On Mon, Apr 13, 2009 at 6:59 AM, wrote:
> Hi Moy,
>
> thanks a lot for your fix, but I'm afraid it doesn't work. I looked your
> patch over and I realize the code never passes by neither of the two lines
> you added with "returnstatus = AGI_RESULT_HANGUP". Even, it seems the
> execution doesn'
Miguel Molina writes:
[...]
> Why don't you just make your billing statistics from the queue log?
> Assuming that you upload the queue log to a database, it would be very easy.
I took a look at this option and it looks like it will work. Thanks
to all who responded for your advice!
Scott
[Apr 15 11:12:19] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR
transmission error to aaa.bbb.ccc.ddd:37259, rtcp halted Operation not
permitted
[Apr 15 11:12:23] ERROR[2624]: rtp.c:2447 ast_rtcp_write_sr: RTCP SR
transmission error to aaa.bbb.ccc.ddd:38563, rtcp halted Operation not
permi
Hi,
I'm using it on 1.4. Actually, the driver does all the work. I have clients
calling in, give them DISA, they dial the number and the call is pushed to the
switch.
Jimmy
> -Original Message-
> From: m...@povo.com
> Sent: Tue, 14 Apr 2009 17:52:23 -0400
> To: asterisk-users@lists.dig
Khaled W. Chehab wrote:
> Any idead on how to begin with AGI
>
>
That I don't.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
___
-- Bandwidth
On Tue, 2009-04-14 at 17:52 -0400, Max Metral wrote:
> I’m trying to get “blind transfer” from an incoming DAHDI line to an
> external number to work on an * 1.6 install using a T1 from XO. The
> documentation is very “distributed” and incomplete, so while it’s not
> working, it’s definitely more
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
> basic info: http://www.voip-info.org/wiki/v
On 4/14/09, Max Metral wrote:
>
> I’m trying to get “blind transfer” from an incoming DAHDI line to an
> external number to work on an * 1.6 install using a T1 from XO. The
> documentation is very “distributed” and incomplete, so while it’s not
> working, it’s definitely more likely my error some
I know what TBCT is, but haven't used it with Asterisk yet. Hopefully will
be soon.
1) Supposed to be supported by Asterisk. Is supported by XO (I use it with a
different platform).
2) Dunno
3) Dunno
4) The ISDN spec sez that one of the calls needs to be active to initiate
the transfer.
--D
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Hash: SHA1
Hi,
Atis Lezdins schrieb:
>
> Ok, at first glance the app_macro looks suspicious, can You try
> calling dial without Macro?
Tried it without macro -> same behavior.
>
> If unsuccessful, You could enable debug level 2, it will tell way much
> more of
I'm trying to get "blind transfer" from an incoming DAHDI line to an
external number to work on an * 1.6 install using a T1 from XO. The
documentation is very "distributed" and incomplete, so while it's not
working, it's definitely more likely my error somehow. Couple questions if
anybody is out
On Tue, 2009-04-14 at 09:45 -0700, awerf...@hotmail.com wrote:
> Hi Friend,
>
> How are you doing recently?
I'm getting bored.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users mailing list
To UNSUBSCRIBE or upda
On Tue, Apr 14, 2009 at 11:11 PM, Christoph Fürstaller
wrote:
> Thanks for the hint. I've looked aht the full log. I've attached a snipplet
> from the file. But I can't see anythin which can help me. Very interesting,
> but not helpful for me : / Is it possible to deactivate the 'd' option? Or
It's been around awhile. I've used it in 1.4 Check out this link for
basic info: http://www.voip-info.org/wiki/view/Asterisk+cmd+SIPdtmfmode
John covici wrote:
> Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
> Is this new in 1.6?
>
> on Tuesday 04/14/2009 Brent Davidson
Y, it can be that someone wants to register with a sniffed SIP packet.
it's basically the nonce="" value is not the same Asterisk sent for
that REGISTER session
Martin
On Tue, Apr 14, 2009 at 11:10 AM, Danny Nicholas wrote:
> http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.htm
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi,
Atis Lezdins schrieb:
> That's CLI interface output, log should have timestamps and much more
> detail in it.
>
> Check /var/log/asterisk/full (assuming default install location).
> You'll need to enable "full" line in logger.conf, restart Asteri
On Tue, Apr 14, 2009 at 9:14 PM, Christoph Fürstaller
wrote:
> -BEGIN PGP SIGNED MESSAGE-
> Hash: SHA1
>
> Hi Atis,
>
> No problem : ) I tried it again, here is the log output:
> -- Executing [...@from-pbx:1] Set("Zap/31-1", "EXITCONTEXT=callback") in
> new stack
> -- Executing [...
