Re: [asterisk-users] broadcast
Hi List,
I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n,System(echo -e Channel: Channel:
local/23@contest-call\\nContext:
contest-call\\nExtension: 23\\nPriority: 1 /tmp/${UNIQUEID}.call)
exten = 1234,n,Konference(43689956,ADMRSTVL)
[contest-call]
exten = _X!,1,Answer()
exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten = _X!,n,Konference(43689956,ADMRSTV)
exten = _X!,n,Wait(10)
exten = _X!,n,Hangup()
in it I am dialing 1234 from softphone then join to conf in mute mode, after
it .call file start playback at it's own channels but I am not able to hear
anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:
Good to know,
I think it'll be a feedback score or a poll from members of the conference.
So if you use the R option and collect DTMF from members, and an AMI script
listening to that particular DTMF event collects all. This way your AMI
listener script should be able to tell you at the end of poll what user
inserted with DTMF.
So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script
should theoretically work for you.
On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.comwrote:
Hi Sam,
You are right. I am looking for the same
On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:
IMHO, I think Bhaati is trying to get feedback from
multiple conference users. See DTMF options in Konference module.
'R' : enable DTMF relay: DTMF tones generate a manager event
If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference
While some file is played and users press any DTMF collect the AMI events
from each user and use them as you require.
Ref:
http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:generalItemid=173
On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.comwrote:
Hi Ahmed,
Konference is also an conferencing application.
On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed gohar.ah...@vopium.comwrote:
Hhhmmm..I dunt have any experience with module Konference. Maybe anyone
else can help you on that.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
*Sent:* Monday, September 12, 2011 1:28 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] broadcast
** **
Hi Ahmed,
I did the same thing earlier to test the load of Digium card. But this
time I want to play file and want to get some DTMF from all the members of
conference.
So in this case I need more control into Konference module. But when I
use .call files then control will not go longer with all events.
Is there any alternate way to do it?
I appreciate your suggestion and will doing in parallel at higher
priority
On Mon, Sep 12, 2011 at 12:33 PM, Gohar Ahmed gohar.ah...@vopium.com
wrote:
Make a .call file..join one leg to local extension which plays the file
and the other leg to conference. The local extension will be like a
conference member.
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra bhati
*Sent:* Monday, September 12, 2011 11:44 AM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* [asterisk-users] broadcast
Hi List,
Is there any way by which I can broadcast any audio file to all members
into the conference ?
I don't want to play file individual channels.
--
-
Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
--
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asterisk-users mailing list
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Thanks and regards
Virendra Bhati
+91-9172341457
Software Engineer
** **
--
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New to Asterisk? Join us for a live
Re: [asterisk-users] broadcast
Virendra,
you need to change your logic just a bit. in call file a Channel is one
which needs to be dialled fires (See
linkhttp://www.voip-info.org/wiki/view/Asterisk+auto-dial+out).
this will be an extension where your Konference is Hosted for all the other
callers to join. i.e *Channel: local/s@Konference*
[Konference]
exten = s,1,ANSWER()
exten = s,n,if(conference is already started//do nothing else: trigger the
system command to make a call file...don't forget to move it to outgoing
directory)
exten = s,n,SET(some thing else you need to set for each incoming call i.e
save CallerID etc)
exten = s,n(message),Konference(43689956,ADMRSTV)
exten = s,n,Hangup()
Note that the call file should be triggered only for the first caller and
not every time a participant joins in. That'll case overlap message
broadcasts.
Next thing in call file is the destination which will be playing broadcast
message once Konference is called.
*Context:*broadcast-message
*Extension: *s
*Priority: *1
*
*
[broadcast-message]
exten = s,1,Answer()
exten = s,n,Set(p=/var/spool/asterisk/monitor/)
exten = s,n,playback(${p}/LQA/12/Biology/Que3)
exten = s,n,playback(${p}/LQA/12/Biology/Que4)
exten = s,n,playback(${p}/LQA/12/Biology/Que5)
exten = s,n,playback(${p}/LQA/12/Biology/Que6)
exten = s,n,playback(${p}/LQA/12/Biology/Que7)
exten = s,n,Wait(10)
exten = s,n,Hangup()
This should work and konference should listen to the playbacks.
Regards,
Sammy.
On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n,System(echo -e Channel: Channel:
local/23@contest-call\\nContext:
contest-call\\nExtension: 23\\nPriority: 1 /tmp/${UNIQUEID}.call)
exten = 1234,n,Konference(43689956,ADMRSTVL)
[contest-call]
exten = _X!,1,Answer()
exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten = _X!,n,Konference(43689956,ADMRSTV)
exten = _X!,n,Wait(10)
exten = _X!,n,Hangup()
in it I am dialing 1234 from softphone then join to conf in mute mode,
after it .call file start playback at it's own channels but I am not able to
hear anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:
Good to know,
I think it'll be a feedback score or a poll from members of the
conference. So if you use the R option and collect DTMF from members, and an
AMI script listening to that particular DTMF event collects all. This way
your AMI listener script should be able to tell you at the end of poll what
user inserted with DTMF.
So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script
should theoretically work for you.
On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.comwrote:
Hi Sam,
You are right. I am looking for the same
On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:
IMHO, I think Bhaati is trying to get feedback from
multiple conference users. See DTMF options in Konference module.
'R' : enable DTMF relay: DTMF tones generate a manager event
If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference
While some file is played and users press any DTMF collect the AMI
events from each user and use them as you require.
Ref:
http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:generalItemid=173
On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.comwrote:
Hi Ahmed,
Konference is also an conferencing application.
On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed
gohar.ah...@vopium.comwrote:
Hhhmmm..I dunt have any experience with module Konference. Maybe
anyone else can help you on that.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra
bhati
*Sent:* Monday, September 12, 2011 1:28 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] broadcast
** **
Hi Ahmed,
I did the same thing earlier to test the load of Digium card. But this
time I want to play file and want to get some DTMF from all the members
of
conference.
So in this case I need more control into Konference module. But when I
use .call files then control will not go longer with all events.
Is there any alternate way to do it?
I appreciate your
Re: [asterisk-users] broadcast
Hey there
You are not moving the call file to spool/outgoing directory. Maybe that's
why you aren't getting anything. I don't feel good about the call file also.
Its not doing what you want it to do.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Tuesday, September 13, 2011 11:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] broadcast
Hi List,
I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n,System(echo -e Channel: Channel:
local/23@contest-call\\nContext: contest-call\\nExtension: 23\\nPriority: 1
/tmp/${UNIQUEID}.call)
exten = 1234,n,Konference(43689956,ADMRSTVL)
[contest-call]
exten = _X!,1,Answer()
exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten = _X!,n,Konference(43689956,ADMRSTV)
exten = _X!,n,Wait(10)
exten = _X!,n,Hangup()
in it I am dialing 1234 from softphone then join to conf in mute mode, after
it .call file start playback at it's own channels but I am not able to hear
anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:
Good to know,
I think it'll be a feedback score or a poll from members of the conference.
So if you use the R option and collect DTMF from members, and an AMI script
listening to that particular DTMF event collects all. This way your AMI
listener script should be able to tell you at the end of poll what user
inserted with DTMF.
