Has anybody tried NVFaxDetect Fax detection for sip SIP/IAX channel?
There is a new application from Newman Telecom for fax detection.
http://www.sineapps.com/news.php?rssid=575
Current Asterisk Fax detection doesn't work for me as I don't have
Digium cards; I'm using Siupra
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#Joseph
I'm just curious why Sipura isn't using free IAX protocol with their
devices instead of SIP?
With IAX NAT traversal would have been easier, so why are they using
SIP.
Is there any politics in it?
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fixes.
Perhaps I have something wrong with my ntp settings, but the other
firmwares work just fine.
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Mine too :)
Thanks.
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No, the bracket wasn't needed or causing the problem.
It seem to me DISA does not know who to macro-tollfree context when
called so I added in [disa-access] context:
include = macro-tollfree
and it worked.
Though I can not explain why.
#Joseph
On Sat, 2005-03-19 at 15:04 -0800, Ed Greenberg
?
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was the
app_nv_backgrounddetect.c and no instruction how how to install it.
The installation instruction from wiki doesn't help as I'm missing the
two modules above.
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After downgrading to CVS stable on Gentoo from *-1.0.5 my DTMF is not
working.
When I call-in and dial an extension phone is not ringing, same is with
password for my mail box is not recognized.
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Has anyone tried compiling the zaptel stuff under the 2.6.11 kernel?
I get all kinds of errors when doing that.
On the 2.6.10 it works fine.
This is using cvs.
Also, any news on when a 1.2 will be released?
Or even available for download? :)
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I didn't do was: make samples
I did not generate new configuration files (maybe I should have).
Everything compiled without any problem.
#Joseph
On Tue, 2005-03-22 at 16:04 -0500, C F wrote:
How are you calling in? what phone are you using? how is the phone
connected to your asterisk box
to troubleshoot DTMF, if so how?
I would think that Stable version should solid.
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My DTFM is not working in current CVS-stable *-1.0.6 and *-1.0.7 but it
works in version 1.0.5 (was working with 1.0.3).
I'm using SPA-3000 and dtmfmode=inband
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If this is the case it would seem to me that chan_sip.c is buggy.
Where did you get R2 version? I'll try it.
I don't understand how such a major bug got into the CVS-Stable branch.
#Joseph
On Wed, 2005-03-23 at 17:00 -0800, Bashir Ullah - www.Lamsre.Com wrote:
Hi
I am not good at coding
This is problem with the SIP chanell in asterisk; my SPA-3000 inband is
not working either so I stay with 1.0.5
#Joseph
[snip]
I noticed this in testing the 1.0.7 Release Candidate w/ my
Polycom phones. I posted the following in
http://bugs.digium.com/bug_view_page.php?bug_id=0003746
feature ATA will have more
advantages over PCI card.
Just makes me wander why Digium doesn't invest in ATA modules?
They have all the hardware so what is the problem putting them into one
box and make ATA
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.
How about a unit that has more ports, fax signal detection etc.
For example:
1-FXO and 4-FXS
or
2-FXO and 6-FXS
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know about it.
So when * answers the call adjust ringing time to whatever you want
(5min?). The calling party will hear ring to tone and will never know
that the phone was picked up by asterisk.
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Does anybody use ATCOM Gateways: G-168, AG-248, AG-468
Any positive/negative commands on them?
I know the AG-468 is not out yet (or new item).
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://www.eezeephone.com or at http://voip-info.org/wiki-ATCOM
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to
this tread.
Thanks,
--Joseph
Tim Bass wrote:
Tom Ivar Helbekkmo. Grow up and stop posting to this tread.
Nobody cares about your bullying insults.
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I got mine from http://sipphone.com/dlink/
Works great, it's their unlocked version as well. Connect it to their
network when you first get it and leave it alone for 10 or 20 minutes,
and every month or so(it gets it's firmware upgrades from there).
If not, you can check out
Regards,
Stefan
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to chan_sccp, but don't know if anyone would use it.
Would this not make the 7914 more useful as well?
I would think that this is a very useful addition.
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Does anyone have the sip subscribe working with the IP600 phones?
I saw a message about the Buddies, but I could not tell if it works.
Thanks.
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reason.
I would like to get min. two port unit. Is it possible to use IAX
protocol without hardware unit that supports IAX?
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I'm still using asterisk v. 0.9 I think it is using IAX1 as the ixa.conf
is set port=5036
Can I register with a provider who is using IAX2 ?
When I set it up and run:
iax2 show registry - it is not displaying any registered provider.
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When loading iax.conf I get warning:
WARNING[884753]: chan_iax2.c:5631 set_config: Ignoring port for now
In iax.conf I have:
[general]
port=5036
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anybody if I'm wrong.