Any idead on how to begin with AGI
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 10:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [ast
Thanks -- can not find sip dtmf mode or sip dtmfmode in asterisk-1.4.
Is this new in 1.6?
on Tuesday 04/14/2009 Brent Davidson(br...@texascountrytitle.com) wrote
> To the best of my knowledge, the only way for you to control the
> duration sent to the PSTN lines is for you to be directly connec
Khaled W. Chehab wrote:
> Man :)
> I want the MOH play until Asterisk receives 180 ringing or 183 from the
> termination GW.
>
I don't think you'll be able to mix and match via the dial application.
You may have to try using AGI for this. That, I can't help you with.
Doug
--
Ben Frank
Man :)
I want the MOH play until Asterisk receives 180 ringing or 183 from the
termination GW.
Here I want to stop the MOH and let the user hear the early media RBT
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
John Todd wrote:
> Sorry for the brief interruption. The user spamming the list with
> vacation replies has been removed.
>
>
Bless you!
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
Khaled W. Chehab wrote:
> Thanks for answering Doug
>
>
> I am using exten => _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
>
Change this to:
_X.,n,Dial(SIP/OPNS/${EXTEN}|300)
The m was causing the music on hold.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Libert
This worked:
Dial(SIP/5198881...@telasip-gw4,,D(12345678))
>
> however, the problem now exists in the disconnection. Asterisk tries to
> bridge the call, play dtmf but never disconnects. What is there a specific
> syntax to the D command that specifies a disconnect period.
> I am thinking a better
On Tuesday 14 April 2009 13:04:02 jonas kellens wrote:
> -A RH-Firewall-1-INPUT -j REJECT --reject-with icmp-host-prohibited
> -A RH-Firewall-1-INPUT -p udp -m udp --dport 5060 -j ACCEPT
For one, these two rules are reversed. For two, you've failed to create holes
in your firewall for UDP/1-2
Your problem is that you put your line after the REJECT line. Your line is
never reached. Move it up one line, before the REJECT, and it will work as
expected.
// T
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kelle
I have a function *1 that starts and stops recording in a call. I use a
function so I can use MixMonitor.
It works well, however I would like to make it a little more integrated for
my users. We have GXP 200 hardphones. So far I've been able to configure a
softkey using the speeddial option to dial
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
No problem : ) I tried it again, here is the log output:
-- Executing [...@from-pbx:1] Set("Zap/31-1", "EXITCONTEXT=callback") in
new stack
-- Executing [...@from-pbx:2] Dial("Zap/31-1", "SIP/236||d") in new stack
-- Called 236
On Tue, Apr 14, 2009 at 10:28:42AM +1000, David Klaverstyn wrote:
> Hi All,
>
> I'm in the process of writing an install script and I would like to change
> some settings for the install process but I don't want the user to go into
> menuselect and make the changes manually.
>
> Is there a way
Anyone have a asterisk pager script that dials a list of pager numbers,
enters the phone number, and enters pound to send the page?
I am looking for a perl or php solution. Ideally I would like to set the
pager area-code and prefix on the fly as the company I work for owns a few
blocks of pagers
Thanks for answering Doug
I am using exten => _X.,n,Dial(SIP/OPNS/${EXTEN}|300|m) with no macros
kindly can you wrote down a macro to stop the MOH RTP in order to let the
GW inband early media rtp heard by the caller
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium
>
> Thanks for your replay. But in my 1st post, I mentioned my dial statement:
> exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
>
> As you can see, there is a d to exit the dial application. And one priority
> earlier, I set the EXITCONTEXT variable. So everything _should_ work, but it
> doesn't : /
Sorry for the brief interruption. The user spamming the list with
vacation replies has been removed.
JT
---
John Todd email:jt...@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6
_
From: Cary Fitch [mailto:ca...@usawide.net]
Sent: Tuesday, April 14, 2009 1:17 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [asterisk-users] Asterisk-beginner : cannot make phone calls
using Asterisk
May I suggest divide and conquer?
I hav
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Atis,
Thanks for your replay. But in my 1st post, I mentioned my dial statement:
exten => _X.,n,Dial(${DIALNUM},${ARG2},dtT)
As you can see, there is a d to exit the dial application. And one priority
earlier, I set the EXITCONTEXT variable. So e
On Tue, Apr 14, 2009 at 9:09 PM, Jim Dickenson wrote:
> At least in version 1.6.0.x you can specify a macro to be executed when the
> agent answers the queued call. This is an argument to the queue application.