So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script should
theoretically work for you.
On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.com wrote:
Hi Sam,
You are right. I am looking for the same
On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:
IMHO, I think Bhaati is trying to get feedback from multiple conference
users. See DTMF options in Konference module.
'R' : enable DTMF relay: DTMF tones generate a manager event
If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference
While some file is played and users press any DTMF collect the AMI events
from each user and use them as you require.
Ref: http://main.voiptoday.org/index.php?option=com_content
http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:
asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:gene
ralItemid=173
view=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-av
ailablecatid=35:generalItemid=173
On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.com wrote:
Hi Ahmed,
Konference is also an conferencing application.
On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed gohar.ah...@vopium.com wrote:
Hhhmmm..I dunt have any experience with module Konference. Maybe anyone else
can help you on that.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 1:28 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] broadcast
Hi Ahmed,
I did the same thing earlier to test the load of Digium card. But this time
I want to play file and want to get some DTMF from all the members of
conference.
So in this case I need more control into Konference module. But when I use
.call files then control will not go longer with all events.
Is there any alternate way to do it?
I appreciate your suggestion and will doing in parallel at higher priority
On Mon, Sep 12, 2011 at 12:33 PM, Gohar Ahmed gohar.ah...@vopium.com
wrote:
Make a .call file..join one leg to local extension which plays the file and
the other leg to conference. The local extension will be like a conference
member.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of virendra bhati
Sent: Monday, September 12, 2011 11:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] broadcast
Hi List,
Is there any way by which I can broadcast any audio file to all members into
the conference ?
I don't want to play file individual channels.
--
-
Thanks and regards
Virendra Bhati
+91-9172341457 tel:%2B91-9172341457
Software Engineer
--
_
-- Bandwidth and Colocation Provided by
Re: [asterisk-users] broadcast
Hi Sam,
I am doing the same things.
into your suggested script you join into context Konference and then .call
file start IVRs .
the same logic I have pasted in which I make .call file and then join into
the Konference and then .call file start it's work.
But As i know they are on different -2 channels and not joined into same
conference. That's why no audio is able to broadcast into conference.
[broadcast-message]
exten = s,1,Answer()
exten = s,n,Set(p=/var/spool/
asterisk/monitor/)
exten = s,n,playback(${p}/LQA/12/Biology/Que3)
exten = s,n,playback(${p}/LQA/12/Biology/Que4)
exten = s,n,playback(${p}/LQA/12/Biology/Que5)
exten = s,n,playback(${p}/LQA/12/Biology/Que6)
exten = s,n,playback(${p}/LQA/12/Biology/Que7)
exten = s,n,Wait(10)
exten = s,n,Hangup()
Where you have mention in which conf. it will be start ?
miss comunication in between .call and rest users.
On Tue, Sep 13, 2011 at 12:34 PM, Sam Govind govoi...@gmail.com wrote:
Virendra,
you need to change your logic just a bit. in call file a Channel is one
which needs to be dialled fires (See
linkhttp://www.voip-info.org/wiki/view/Asterisk+auto-dial+out).
this will be an extension where your Konference is Hosted for all the other
callers to join. i.e *Channel: local/s@Konference*
[Konference]
exten = s,1,ANSWER()
exten = s,n,if(conference is already started//do nothing else: trigger the
system command to make a call file...don't forget to move it to outgoing
directory)
exten = s,n,SET(some thing else you need to set for each incoming call i.e
save CallerID etc)
exten = s,n(message),Konference(43689956,ADMRSTV)
exten = s,n,Hangup()
Note that the call file should be triggered only for the first caller and
not every time a participant joins in. That'll case overlap message
broadcasts.
Next thing in call file is the destination which will be playing broadcast
message once Konference is called.
*Context:*broadcast-message
*Extension: *s
*Priority: *1
*
*
[broadcast-message]
exten = s,1,Answer()
exten = s,n,Set(p=/var/spool/asterisk/monitor/)
exten = s,n,playback(${p}/LQA/12/Biology/Que3)
exten = s,n,playback(${p}/LQA/12/Biology/Que4)
exten = s,n,playback(${p}/LQA/12/Biology/Que5)
exten = s,n,playback(${p}/LQA/12/Biology/Que6)
exten = s,n,playback(${p}/LQA/12/Biology/Que7)
exten = s,n,Wait(10)
exten = s,n,Hangup()
This should work and konference should listen to the playbacks.
Regards,
Sammy.
On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati virbh...@gmail.comwrote:
Hi List,
I make a script for .call file and then I started playback on local
channel but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n,System(echo -e Channel: Channel:
local/23@contest-call\\nContext:
contest-call\\nExtension: 23\\nPriority: 1 /tmp/${UNIQUEID}.call)
exten = 1234,n,Konference(43689956,ADMRSTVL)
[contest-call]
exten = _X!,1,Answer()
exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten = _X!,n,Konference(43689956,ADMRSTV)
exten = _X!,n,Wait(10)
exten = _X!,n,Hangup()
in it I am dialing 1234 from softphone then join to conf in mute mode,
after it .call file start playback at it's own channels but I am not able to
hear anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:
Good to know,
I think it'll be a feedback score or a poll from members of the
conference. So if you use the R option and collect DTMF from members, and an
AMI script listening to that particular DTMF event collects all. This way
your AMI listener script should be able to tell you at the end of poll what
user inserted with DTMF.
So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script
should theoretically work for you.
On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.comwrote:
Hi Sam,
You are right. I am looking for the same
On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:
IMHO, I think Bhaati is trying to get feedback from
multiple conference users. See DTMF options in Konference module.
'R' : enable DTMF relay: DTMF tones generate a manager event
If neither 'X' nor 'R' are present, DTMF tones will be forwarded to
all members in the conference
While some file is played and users press any DTMF collect the AMI
events from each user and use them as you require.
Ref:
http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:generalItemid=173
On Mon,
Re: [asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously
Hi 1st check that how many manager is connected into the server. 1 or more then you can say that 2 DTMF is capture by asterisk for same events. manager show connected Username IP Address root 127.0.0.1 it should be one only. I face the same case then I found that more then 1 manager was working into the server. On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban vrban.l...@googlemail.comwrote: Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO simultaneously. That has the effect, that asterisk read every dtmf twice. and yes, it's mainly the carriers mistake. but is there a configure option, that asterisk accept only one DMTF method for inbound dtmf? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting for Asterisk Call Center
On Friday 09 September 2011, bilal ghayyad wrote:
Hi All;
Anyone advise for a free (open source) reporting to be used for asterisk
call center?
Regards
Bilal
Problem is, reporting is such a nebulous thing, about the only thing that will
give you the required level of generality of purpose is a programming
language. So you might find it simplest just to roll your own.
Assuming you have set up CDR using some kind of database backend, then all you
really need to do is devise a set of queries which will provide the
information you want to present in your reports; then write a script in your
favourite language to make a CSV file (which will load into any modern
spreadsheet program) and e-mail it to whoever needs it. Lastly, put an entry
in your crontab to run the report whenever required.