I'm located in Alberta Canada so my chases are even smaller.
I've another incoming fax line, so I guess I could set it somehow as an
alternative incoming line if my main line is busy.
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external IP before firewall isn't it?
When I try to register, I get: sip_reg_timeout: Registration for ...
What am I missing?
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Joseph,
1 - 0.9 still uses IAX2 (I think - pretty sure).
2 - Why are you using 0.9?
Maybe they should be the other way around...
I'm just using default installation whatever Gentoo is providing; this
is their stable version.
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I have in iax.conf
register = name:[EMAIL PROTECTED]
but I can not make a call, it hangs up on me.
How can I check if I'm registered with iaxtel?
What do I have to have in iax.conf in order to register?
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-voipjet]
exten = _1NXXNXX,1,SetCallerID(4757894789);
exten = _1NXXNXX,2,Dial,IAX2/[EMAIL PROTECTED]/${EXTEN}
Whatever I try to dial it goes through voipjet.
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I'm just using default installation whatever Gentoo is providing; this
is their stable version.
Joseph,
While I also use Gentoo(as do many others), most will tell you NOT to
install * from portage. You can save yourself trouble by getting 1.0.3
or CVS and ditch the builds
On Mon, 2005-01-17 at 09:02 +0100, Jens Vagelpohl wrote:
On Jan 17, 2005, at 7:29, Joseph wrote:
How do I solve the problem with between patterns:
_1800
_1NXX
I would like all numbers 1800, 1877 etc to go through iaxtel
but all other numbers 1xxx via voipjet
When you combine
1800 calls via iaxtel?
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On Mon, 2005-01-17 at 12:20 -0600, Eric Wieling aka ManxPower wrote:
Joseph wrote:
What is the best codex for iaxtel?
When I set in iax.conf
bandwidth=high
disallow=all
allow=ulaw
The call will not go through, if I set allow=all
it sets the format to ADPCM and the first 15sec
I've tried gsm but the call doesn't go through.
bandwidth=high could be screwing it up.
Post the CLI output of the failed call.
Executing Dial(SIP/11-0b9e,
IAX2/joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED]) in new stack
-- Called joseph:[EMAIL PROTECTED]/[EMAIL PROTECTED
: ADPCM?
PS. It seems to me iaxtel has a problem with connection today, can
anybody confirm it?
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-voipjet context. The
system should stop at the first match.
Good luck,
Robert Jackson
Thank you!
I was under impression that the order in extension.conf is important but
actually it is iax/sip.conf file.
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the extension to certain number as I don't know what
number I will be dialing.
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/asterisk/mohmp3,-z
specify in you extension.conf what you want to play, example:
[office-open]
exten = s,1,Wait(2)
exten = s,2,Answer()
exten = s,3,SetMusicOnHold(loud)
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explain to me what passcode is used for?
If I enter no-password I can make a call but if I enter any number
instead of word passcode it will not let me IN.
Is passcode a second level password; the asterisk is not prompting me to
enter anything.
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it will not let me IN.
How do you use it?
Does it need to be used in connection with a file?
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authenticationdoes not exist
When I try to call out I get:
Called 495771:[EMAIL PROTECTED]
WARNING[114696]: chan_iax2.c:4534 socket_read: Call rejected by 65.39.205.121:
No such context/extension
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On Tue, 2005-01-25 at 18:27 -0700, Joseph wrote:
I'm trying to test IAX2 with FWD
It registers fine but when I try to receive the call I get:
chan_iax2.c:476 iax_error_output: Ignoring unknown information element
'Unknown IE' (38) of length 1
Jan 25 18:02:12 WARNING[114696]: chan_iax2.c
this file already.) If not, you can wget it from
this link.
The link is http://downloads.fwdnet.net/freeworlddialup.pub
Kris
Thanks Kris, I've missed that section.
I need learn to upgrade my outdated asterisk with cvs, Gentoo is too
much behind in stable version.
--
#Joseph
by 65.39.205.121 (format ULAW)
-- Format for call is ULAW
-- Hungup 'IAX2[65.39.205.121:4569]/3'
Dialing *1800... doesn't do anything.
What am I doing wrong?
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On Wed, 2005-01-26 at 00:17 -0500, Kris Stark wrote:
Joseph wrote:
I've setup my IAX2 over FWD and it is working I can receive a test call
and I can call out.
Though I cannot figure out how to dial 1-800 numbers over FWD
When I dial 1-800 it hangs up on me.