>
> Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,g
> osu
CLI> core show application Dial
d- Allow the calling user to dial a 1 digit extension while waiting for
a call to be answered. Exit to that extension if it exists in the
current context, or the context defined in the EXITCONTEXT variable,
if it exists.
Reg
At least in version 1.6.0.x you can specify a macro to be executed when the
agent answers the queued call. This is an argument to the queue application.
Queue(queuename[,options[,URL][,announceoverride][,timeout][,AGI][,macro][,g
osub][,rule])
The optional macro parameter will run a macro on the
There is something wrong with my IPtables !!!
When i do :
service iptables stop
I see my phones register on the CLI !!
I can place a call and the phone rings !! I see a whole lot of
SIP-requests on the CLI with SDP-message in body !! That's good news...
What is wrong with my IPtables-rule I've
I will summarize everything again and try to answer all the questions
asked while I was away.
First I stop Asterisk :
[r...@asterisk asterisk]# /usr/sbin/asterisk -r
Asterisk 1.4.24, Copyright (C) 1999 - 2008 Digium, Inc. and others.
Created by Mark Spencer
Asterisk comes with ABSOLUTELY NO WAR
Khaled W. Chehab wrote:
> Dear Ben,
>
> I tried a lot ,Kindly can you give me an example on how to do that using a
> macro.
>
> Remove the 'm' out of your dial command:
>
> Doug
>
>
I'm not Ben, but I'll answer.
Shows us what your macro looks like and we'll chime in with some pointers.
Do
Try to parser queue_log file in real time and catch the event CONNECT
Regards,
Luis Morales
On Tue, Apr 14, 2009 at 12:44 PM, Ryan M. Colbert
wrote:
> Is there a way in the dialplan to figure out which agent in a ring all queue
> answered a line? I’d like to take specific action based on the ag
I have a same problem i add to the question
Francisco
Ryan M. Colbert wrote:
>
> Is there a way in the dialplan to figure out which agent in a ring all
> queue answered a line? I’d like to take specific action based on the
> agent upon hangup.
>
> Ryan M. Colbert
> Director of Information Tec
Is there a way in the dialplan to figure out which agent in a ring all queue
answered a line? I'd like to take specific action based on the agent upon
hangup.
Ryan M. Colbert
Director of Information Technology
Rissman, Barrett, Hurt,
Donahue & McLain, P.A.
201 E. Pine Street, Suite 1500
Orlando,
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Hash: SHA1
Hi,
Thanks for your replay. But this can only be done before or after the dial, but
I wanna do it during the dial, when user A is waiting for user B, answering the
phone. This should be possible, right?
I hope anyone knows if this is possible.
Chr
David Backeberg wrote:
> What I was specifically getting at in the context of that response was
> a comparison of dynamic modem pool versus fixed-size modem pool. When
> faced with the choice between a fixed-size modem pool or one that
> would grow or shrink dynamically with demand, I think the dyn
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Dear Ben,
I tried a lot ,Kindly can you give me an example on how to do that using a
macro.
Regards
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent: Tuesday, April 14, 2009 5:53 PM
To: Aster
Sorry - this is a bit off topic, but there is almost certainly someone
here who will know the answer... Perhaps even a snom employee :)
In recent snom firmware releases, the following sequence always causes
a call to be sent from line 'n'
Receive call on Line 'n' (where n > 1)
Press Hold
Di
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Let's just simplify this a LOT:
Your phones have no dialtone. This means they are not registering
with asterisk. I see in your sip.conf, for both you phones, you have:
host=X.X.X.X
If you specify an address here, your phones will not register.
Instead, to make your phones register, set it to:
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Steve Underwood wrote:
> Lee Howard wrote:
>
>> David Backeberg wrote:
>>
>>
>>> It may be possible to use hylafax, but
>>> I don't know how or why you would.
>>>
>>>
>> The reason *why* is generally due to support issues.
>>
>> For one, HylaFAX probably has a better T.30 imp
on Tuesday 04/14/2009 Kristian Kielhofner(kristian.kielhof...@gmail.com) wrote
> On Mon, Apr 13, 2009 at 5:32 PM, 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
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
On Mon, Apr 13, 2009 at 5:32 PM, 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 when RFC2833
awerf...@hotmail.com wrote:
>
> Hi Friend,
>
Oh boy, this is fun.