Note that OpenOffice.org calc will treat an entry in a CSV starting with an =
sign as a formula, so you can insert things such as
${name},${ext},${answd},${busy},${unobtain},=C${row}+D${row}+E${row}
and then column F in each row will contain the total of columns C, D and E in
that row (assuming you are properly updating $row as you go along .) My
own preference is to use the spreadsheet program to evaluate formulae rather
than hard-code the answer into the spreadsheet. That way, if you edit a cell,
you will not throw everything else out.
I know from past experience that Microsoft Excel will happily accept CSV files
with an XLS extension (N.B. it might need \r\n for line endings); however, I
am unable to test whether or not it will accept formulae as above, as I do not
have access to a machine with Windows and Excel.
--
AJS
--
AJS
Answers come *after* questions.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
sox -h will list the formats supported by your install of sox. If mp3 is not listed, then your sox does not support mp3. This is not uncommon. Many Linux distros do not ship support for patent encumbered formats. Either stop using mp3 (this is what I suggest) or compile and install sox with mp3 support. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kaushal Shriyan Sent: Monday, September 12, 2011 8:12 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 Hi, Can someone please comment about the below issue [root@host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root@host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root@host0040 kaushal]# sox -v 0.125 -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 obd-demo.ulaw resample -ql -bash: obd-demo.ulaw: No such file or directory [root@host0040 kaushal]# sox -V obd-demo.mp3 -t au -r 8000 -U -b -c 1 obd-demo.ulaw resample -ql sox: Failed reading obd-demo.mp3: Do not understand foReply rmat type: mp3 [root@host0040 kaushal]# When i invoke the same obd-demo.mp3 it works perfectly fine host0040*CLI channel originate DAHDI/g0/xx Application MP3Player /home/kaushal/obd-demo.mp3 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:985 sig_pri_request: sig_pri_request 1 [Sep 9 16:44:56] DEBUG[12691]: sig_pri.c:6427 sig_pri_call: CALLER NAME: NUM: -- Requested transfer capability: 0x00 - SPEECH -- Launching MP3Player(/home/kaushal/obd-demo.mp3) on DAHDI/i1/9833754756-1 [root@host0040 ~]# rpm -qa | grep sox sox-12.18.1-1.el5_5.1 [root@host0040 ~]# rpm -qa | grep lame lame-3.98.4-1.el5.rf lame-devel-3.98.4-1.el5.rf [root@host0040 ~]# MP3 support in SoX is optional and requires access to either or both the external libmad and libmp3lame libraries. To see if there is support for Mp3 run sox -h and look for it under the list of supported file formats as mp3. [root@host0040 ~]# sox -h sox: Version 12.18.1 Usage: [ gopts ] [ fopts ] ifile [ fopts ] ofile [ effect [ effopts ] ] gopts: -e -h -p -q -S -V fopts: -r rate -c channels -s/-u/-U/-A/-a/-i/-g/-f -b/-w/-l/-d -v volume -x effect: avg band bandpass bandreject chorus compand copy dcshift deemph earwax echo echos fade filter flanger highp highpass lowp lowpass mask mcompand noiseprof noisered pan phaser pick pitch polyphase rate repeat resample reverb reverse silence speed stat stretch swap synth trim vibro vol effopts: depends on effect Supported file formats: aiff al alsa au auto avr cdr cvs dat vms gsm hcom la lu maud nul ossdsp prc raw sb sf sl smp sndt sph 8svx sw txw ub ul uw voc vorbis vox wav wve Which package contains libmad and libmp3lame libraries available on CentOS 5.6 Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] broadcast
I don't know why you are running into problems.
Once a call file is executed it creates two legs (according to call file
structure) A leg is Channel: Local/1234@conference and once it Answers the
call file the second leg is bridged which should be
Context-Extension-priority. So what I'm asking is make your conference
A-leg and your Playback/messages dial plan B-leg.
take a look at the changes I made to your dial-plan
[conference]
exten = 1234,1,Answer()
exten = 1234,n,Gotoif($[${FIRST-CALLER} 1]?startmsg:pass)
exten = 1234,n(startmsg),System(echo -e
Channel:local/1234@conference\\nContext: contest-call\\nExtension:
23\\nPriority: 1 /tmp/${UNIQUEID}.call)
exten = 1234,n,system(mv /tmp/${UNIQUEID}.call
/var/spool/asterisk/outgoing)
exten = 1234,n(pass),Konference(43689956,ADMRSTVL)
exten = 1234,n,Hangup()
[contest-call]
exten = 23,1,Answer()
exten = 23,n,Set(p=/var/spool/asterisk/monitor/)
exten = 23,n,playback(${p}/LQA/12/Biology/Que3)
exten = 23,n,playback(${p}/LQA/12/Biology/Que4)
exten = 23,n,playback(${p}/LQA/12/Biology/Que5)
exten = 23,n,playback(${p}/LQA/12/Biology/Que6)
exten = 23,n,playback(${p}/LQA/12/Biology/Que7)
exten = 23,n,Wait(10)
exten = 23,n,Hangup()
Here, changed your script to what I'm thinking. use the above tweak
accordingly. make sure to find out FIRST-CALLER so your tapes start playing
into conference just for once.
-Sammy
On Tue, Sep 13, 2011 at 11:25 AM, virendra bhati virbh...@gmail.com wrote:
Hi List,
I make a script for .call file and then I started playback on local channel
but nothing was hearing at another channles.
exten = 1234,1,Answer()
exten = 1234,n,System(echo -e Channel: Channel:
local/23@contest-call\\nContext:
contest-call\\nExtension: 23\\nPriority: 1 /tmp/${UNIQUEID}.call)
exten = 1234,n,Konference(43689956,ADMRSTVL)
[contest-call]
exten = _X!,1,Answer()
exten = _X!,n,Set(p=/var/spool/asterisk/monitor/)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que3)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que4)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que5)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que6)
exten = _X!,n,playback(${p}/LQA/12/Biology/Que7)
exten = _X!,n,Konference(43689956,ADMRSTV)
exten = _X!,n,Wait(10)
exten = _X!,n,Hangup()
in it I am dialing 1234 from softphone then join to conf in mute mode,
after it .call file start playback at it's own channels but I am not able to
hear anything into conf.
As i know localdial is not joining into the conf. but how I will do it so
that I will be able to hear any played file into conference ?
On Mon, Sep 12, 2011 at 3:36 PM, Sam Govind govoi...@gmail.com wrote:
Good to know,
I think it'll be a feedback score or a poll from members of the
conference. So if you use the R option and collect DTMF from members, and an
AMI script listening to that particular DTMF event collects all. This way
your AMI listener script should be able to tell you at the end of poll what
user inserted with DTMF.
So overall insertion of a broadcast message using Ahmed's method of .call
file and later on collecting DTMF events from AMI script
should theoretically work for you.
On Mon, Sep 12, 2011 at 2:37 PM, virendra bhati virbh...@gmail.comwrote:
Hi Sam,
You are right. I am looking for the same
On Mon, Sep 12, 2011 at 3:01 PM, Sam Govind govoi...@gmail.com wrote:
IMHO, I think Bhaati is trying to get feedback from
multiple conference users. See DTMF options in Konference module.
