working fine but stopped working without any reason
On Wed, Apr 15, 2009 at 11:51 AM, Brandon B. bran...@brellsystems.comwrote:
Is your system configured to send the dialed calls to the [incoming]
context?
On Wed, Apr 15, 2009 at 11:22 AM, Bayardo Sanchez
For an Asterisk-environment with no more then 10 SIP-phones, I would
open 10 x 4 = 40 UDP ports for RTP/RTCP-traffic ( 4/call). Can you
confirm ?!
rtp.conf :
rtpstart=30500
rtpend=30550
Ok, there's 50 here... a round number right ?!
All SIP-communication stays on the LAN. There's a NIC
If you set your ARG2 to a value like 6, the phone would only ring twice
before noanswer.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christoph
Fürstaller
Sent: Wednesday, April 15, 2009 11:34 AM
To:
On Wed, Apr 15, 2009 at 09:24:54AM -0500, Danny Nicholas wrote:
This is what you Really want; It should work with SIP or Zap
exten = _X.,1,Dial(${DIALNUM},${ARG2},tT)
exten = _X.-NOANSWER,1,background(press5tocallback)
exten = -X.-NOANSWER,2,waitexten(5)
Anything after a '.' in a pattern
Mea culpa. Just being a bit lazy. In real use, the _X.-noanswer would be
s-NOANSWER (at least that's how it works in MY Dialplan).
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir Cohen
Sent:
May I suggest divide and conquer?
I haven't followed every detail, but it seems that your phones are not
registering.
Put them on the same net as the sip server and get them to register.
Then get it to where you can make a call from one to the other.
Then back off through your
On Wed, Apr 15, 2009 at 09:07:37AM -0500, Cary Fitch wrote:
Re Below:
This list is getting more and more useful! Build muscles somewhere besides
my finger tips. :-(
[ Original spam snipped. Subject slightly altered ]
If you actually choose to answer a spam message, don't do its author
Bayardo Sanchez wrote:
tollfree calls was working fine but stopped working without any reason
Oh, there's a reason.
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asterisk-users mailing list
To UNSUBSCRIBE or update
On Tue, Apr 14, 2009 at 4:52 PM, Florian Hackenberger
f.hackenber...@chello.at wrote:
On Tuesday 14 April 2009, Michael wrote:
asterisk-1.6 with app_fax built-in
Try 1.6. You'll be glad you did.
While I have not tried Asterisk 1.6 because I settled on Callweaver
at the time (which has
Hi,
I have a problem with TDM2400P card. The card is detected ok, I can make a call
but only with pulse dialing (not tone dialing) without hear sounds from the
headset. When I receive a call, I can to establish a communication, but without
hear sounds from the headset. When I dial any phone
- Original Message -
From: Tilghman Lesher tilgh...@mail.jeffandtilghman.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, April 15, 2009 9:28 AM
Subject: Re: [asterisk-users] astcanary not exiting in asterisk V1.6.1
On
I understand - Thanks for the reply.
Yes, I have been registering with the sip provider Voicepulse for about 2
years,
but never saw the message before. In the last 2 days or so it has popped up
about 5 times each of these days, which started me wondering what the
messages
really meant.
Gerald
Hi, excuse me, but I see in your code that you configure DAHDI with OSLEC.
How do you do? Which version you have installed?
Thank you.
Marco
2009/4/16 Giovanni Magallanes gmagalla...@gmail.com
Hi,
I have a problem with TDM2400P card. The card is detected ok, I can make a
call but only
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