Doug
--
Ben Franklin quote:
"Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty nor Safety."
___
-- Bandwidth and Colocation
Since we don't know if he's having his phone's register, why not have
asterisk try to register them? If not, just another dumb suggestion from
me...
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edward
http://lists.digium.com/pipermail/asterisk-users/2005-July/110220.html
It means that a SIP device is re-using an old authentication challenge.
If it still registers and can place calls, there's no problem to worry
about. It's just a warning.
-Original Message-
From: asterisk-users-boun..
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi masters!
I've this Asterisk 1.4.15 running. yesterday I had to change the
firewall schema that I had before.
I use to have a FW that would be my network FW/Proxy and do the NATs
for Asterisk. This FW was receiving too many requests from my LAN and
it was making the Asterisk 'cut' the calls or
On Tue, 14 Apr 2009, Danny Nicholas wrote:
> Put register=yes in the BT201 and GXP1200 contexts of sip.conf
I admit to being a 1.2 Luddite, but isn't "register" asking your Asterisk
server to register with another endpoint? As in:
register => ::[authid]@/
Or is this some sort of 1.4/1.
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Tim Dobson wrote:
> 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample
> as ffmpeg borks at it:
Gah I meant
sox 1239101491.30.gsm -ql -r 64000 -t wav 1239101491.30.conv.wav resample
--
Thanks everyone! Really appreciate it - all the documentation via google
is to convert *to* g
Lee Howard wrote:
> David Backeberg wrote:
>
>> It may be possible to use hylafax, but
>> I don't know how or why you would.
>>
>
> The reason *why* is generally due to support issues.
>
> For one, HylaFAX probably has a better T.30 implementation in its Class
> 1 driver than does app_fax.
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Steve Edwards had (IMO) the best answer to this. Here is an example of
how-to in a regular dialplan:
[sat]
exten => s,1(start),Noop(in test Section)
exten => s,n,AGI(satver.agi|)
exten => s,n,Set(Play1=record/silence)
exten => s,n,Set(Play2=record/silence)
exten => s,n,Set(Press1=record/sile
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
Hi Friend,
How are you doing recently? I would like to introduce you a very good company which I know. Their website is www.myewell.com. They can offer you all kinds of
Electronic products like laptops, gps,TV LCD,cell phones,ps3,MP3/4, etcPlease take some time to have a check, There must h
just for a test, run "service iptables stop" as root on the asterisk
server and then reboot your phones. after that, try again and see if
the phones are making communications with asterisk.
you can turn the firewall back on with "service iptables start"
jonas kellens wrote:
> Hi there,
>
> th
On Tue, Apr 14, 2009 at 10:46 AM, Lee Howard wrote:
> David Backeberg wrote:
>> It may be possible to use hylafax, but
>> I don't know how or why you would.
>
> The reason *why* is generally due to support issues.
What I was specifically getting at in the context of that response was
a comparison
I'd change callback to this
[callback]
Exten => s,1,Playback(press5msg)
Exten => s,n,Waitexten(5)
Exten => s,n,Hangup
exten => 5,1,agi(str_concat.sh)
exten => 5,n,Hangup
This will play a message, wait 5 seconds for user to press 5, then hangup if
they don't.
-Original Message-
From: aster
To the best of my knowledge, the only way for you to control the
duration sent to the PSTN lines is for you to be directly connected to
the lines so you can set the tone duration in zapata.conf / dahdi.conf
or to use inband signalling.
One thing you might try is researching the "SipDtmfMode" co
Put register=yes in the BT201 and GXP1200 contexts of sip.conf
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 11:19 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-use
This is what I derive from your emails so far:
1. Asterisk is up and running
2. you have two SIP Peers that are running but not registered (sip show
peers shows not monitored, phones don't go off-hook, dialtone)
3. I've experienced similar problem with a X-Lite softphone. When the
You are doing 210 (enter), 210(dial) or 210 (#)? You have to engage the
dialer in some fashion.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 12:20 PM
To: asterisk-users@lists.di
In sox 14.0.1 the -1 is a one byte sample, -2 is a two byte sample,
therefore the command is sampling one-in, two-out.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, April 14, 2009
You are closing in. what does users.conf look like?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Monday, April 13, 2009 3:40 PM
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Asteri
Khaled W. Chehab wrote:
> Dears
>
> -How can I stop MOH when status of the dial is ringing and let the user hear
> the Ring Back Tone from the termination Gateway.