'R' : enable DTMF relay: DTMF tones generate a manager event
If neither 'X' nor 'R' are present, DTMF tones will be forwarded to all
members in the conference
While some file is played and users press any DTMF collect the AMI
events from each user and use them as you require.
Ref:
http://main.voiptoday.org/index.php?option=com_contentview=articleid=566:asterisk-conferencing-module-appkonference-16-is-now-availablecatid=35:generalItemid=173
On Mon, Sep 12, 2011 at 2:20 PM, virendra bhati virbh...@gmail.comwrote:
Hi Ahmed,
Konference is also an conferencing application.
On Mon, Sep 12, 2011 at 2:12 PM, Gohar Ahmed
gohar.ah...@vopium.comwrote:
Hhhmmm..I dunt have any experience with module Konference. Maybe
anyone else can help you on that.
** **
*From:* asterisk-users-boun...@lists.digium.com [mailto:
asterisk-users-boun...@lists.digium.com] *On Behalf Of *virendra
bhati
*Sent:* Monday, September 12, 2011 1:28 PM
*To:* Asterisk Users Mailing List - Non-Commercial Discussion
*Subject:* Re: [asterisk-users] broadcast
** **
Hi Ahmed,
I did the same thing earlier to test the load of Digium card. But this
time I want to play file and want to get some DTMF from all the members
of
conference.
So in this case I need more control into Konference module. But when I
use .call files then control will not go longer with all events.
Is there any alternate way to do it?
I appreciate your suggestion and will doing in parallel at higher
priority
On Mon, Sep 12,
Re: [asterisk-users] Reporting for Asterisk Call Center
Dear Tareq; I am not using mysql, the configuration on the text configuratoin files and the logs are existed under the directory (/var/log/asterisk). Well, to use mysql: then it means the configuration will be also in the database or I can use mysql only for reporting? What is the Flash Operator? By the way, I have another question if you can help me if you used the database with sql, actually I was facing one time a case and maybe the Database usage will help me if you can advise me: If I have multiple Asterisk servers are running, and I need them to work centralized (I mean from one configuration) so to work as one system, then if I have database for configuration, I can acheive this by making all the servers read and write the configuration from the database server? Thanks for your help Tareq. Regards Bilal --- those reports can be easiely extracted from the MYSQL database my friend.. and you can add the Flash Operator Panel if you want to monitor live activities like how many in queue and how many ON CALL .. etc anyway the Elastix is a stand alone distribution you can find more info and downloads at : http://elastix.org/ regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] SIP Realtime Templates (!)
Hello, Is it possible to assign templates defined in sip.conf to sip realtime peers? There was another mail in 2008 which asked the same question but never received a response. Thanks, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] SIP Realtime Templates (!)
Hi To the best of my knowledge there isn't. But, if you're using realtime you can create a program to add your extensions to the database and you can create the concept of templates within that. Regards Ish On Tue, 2011-09-13 at 12:27 +0200, Alexandru Oniciuc wrote: Hello, Is it possible to assign templates defined in sip.conf to sip realtime peers? There was another mail in 2008 which asked the same question but never received a response. Thanks, Alex -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Ishfaq Malik Software Developer PackNet Ltd Office: 0161 660 3062 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468)
Hi All; Asterisk version is: 1.8.5.0 But I see at the consol the following warning and really I did google but did not understand if it is bug or related to settings: [Sep 13 15:04:56] WARNING[2209]: chan_sip.c:19667 handle_response_invite: just did sched_add waitid(3429468) for sip_reinvite_retry for dialog 3c581fa96f2b-53yysntgjmwb in handle_response_invite But actually, we see some SNOM IP Phones has NR (Not Register) at the LCD, and it is able to receive and originate calls !! I was think if this is bug or if it is related to session expire .. but I am not able to determine until now. Any help? By the way: which paramter in the sip.conf can be used to determine the timeout of the sip registration (so the IP Phone should send the registartion packet to keepalive within this timeout, otherwise it will be considered not register)? Is it the defaultexpiry or something else? Also, the above warning, to what it could be related? Is it a bug? Regards Bilal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
Kaushal, Your version of SoX does not have MP3 support. Since you have LAME installed, use it as a first step to produce an intermediate file that SoX supports. Then use SoX to convert the intermediate file to the desired format. Step 1 -- # lame --decode obd-demo.mp3 obd-demo.wav input: obd-demo.mp3 (8 kHz, 1 channel, MPEG-2.5 Layer III) output: obd-demo.wav (16 bit, Microsoft WAVE) skipping initial 1105 samples (encoder+decoder delay) Frame# 16818/16818 16 kbps # file obd-demo.wav obd-demo.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Step 2 -- # sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.ulaw sox: Detected file format type: wav sox: WAV Chunk fmt sox: WAV Chunk data sox: Reading Wave file: Microsoft PCM format, 1 channel, 8000 samp/sec sox: 16000 byte/sec, 2 block align, 16 bits/samp, 19372126 data bytes sox: 9686063 Samps/chans sox: Input file obd-demo.wav: using sample rate 8000 size shorts, encoding signed (2's complement), 1 channel sox: Output file obd-demo.ulaw: using sample rate 8000 size bytes, encoding u-law, 1 channel sox: Output file: comment Processed by SoX Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously
hello Virendra, thx for your response. but after i made clear to the carrier that i want the dmtf only via rfc2833 and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed. Kristijan 2011/9/13 virendra bhati virbh...@gmail.com: Hi 1st check that how many manager is connected into the server. 1 or more then you can say that 2 DTMF is capture by asterisk for same events. manager show connected Username IP Address root 127.0.0.1 it should be one only. I face the same case then I found that more then 1 manager was working into the server. On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban vrban.l...@googlemail.com wrote: Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO simultaneously. That has the effect, that asterisk read every dtmf twice. and yes, it's mainly the carriers mistake. but is there a configure option, that asterisk accept only one DMTF method for inbound dtmf? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] High delay from Asterisk as PSTN simulator
I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP - Asterisk 1 Asterisk 2 Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2. I've read the source code of chan_dahdi, and I think the channel has a 20 ms buffer (160 samples). Algorithms like mg2 and kb1 are configured to accept 128 taps (16 ms), so 20 ms is too high. Someone knows how I can reduce the delay to at least 10 ms? Should I change something in the source code? Thanks in advance, Gustavo Santos. -- Atenciosamente, Gustavo Santos. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Issues After Upgrade
I disabled the echo cancelled on the PRI and the same issues are still popping up: PRI Span: 1 !! Unknown IE 128 (cs0) -- Span 1: Channel 0/22 got hangup, cause 16 Anything else I can try? Stephen H. Gerstacker Sr. Database Developer Electronic Data Payment Systems Phone: 866.578.9740 ext. 114 Fax: 866.528.3854 www.edpaymentsystems.comhttp://www.edpaymentsystems.com On Sep 11, 2011, at 9:52 AM, Stephen H. Gerstacker wrote: When we moved buildings, the PRI provider specifically asked to switch to dms100. That's how the old server was as well. I'll try the echo canceller first. - Stephen H. Gerstacker On Sep 11, 2011, at 9:07, Doug Lytle supp...@drdos.infomailto:supp...@drdos.info wrote: Stephen H. Gerstacker wrote: switchtype=dms100 Are you sure that your switchtype is correct and your provider has a dms100? If not, change this to national. If so, the two things I'd try to narrow it down are: 1.) Temporarily remove the AEX410 card and test again 2.) Temporarily disable OSLEC for your echo canceller and test again. 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 Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Issues After Upgrade
Stephen H. Gerstacker wrote: Anything else I can try? Try switchtype=national just for testing. 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 Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] DMTF via rfc2833 and SIP-INFO simultaneously
Hi , What was the solution of that problem ? Did provider change the setting at there end or else ? On Tue, Sep 13, 2011 at 7:37 PM, Kristijan Vrban vrban.l...@googlemail.comwrote: hello Virendra, thx for your response. but after i made clear to the carrier that i want the dmtf only via rfc2833 and not via rfc2833 and SIP-INFO simultaneously, the problem is fixed. Kristijan 2011/9/13 virendra bhati virbh...@gmail.com: Hi 1st check that how many manager is connected into the server. 1 or more then you can say that 2 DTMF is capture by asterisk for same events. manager show connected Username IP Address root 127.0.0.1 it should be one only. I face the same case then I found that more then 1 manager was working into the server. On Fri, Aug 26, 2011 at 3:11 PM, Kristijan Vrban vrban.l...@googlemail.com wrote: Hello, i have a carrier that send DMTF via rfc2833 and SIP-INFO simultaneously. That has the effect, that asterisk read every dtmf twice. and yes, it's mainly the carriers mistake. but is there a configure option, that asterisk accept only one DMTF method for inbound dtmf? Kristijan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- - Thanks and regards Virendra Bhati +91-9172341457 Software Engineer -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Sip profiles per customer, behind a SIP proxy. How?