Here is a typical session
On Wed, 2005-01-26 at 18:17 +1100, Duane wrote:
Joseph wrote:
Thanks Kris, I found the solution:
Here is how it suppose to look like:
You can minimise all that with a simple macro and a little pattern
matching, and it makes dial plans so much easier to track down problems
with etc
On Wed, 2005-01-26 at 18:17 +1100, Duane wrote:
Joseph wrote:
Thanks Kris, I found the solution:
Here is how it suppose to look like:
You can minimise all that with a simple macro and a little pattern
matching, and it makes dial plans so much easier to track down problems
with etc
jumps to
priority 102?
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exits with is 0; so next
priority is executed n+101 (that is easy)
What if the priority is dial returns is -1 what next? How to jump to
next line?
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unit.
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When I use my phone to make VOIP call and another calls comes from POTS
my phone rings to POTS caller. Why?
Shouldn't it generate busy signal!
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On Thu, 2005-01-27 at 08:40 +, Tony Mountifield wrote:
In article [EMAIL PROTECTED],
Joseph [EMAIL PROTECTED] wrote:
When I use my phone to make VOIP call and another calls comes from POTS
my phone rings to POTS caller. Why?
Shouldn't it generate busy signal!
Yes
Were you able to make it to work ChanIsAvail application?
I have a similar problem.
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On Wed, 2004-12-15 at 14:49 -0800, voipbuilder wrote:
Hello Everyone,
I am trying to use the ChanIsAvail application but I am not getting
the results I expect when making calls...
exten
%
20ChanIsAvail#comments
The channels are checked in the order listed, returning the first
available channel in the list in ${AVAILCHAN}.
so when my SIP/21 is available and it is it should ring it but it is
not.
What am I doing wrong?
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#Joseph
instruction instruction.
According to notes:
The channels are checked in the order listed, returning the first
available channel in the list in ${AVAILCHAN}.
so when my SIP/21 is available, and it is, it should ring it but it is
not.
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I dial UPS tall free my phone2 will ring as phone1 is busy.
I can not explain why???
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copied the Sip.conf; extenson.conf and iax.conf from old version to
1.0.5
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When I try to reload configuration *-1.0.5 can not find my sip.conf.
I don't see anything like:
== Parsing '/etc/asterisk/sip.conf': Found
WARNING[10301]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 102
(Non-critical Request)
--
On Sat, 2005-01-29 at 00:38 -0700, Joseph wrote:
When I try to reload configuration *-1.0.5 can not find my sip.conf.
I don't see anything like:
== Parsing '/etc/asterisk/sip.conf': Found
WARNING[10301]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED
=fromiaxfwd
auth=rsa
inkeys=freeworlddialup
disallow=all
allow=ulaw
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executed
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Conflict with with codec.
bandwidth=low should be disabled as ulaw is not a low bandwidth codec.
#Joseph
On Sat, 2005-01-29 at 13:13 -0700, Joseph wrote:
When I try to call out to FWD over IAX2 I get:
Call rejected by 65.39.205.121: Unable to negotiate codec
I'm using asterisk-1.0.5
a number?
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On Mon, 2005-02-07 at 17:02 -0500, Andrew Thompson wrote:
Joseph wrote:
Is it possible to establish communication over VOIP between two asterisk
servers without going through any FWD etc. service?
If so how to ring it if I know the IP address.
The answer to your question is most
it iax.conf is not loading.
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Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
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On Thu, 2005-02-10 at 01:22 +1100, Duane wrote:
Joseph wrote:
Can anybody confirm if IAX on FWD is down again?
I can not register IAX with FWD.
I got fed up with the yo-yo, which then led me to dump fwd and install
asterisk and start playing with inter-asterisk routing via e164.org
[snip]
It is working again :-)
It appears something broke after recent upgrades (on Gentoo) as I wasn't
even able to dial VOIPJET, though I don't know what was broken :-/
I re-emerged asterisk and it is working gain.
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I wander what is causing the problem, I was thinking
and problem was corrected; but this is not a
solution since I still don't know what was broken and there were no
error messages on the command line.
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[snip]
Note that this service is unmonitored and unsupported, and we periodically
use it for testing purposes. However, if enough people
What dtmfmode should I set for IAX protocol?
When I dial FWD over IAX it doesn't recognize the numbers I press.
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On Thu, 2005-02-10 at 12:15 -0600, Eric Wieling wrote:
Joseph wrote:
What dtmfmode should I set for IAX protocol?
When I dial FWD over IAX it doesn't recognize the numbers I press.
IAX and IAX2 do not support a DTMF mode option. They use out of band
DTMF *ALWAYS*.
According to IAX
IAX and IAX2 do not support a DTMF mode option. They use out of band
DTMF *ALWAYS*.
So what you are saying I can not press (# 1, 2 etc) when I dial
somewhere and ask to press a number?