>
Remove the 'm' out of your dial command:
m([class]) - Provide hold music to the calling party until a requested
chann
David Backeberg wrote:
> It may be possible to use hylafax, but
> I don't know how or why you would.
The reason *why* is generally due to support issues.
For one, HylaFAX probably has a better T.30 implementation in its Class
1 driver than does app_fax. At least that historically has been true.
Scott Gifford escribió:
> Hello,
>
> I'm working on an Asterisk configuration for a call center, and they
> bill based on the time spent talking to an agent, but not for any time
> spent waiting in a queue. The CDR information contains the entire
> duration of the call as billable seconds, includi
Dears
-How can I stop MOH when status of the dial is ringing and let the user hear
the Ring Back Tone from the termination Gateway.
Even I can see in the CLI debugging the status is ringing
-my idea is to add music on hold stop when asterisk detect --
SIP/OPNS-096456c0 is ringing line
In whic
On Tue, Apr 14, 2009 at 9:52 AM, Florian Hackenberger
wrote:
> With asterisk 1.6, is it possible to use hylafax, or would asterisk
> terminate the fax calls itself?
With app_fax integrated into asterisk-1.6, you have an 'infinite'
modem pool that you control through the dial-plan. Using dialplan
On Tue, 14 Apr 2009, Arjan Kroon | Mobillion wrote:
> I record the message in ULAW
>
> After that I call sox with this command:
> /usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1
> $wav_fl
What are "-1" and "-2" for? Both sox 12.17.5 and 12.18.1 say they are
invalid.
Thanks in a
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 native T38 support), I *strongly* recommend
> going with software that has native T38 s
On Tue, Apr 14, 2009 at 4:15 PM, Jared Smith wrote:
> - "Scott Gifford" wrote:
>> The CDR information contains the entire
>> duration of the call as billable seconds, including time spent
>> waiting
>> in the queue. I would like the billable seconds to only include the
>> time spent actually
Michael wrote:
> On Tue, 14 Apr 2009 20:47:29 you wrote:
>
>> Hi michael,
>>
>> you should open both tcp,udp 5060,5061 too and as you mentioned between
>> 1-2.
>>
>
> AFAIK 5061 TCP is for TLS SIP which isn't used much yet?
>
> Is TCP the default for 5060, with UDP as fallback, or i
Turn off the 'qualify' setting on the peer(s) in sip.conf to stop it
sending OPTIONS pings.
As for it responding with a 200 OK to such, there is no way to turn
that off.
--
Sent from mobile device
On Apr 14, 2009, at 8:14 AM, "Khaled W. Chehab"
wrote:
> Dears
>
> How to disallow asterisk
- "Scott Gifford" wrote:
> The CDR information contains the entire
> duration of the call as billable seconds, including time spent
> waiting
> in the queue. I would like the billable seconds to only include the
> time spent actually talking to an agent.
You're absolutely right -- the CDR in
On Wed, 15 Apr 2009 00:43:45 you wrote:
> Now for the part I do know something about. Native asterisk fax
> support and native asterisk sip support improved in 1.6. With 1.6
> there is a built-in app_fax module which works quite well for sending
> fax over SIP with T.38. I found the configuration
On Wed, 15 Apr 2009 00:30:43 David Backeberg wrote:
> Once it's a wav you can mp3 it with lame or your preferred encoder,
> but encoding and playing mp3s takes more cpu than just playing it in
> gsm, or stopping after sox and playing as a wav.
>
> > Has anyone got any suggestions based on previous
On Tue, Apr 14, 2009 at 7:11 AM, Michael wrote:
> On Tue, 14 Apr 2009 23:00:02 Florian Hackenberger wrote:
>
>> Can somone spot the problem? Is someone using t38modem with asterisk
>> successfully?
> The best advice I can offer is to give up now and use Callweaver otherwise you
> can spend hours,
My suggestion is to use a tool made specifically for this - we happen to
sell one, but there are many options with different prices and licencing
model. Don't reinvent the wheel and concentrate on added value.
l.
2009/4/14 Scott Gifford
> Hello,
>
> I'm working on an Asterisk configuration for
Hey,
I record the message in ULAW
exten => s,1,Record(${A_record}:ulaw,0,60)
After that I call sox with this command:
/usr/bin/sox -c 1 -1 -t ul -r 8000 $in_fl -t wav -2 -r 8000 -c 1
$wav_fl
Regards,
Arjan Kroon
Mobillion BV
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