On 09/10/2011 09:16 PM, Robert Thomas wrote: Hello List, I have been trying to configure a sip profile ( peer / friend ) for each of my customers behind a sip proxy for some time, but I have had no success, so I would appreciate your help. Customer - OpenSIPS - Asterisk - PSTN The opensips is working as a sip proxy with record route, for billing, load balancing and authentication purposes. I would like to be able to define a particular context, or settings per customer. Since the customer is behind the SIP proxy, all I see if the packet comming from the SIP proxy. So I have created a peer profile with the IP Address of my proxy. Problem been any setting I enable affect all traffic coming throught the SIP proxy. I was reading that Asterisk checks the SIP From: address username and matches against names of devices with type=user- However I have some problems with Asterisk 1.6.2, taking the caller id either from the RPID so I manually parse the PAI header. I was thinking about replacing the From with a customer ID, and for those customers that use the FROM to signal caller id, to copy it over the PAI header at the SIP proxy. I don't know if overwriting the FROM would cause any problem with the SIP clients behind the proxy. Is there any better or different way to acomplish this? I would say we should have some flexibility although we have a SIP proxy as the source IP for all of our traffic. Olle Johansson has a developer branch that includes a method to do exactly what you are looking for; I suggest you look him up and find out what state it is in, and see whether you can help test it. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] High delay from Asterisk as PSTN simulator
On 09/13/2011 08:56 AM, Gustavo Santos wrote: I'm trying to use Asterisk as a PSTN simulator to run performance tests for echo cancellation algorithms. I'm using the following configuration: SIP - Asterisk 1 Asterisk 2 Echo() Asterisk 1 and Asterisk 2 are connected using E1. Echo() is the dialplan application. The problem is the high delay using this configuration: 20 ms only in Asterisk 2. I've read the source code of chan_dahdi, and I think the channel has a 20 ms buffer (160 samples). Algorithms like mg2 and kb1 are configured to accept 128 taps (16 ms), so 20 ms is too high. Someone knows how I can reduce the delay to at least 10 ms? Should I change something in the source code? 20 milliseconds is far from a 'high' (long) delay. Asterisk handles audio in packets, it does not directly switch TDM streams. As a result, there is always going to be (at least) the delay of one packet time for audio passing into Asterisk and back out via the Echo() application. This is unavoidable. An alternative solution would be to send a call into Asterisk2 and have it dial back to Asterisk1 (and then back to the originating endpoint) and bridge those two calls in Asterisk2; if both calls are on the same E1, then Asterisk will let the DAHDI hardware directly connect the two channels, resulting in a 1 or 2 millisecond delay. But realistically... configuring an echo canceller with only a 16ms window of operation is not very practical. Sending a call through *any* network element that packetizes the audio will result in a delay longer than 16ms. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 tel:%2B1-760-468-3867 PST Newline Fax: +1-760-731-3000 tel:%2B1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk 1.4 Vs 1.6 Vs 1.8
On 12/09/11 09:48 PM, Joseph wrote: Was NAT problem fixed in 1.8.7 ? I'm using 1.4 but I've tried 1.6 and 1.8.5 and in both cases connection is not working with my provider with SIP + NAT. The connection is showing up as registered but the call is not coming IN (congestion). Can you define NAT problem? I'm unaware of any issues with Asterisk (or end points) behind NAT. It is mostly likely a configuration issue rather than a bug. -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk server: Console or GUI OS ? Init level 3 or init level 5 ?
On 12/09/11 02:21 PM, linux guy wrote: I'm about to start building my asterisk server and I can't seem to find anything that discusses the pros and cons of installing the OS (Fedora 15) as console only or GUI, ie install KDE as well. http://ofps.oreilly.com/titles/9780596517342/asterisk-Install.html#Installing_id291070 -- Leif Madsen http://www.oreilly.com/catalog/asterisk -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? ** ** *From:* asterisk-users-boun...@lists.digium.com [mailto: asterisk-users-boun...@lists.digium.com] *On Behalf Of *naren *Sent:* Tuesday, September 13, 2011 2:22 PM *To:* John Novack *Cc:* Asterisk Users Mailing List - Non-Commercial Discussion *Subject:* Re: [asterisk-users] Question about voip.ms service. ** ** Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? ** ** Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. ** ** The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. ** ** But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) ** ** Thanks ** ** On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: ** ** I also found this... seems like voip.ms outbound is broken for now! ** ** http://pbxinaflash.com/forum/showthread.php?t=10735 ** ** ** ** On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, ** ** I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? ** ** I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. ** ** I would really appreciate it if you could post the relevant section of your sip.conf for me. ** ** Thanks! Naren ** ** On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx ** ** 'slam-dunk.' ** ** Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! ** ** is ** ** Suggest you check your firewall and your configs, and above all post some more information ** ** IAX ** ** If you really want to upset some, top post as I have just done! ** ** Agreed. ** ** The real issue is communication, top bottom or in the middle ** ** Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ** ** -- _ -- Bandwidth and Colocation Provided by
Re: [asterisk-users] Question about voip.ms service.