Is there a solution for it?
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of the option provided
it doesn't work.
Could it be their phone system?
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On Thu, 2005-02-10 at 21:36 -0600, Michael Giagnocavo wrote:
Actually, there are some phones that will do inband DTMF over IAX2. So if
he's using one of these, he has to make sure his settings are correct. Yes,
the PA168
On Fri, 2005-02-11 at 00:17 -0600, Rich Adamson wrote:
Joseph has been working at bringing up an asterisk box as kind of a
newbie, and I think he's using a Sipura as his fxs interface into
asterisk. He's having a problem with asterisk passing dtmf to FWD,
but didn't say how he's accessing
INFO
Auto
InBand+INFO
AVT+INFO
There is no Out-of-band setting.
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to FWD, recording is different.
But again the same thing is with FedEx I'm connected to two different
call centers I think, but everything works with FedEx.
Calling DHL over IAX and FWD, DTMF doesn't work either but going though
PSTN line on Sipura-3000 works fine.
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I'm testing new cordless Motorola phone and the handset is constantly
displaying message: MSG WAITING OFF
According to the manual this message suppose to go off but it doesn't
Is it something that I can control via * ?
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I'm testing new cordless Motorola phone and the handset is constantly
displaying message: MSG WAITING OFF
According to the manual this message suppose to go off but it doesn't
Is it something that I can control via * ?
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According to Motorola support the communication has to be setup at
1200pbs.
Does anybody have an idea how to do it?
The phone is connected to Sipura-3000
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On Fri, 2005-02-18 at 11:42 -0700, Joseph wrote:
I'm testing new cordless Motorola phone and the handset is constantly
On Fri, 2005-02-18 at 12:32 -0800, Trevor Peirce wrote:
Joseph wrote:
I'm testing new cordless Motorola phone and the handset is constantly
displaying message: MSG WAITING OFF
According to the manual this message suppose to go off but it doesn't
Is it something that I can control via
),
very nice phone with a lot of features, and no more problems with MSG
Waiting, the quality of the voice is much better as well.
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that phone doesn't even have a webpage
offering firmware upgrade. So compare Sipura new phone feature with
this one. At least Sipura is offering constant firmware upgrade; you
will most likely never see one for this one.
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I think you are trying to use SIP with FWD, isn't it?
Set FWD over IAX, follow the instructions:
http://www.fwd.pulver.com/advanced/iax
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#Joseph
On Sun, 2005-02-20 at 18:12 -0600, Anton Krall wrote:
Guys.
Im trying to figure out how to confgure FWD and NAT. I tried some configs
Try to analyze this link: Asterisk - Dual -Server:
http://www.voip-info.org/wiki-Asterisk+-+dual+servers
#Joseph
On Mon, 2005-02-21 at 15:41 -0700, [EMAIL PROTECTED] wrote:
Hello,
Can anyone help with this please?
thx,
chuks
Original Message
= i,2,Playback(pbx-invalid)
exten = i,3,Hangup()
But this doesn't work when I press any non-existent extension I get
congestion.
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#Joseph
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I can not make a call pickup to work with Sipura-3000.
I have one SIP phone and one is connected to ATA Sipura-3000
I've in all sip.conf context
callgroup=1
pickupgroup=1
in features.conf I've tired:
pickupexten = *88
pickupexten = *8
Nothing works.
What am I missing?
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#Joseph
features.conf contains
any define recording options.
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#Joseph
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. has passed and the call wasn't disconnected nor I hear any warning
to message how many minutes are left.
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#Joseph
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On Sat, 2005-02-26 at 18:22 -0700, Joseph wrote:
I can not make a call pickup to work with Sipura-3000.
I have one SIP phone and one is connected to ATA Sipura-3000
I've in all sip.conf context
callgroup=1
pickupgroup=1
in features.conf I've tired:
pickupexten = *88
pickupexten = *8
; Disconnect
;automon = *1 ; One Touch Record
;atxfer = *2 ; Attended transfer
but even adding it and commenting out automon = *1 didn't work.
and of course I restart asterisk after modifying features.conf
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#Joseph
] to work
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#Joseph
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On Sun, 2005-02-27 at 15:53 -0600, Eric Wieling wrote:
Joseph wrote:
Note: I added all this section manually, when I compiled * 1.0.5 this
section wasn't there (I don't know why).
[featuremap]
;blindxfer = #1; Blind transfer
;disconnect = *0
use realtime asterisk.
Translate it to a time value, whatever you charge per minute or $5.00 is
worth,
Use L option in Dail string, example:
exten =
_9NXX,4,Dial(SIP/${EXTEN:[EMAIL PROTECTED],60,trL(60:24:18))
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#Joseph
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