I see what you mean. Maybe if you call their support they can tell you what you need to know. If not, voicepulse is a pretty good provider. From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:59 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 tel:%2B1-760-468-3867 PST Newline Fax: +1-760-731-3000 tel:%2B1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
I was lurking in this conversation and I went to look more carefully at the voip.ms site. I found sample files at http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 Hope that helps. On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote: I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.' Though they suggest SIP, I chose IAX and have 4569 UDP open in my firewall a Their on line config samples just work! is Suggest you check your firewall and your configs, and above all post some more information IAX If you really want to upset some, top post as I have just done! Agreed. The real issue is communication, top bottom or in the middle Sometimes, it's just about being considerate to 'the next guy.' -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every
[asterisk-users] Send DTMF
¿How can i could to Send DTMF digits on a current call by scripting or similar methods? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Send DTMF
http://www.voip-info.org/wiki/view/Asterisk+cmd+SendDTMF From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ezequiel Lovelle Sent: Tuesday, September 13, 2011 3:37 PM To: Asterisk Users Subject: [asterisk-users] Send DTMF ¿How can i could to Send DTMF digits on a current call by scripting or similar methods? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
Yup, that part I got. What I am not clear about is how to set up the DID to go to my URI. When I select manage DIDs and click on the one I want to change, I see the following options for routing the DID x SIP/IAX - [main account] IAX2/10 - with my account number x SIP URI - SIP:mysi...@myuri.com:5060 x System - Hangup There are several other options but they are not selectable for me because I have not set up to use them. I used to have the routing set to SIP URI where I was able to specify my URI where the call was routed to. But with the SIP/IAX option I do not have that ability. I am missing something fundamental here. My asterisk has the iax.conf and extensions.conf entries ready to receive calls from voip.ms, but I don't know how to tel voip.ms to send the calls to my asterisk with the IAX protocol. I understand this is probably a question for the voip.ms folks, but since a couple of people mentioned earlier that they were rocking with IAX, I thought it would be an easy question for them to point me in the right direction. Thanks. On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel dai...@pervasivetelecom.comwrote: I was lurking in this conversation and I went to look more carefully at the voip.ms site. I found sample files at http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 Hope that helps. On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote: I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are correct - but I am no expert. I would really appreciate it if you could post the relevant section of your sip.conf for me. Thanks! Naren On Thu, Jun 9, 2011 at 3:22 PM, Steve Edwards asterisk@sedwards.com wrote: On Thu, 9 Jun 2011, John Novack wrote: I use voip.ms and have no issues using IAX and Asterisk 1.4.xx 'slam-dunk.'
Re: [asterisk-users] Question about voip.ms service.
That’s what this part of extensions.conf should do: ; inbound context example for your DID numbers, do not add the number 1 in front [voipms-inbound] exten = 7863643011,1,Answer() ;your DID From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Yup, that part I got. What I am not clear about is how to set up the DID to go to my URI. When I select manage DIDs and click on the one I want to change, I see the following options for routing the DID x SIP/IAX - [main account] IAX2/10 - with my account number x SIP URI - SIP:mysi...@myuri.com:5060 x System - Hangup There are several other options but they are not selectable for me because I have not set up to use them. I used to have the routing set to SIP URI where I was able to specify my URI where the call was routed to. But with the SIP/IAX option I do not have that ability. I am missing something fundamental here. My asterisk has the iax.conf and extensions.conf entries ready to receive calls from voip.ms, but I don't know how to tel voip.ms to send the calls to my asterisk with the IAX protocol. I understand this is probably a question for the voip.ms folks, but since a couple of people mentioned earlier that they were rocking with IAX, I thought it would be an easy question for them to point me in the right direction. Thanks. On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel dai...@pervasivetelecom.com wrote: I was lurking in this conversation and I went to look more carefully at the voip.ms site. I found sample files at http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 Hope that helps. On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote: I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that they have on their website? I have pretty much the same settings. When I dial out, the destination rings, but I can't hear a ringback tone from on the source side ( I am using a PAP2T router with a phone). I have set up outgoing with actionvoip before and that is working fine, so I am thinking my router settings for my ports are
[asterisk-users] Determine negotiated codec in script
Sorry if this is an obvious question and perhaps my Google foo isn't right on this one: I have calls coming into an Asterisk server that may be using 2 different codecs. I am recording audio in both cases but the challenge is knowing which codec was negotiated at call setup. I need to pass the proper format to the record command as the codecs cannot be transcoded and are only supported for playback/record/passthru etc. Is there some global variable present that I can look at for codec identification? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone On Sep 13, 2011, at 5:14 PM, Danny Nicholas da...@debsinc.com wrote: That’s what this part of extensions.conf should do: ; inbound context example for your DID numbers, do not add the number 1 in front [voipms-inbound] exten = 7863643011,1,Answer() ;your DID From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 4:09 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Yup, that part I got. What I am not clear about is how to set up the DID to go to my URI. When I select manage DIDs and click on the one I want to change, I see the following options for routing the DID x SIP/IAX - [main account] IAX2/10 - with my account number x SIP URI - SIP:mysi...@myuri.com:5060 x System - Hangup There are several other options but they are not selectable for me because I have not set up to use them. I used to have the routing set to SIP URI where I was able to specify my URI where the call was routed to. But with the SIP/IAX option I do not have that ability. I am missing something fundamental here. My asterisk has the iax.conf and extensions.conf entries ready to receive calls from voip.ms, but I don't know how to tel voip.ms to send the calls to my asterisk with the IAX protocol. I understand this is probably a question for the voip.ms folks, but since a couple of people mentioned earlier that they were rocking with IAX, I thought it would be an easy question for them to point me in the right direction. Thanks. On Tue, Sep 13, 2011 at 3:32 PM, Dave Aibel dai...@pervasivetelecom.com wrote: I was lurking in this conversation and I went to look more carefully at the voip.ms site. I found sample files at http://wiki.voip.ms/article/PBXs#Asterisk_.28SIP.29 Hope that helps. On Tue, Sep 13, 2011 at 3:59 PM, naren naren.sa...@gmail.com wrote: I see the section you are talking about. It is on the home page if I am not logged in. I see the Authentication section and the text IAX/SIP registration, but it doesn't seem to be a link. I am not sure how I can find the page that has the details about the IAX/SIP registration. I see in the wiki there is a page that has the configuration info for iax.conf and extensions.conf. Thanks for your help. naren On Tue, Sep 13, 2011 at 2:25 PM, Danny Nicholas da...@debsinc.com wrote: Did you read the “IAX/SIP registration” section (under Authentication) on voip.ms? From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of naren Sent: Tuesday, September 13, 2011 2:22 PM To: John Novack Cc: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Question about voip.ms service. Ok... this is probably a dumb question but I can't figure out how to set voip.ms to use IAX for my DID... with SIP I was able to specify the URI so I pointed it to my asterisk installation, but with IAX I don't have that option. Is that supposed to work some other way? Thanks a bunch! On Mon, Sep 12, 2011 at 11:18 PM, naren naren.sa...@gmail.com wrote: I am novice with Asterisk, I had to piece together a lot of bits of info from lots of internet searches to get my very basic setup working. I probably shouldn't say that because it seems like Nat is not a very basic setup with Asterisk. The reason for wanting to stay with SIP is because I have my setup working with that protocol with an incoming and an outgoing line. I just wanted to add a second outgoing with voip.ms. But, I have come so far, so well why not... I will give IAX a shot, and see what traps I need to wade through :) Thanks On Mon, Sep 12, 2011 at 11:09 AM, John Novack jnov...@stromberg-carlson.org wrote: Never have had a problem with their IAX service. And ( for now ) a little hedge against the hackers. Since Asterisk is involved, why not use IAX anyway? John Novack naren wrote: I also found this... seems like voip.ms outbound is broken for now! http://pbxinaflash.com/forum/showthread.php?t=10735 On Sun, Sep 11, 2011 at 10:34 PM, naren naren.sa...@gmail.com wrote: Hi, I am trying to set up my asterisk 1.8.5 with voip.ms. I had no problem with the incoming, but my outgoing is not working. If at all possible, I would like to stick with SIP. Since the original poster (Glen) had mentioned that he had gotten outgoing working, I was wondering if you would be kind enough to post some thoughts on that. Were you able to get it working with just the default example sip.conf / extensions.conf settings that
Re: [asterisk-users] Determine negotiated codec in script
Sip show channels will give you the active codec. You can get the information using an AGI or a system command. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tom Browning Sent: Tuesday, September 13, 2011 4:19 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: [asterisk-users] Determine negotiated codec in script Sorry if this is an obvious question and perhaps my Google foo isn't right on this one: I have calls coming into an Asterisk server that may be using 2 different codecs. I am recording audio in both cases but the challenge is knowing which codec was negotiated at call setup. I need to pass the proper format to the record command as the codecs cannot be transcoded and are only supported for playback/record/passthru etc. Is there some global variable present that I can look at for codec identification? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
That's what I am hoping to do as well. Could you share some insight on how you set up the DID on the voip.ms web site to forward to Asterisk using IAX? In particular I am trying to find out where you set the url / ip address of your asterisk installation on the voip.ms web site. Thanks! On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.comwrote: I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Voicemail config
Hi
Is there a way to use variables in voicemail.conf.
I want to have an oncall tech system. The tech oncall has his number and email
set in astdb. When a tech call comes in the dial plan checks astdb and sets 2
variable ${oc} for the number and ${ocem} for the email. I can easily dial the
number using the variable, but if the call should go to voicemail I am not
getting the email. Below is my voicemail.conf entry for the mailbox. I can hard
code my email and it works so I know there is nothing wrong on the system.
121 = 121,Tech Support,${ocem},1113334...@vtext.net
Any help would be greatly appreciated.
Kelly--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
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Re: [asterisk-users] Voicemail config
If you change ${ocem} to ${ocem}@default, this will probably work as you
want it to.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Kelly opal
Sent: Tuesday, September 13, 2011 4:36 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voicemail config
Hi
Is there a way to use variables in voicemail.conf.
I want to have an oncall tech system. The tech oncall has his number and
email set in astdb. When a tech call comes in the dial plan checks astdb and
sets 2 variable ${oc} for the number and ${ocem} for the email. I can easily
dial the number using the variable, but if the call should go to voicemail I
am not getting the email. Below is my voicemail.conf entry for the mailbox.
I can hard code my email and it works so I know there is nothing wrong on
the system.
121 = 121,Tech Support,${ocem},1113334...@vtext.net
Any help would be greatly appreciated.
Kelly
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Reporting for Asterisk Call Center
actually Bilal, the Asteirsk CDR reports are placed on a different Database than the configurations .. you will need to install asterisk-addons which includes a module for cdr reporting to MYSQL DB. so you don't have to do the configs from the DB at all second.. in regards to the Flash Operator Panel you can have a look at a demo here: http://www.asternic.org/demo.php its a nice web interface gives you a live look at your call center .. who is active who is idle .. how many in Queue .. who is online and who is offline.. what trunks are busy ... etc third: theoretically you can set all Asterisk boxes to load from one database server (never done it myself).. actually it's one of the methods used for redundancy (somebody correct me if i'm wrong?). my only concern is with DB Management systems there is what we call it LOCK where a process locks the whole db or a part or it in order to do it's manipulation.. so i'm not sure if the database will be locked by one of the asterisk boxes when writing to it? which prevents the rest from writing to it at the same time? regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Tue, 13 Sep 2011 02:43:05 -0700 From: bilmar...@yahoo.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Reporting for Asterisk Call Center Dear Tareq; I am not using mysql, the configuration on the text configuratoin files and the logs are existed under the directory (/var/log/asterisk). Well, to use mysql: then it means the configuration will be also in the database or I can use mysql only for reporting? What is the Flash Operator? By the way, I have another question if you can help me if you used the database with sql, actually I was facing one time a case and maybe the Database usage will help me if you can advise me: If I have multiple Asterisk servers are running, and I need them to work centralized (I mean from one configuration) so to work as one system, then if I have database for configuration, I can acheive this by making all the servers read and write the configuration from the database server? Thanks for your help Tareq. Regards Bilal --- those reports can be easiely extracted from the MYSQL database my friend.. and you can add the Flash Operator Panel if you want to monitor live activities like how many in queue and how many ON CALL .. etc anyway the Elastix is a stand alone distribution you can find more info and downloads at : http://elastix.org/ regards Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Voicemail config
I would not use voicemail for this.
I would do the following.
Have the call come in.
Ring the on call tech with a dial.
If they don't pickup and press an accept key then.
Answer the call.
Record a message from the caller.
Use a script to e-mail the message to the tech.
(I would recommend you use some kind of ticketing system to e-mail to
that way you could have a progression if the on call tech does not respond
in a timely manner. )
This is just one possible way to address this. problem I can think of about
10 others depending on the actual requirements.
Thanks
Bryant Zimmerman (ZK Tech Inc.)
616-855-1030 Ext. 2003
From: Kelly opal ke...@ncwcom.com
Sent: Tuesday, September 13, 2011 5:35 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Voicemail config
Hi
Is there a way to use variables in voicemail.conf.
I want to have an oncall tech system. The tech oncall has his number and
email set in astdb. When a tech call comes in the dial plan checks astdb
and sets 2 variable ${oc} for the number and ${ocem} for the email. I can
easily dial the number using the variable, but if the call should go to
voicemail I am not getting the email. Below is my voicemail.conf entry for
the mailbox. I can hard code my email and it works so I know there is
nothing wrong on the system.
121 = 121,Tech Support,${ocem},1113334...@vtext.net
Any help would be greatly appreciated.
Kelly
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
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Re: [asterisk-users] Asterisk 1.8 not accepting call from DID
you didn't provide your dialplan for the incoming call context from_poland? nor registration string? could be a dial plan problem .. or codec issue.. as long as you register properly the server has no problem with NAT.. it's a routing or codec issue i think. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Mon, 5 Sep 2011 19:50:34 -0600 From: syscon...@gmail.com To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Asterisk 1.8 not accepting call from DID It seems to me nat=yes is not working correctly in asterisk 1.8.5 rtp set debug on shows: Got RTP packet from 10.0.0.110:6000 (type 00, seq 029667, ts 2129095321, len 000160) Sent RTP packet to 10.0.0.110:6010 (type 00, seq 065112, ts 2129095320, len 000160) I've tried 'nat=yes' 'nat=comedia' it makes no differece. -- Joseph On 09/05/11 15:00, Joseph wrote: I have DID, it registers OK with the provider, but when I try to call this number (it suppose to ring my Asterisk) asterisk 1.8 does not respond. sip show peers Name/username Host Dyn Forcerport ACL Port Status actio-out/48746612254 81.15.150.20 N 5060 OK (201ms) sip.conf part: [general] context=default allowguest=no allowoverlap=no udpbindaddr=0.0.0.0 useragent = Centrala [actio-out] type=friend secret= user=48746612254 username=48746612254 fromuser=48746612254 authname=48746612254 callerpage=48746612254 fromdomain=sip.actio.pl host=sip.actio.pl insecure=port,invite nat=yes qualify=yes dtmfmode=inband disallow=all allow=ulaw allow=alaw context=from_poland canreinvite=no The setting above worked OK with Asteriks 1.4 Here is debug info, which I don't know how to interpret. -- Executing [901148746612254@internal:1] Dial(SIP/11-0002, SIP/901148746612254@pstn-1270,60,tr) in new stack [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:25695 sip_request_call: Asked to create a SIP channel with formats: 0x4 (ulaw) == Using UDPTL CoS mark 5 [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:7496 sip_alloc: Allocating new SIP dialog for 5a2cdf8339e0ad2911ad393036c05165@127.0.0.1:0 - INVITE (No RTP) [Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:347 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x88c3b10' [Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:474 ast_rtp_new: Allocated port 16690 for RTP instance '0x88c3b10' [Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:356 ast_rtp_instance_new: RTP instance '0x88c3b10' is setup and ready to go [Sep 5 14:04:35] DEBUG[26209]: res_rtp_asterisk.c:2372 ast_rtp_prop_set: Setup RTCP on RTP instance '0x88c3b10' == Using SIP RTP CoS mark 5 [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4928 do_setnat: Setting NAT on RTP to Off [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:4936 do_setnat: Setting NAT on UDPTL to Off [Sep 5 14:04:35] DEBUG[26209]: rtp_engine.c:1459 ast_rtp_instance_early_bridge_make_compatible: Seeded SDP of 'SIP/pstn-1270-0003' with that of 'SIP/11-0002' [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALEDTIME. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable ANSWEREDTIME. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALEDPEERNAME. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALEDPEERNUMBER. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable DIALSTATUS. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable SIPCALLID. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable SIPDOMAIN. [Sep 5 14:04:35] DEBUG[26209]: channel.c:5989 ast_channel_inherit_variables: Not copying variable SIPURI. [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:5463 sip_call: Outgoing Call for 901148746612254 [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10989 add_sdp: ** Our capability: 0xc (ulaw|alaw) Video flag: False Text flag: False [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:10990 add_sdp: ** Our prefcodec: 0x4 (ulaw) [Sep 5 14:04:35] DEBUG[26209]: chan_sip.c:3054 initialize_initreq: Initializing initreq for method INVITE - callid 770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060 -- Called SIP/901148746612254@pstn-1270 [Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060' Request 102: Found [Sep 5 14:04:35] DEBUG[26083]: chan_sip.c:4053 __sip_semi_ack: (Provisional) Stopping retransmission (but retaining packet) on '770d283f78ef7d00782d2dd043212ed2@10.0.0.103:5060' Request 102: Found [Sep 5 14:04:35] DEBUG[26083]: rtp_engine.c:538
Re: [asterisk-users] Question about voip.ms service.
Voip.ms has excellent support if you need it, which many do not. You log in to your account, then you can change from SIP to IAX, and if you click on the correct link they will give you your sample with your account information You need to set up a registration line in IAX, then a context in IAX that points to a context in extensions.conf Registration takes care of voip.ms finding you their web site setup is about as complete a site as I have seen, with many more options than I would ever need The only somewhat confusing issue is when using IAX they will not show you as registered Your Asterisk will, though. John Novack naren wrote: That's what I am hoping to do as well. Could you share some insight on how you set up the DID on the voip.ms http://voip.ms web site to forward to Asterisk using IAX? In particular I am trying to find out where you set the url / ip address of your asterisk installation on the voip.ms http://voip.ms web site. Thanks! On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.com mailto:rhuddles...@gmail.com wrote: I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Anyone using Asterisk on VirtualBox ?
i did do some Asterisk tests on SUN VBOX .. works like a charm but you need to dedicate some good resources to the virtual box! Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 From: zhulizh...@live.com To: asterisk-users@lists.digium.com Date: Fri, 2 Sep 2011 08:37:55 + Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? hi: please check the redfone solution. Best regards, James.zhu Doing asterisk/PRI/ss7/dahdi, linux, asterisk cards, gateway(fxs/fxo/pri-SIP). website: www.voipviews.com From: aster...@a-domani.nl To: asterisk-users@lists.digium.com Date: Thu, 1 Sep 2011 23:48:46 +0200 Subject: Re: [asterisk-users] Anyone using Asterisk on VirtualBox ? On Thu, 2011-09-01 at 21:32 +0530, RSCL Mumbai wrote: My main interest of being on Virtual platform is portability / Backup. In case of any h/w issues, or crashes, simply copy the VM on to another box and you are up in minutes. Sanjay -- Doing that right now, although in my case i use XEN. Besides being hw independant, it is easier to play with a different version for a while (1.4 / 1.6.0 / 1.6.1 / 1.6.2 / 1.8.0) and being able to switch back in minutes. hw -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ?
try to look for N82 nokia mobile devices.. you get the benefits of a Mobile device with it's phone book and mobility features (games when you are bored :P) .. and other features.. and the native SIP client works fluently with no problems at all supporting almost commercial codecs like (G729).. and it works with WIFI.. i use it at home. Tarek Sawah Information Technology Adviser Integrated Digital Systems CCNP, MCSE, RHCE, TELECOM USA: +1 386 492 9993 Date: Sat, 27 Aug 2011 10:14:24 +0100 From: gordon+aster...@drogon.net To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] Good, inexpensive wireless VOIP handsets ? On Sat, 27 Aug 2011, Alan Lord (News) wrote: On 26/08/11 19:02, linux guy wrote: I'm looking for 4 to 6 good, inexpensive VOIP handsets for my home asterisk system. We've been using the Siemens Gigaset 685IP range for over three years and I'm (still) very pleased with them: +1 The base station is separate from the handsets - which is typically different from most DECT setups - the plus point is that you can position the base in a good location - ie. high on a wall, rather than anywhere else. Another plus is that the base has a single built-in ATA, so it can connect to the home PSTN line. The base also has an Ethernet socket to connect to the LAN and it can have up to 6 SIP accounts - each handset (up to 6) can be configured to ring on a particular SIP account or many SIP accounts and/or the PSTN line. Each handset has a default SIP account (or PSTN) to make outgoing calls on, but you can select any other SIP account or the PSTN by appending a code to the number you dial. They are very flexible - and being DECT, have superb range. I've installed many of these for my customers - typically the home office types - where they only want one phone on their desk - so the same handset can answer their home phone or their office SIP account, while providing wireless handsets throughout the rest of the house. A limitation is that one base can only handle 2 simultaneous SIP calls (plus a call via the PSTN), so if 2 phones are in-use, then the system can't take a 3rd call, however that's rarely a limitation in a domestic environment. Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
Ok that makes sense. I will take a look at my set up and see why it is not registering with voip.ms. I opened a ticket with voip.ms as well about an hour ago. I do like their service as well, that is why I want to try and get it working with them. Thanks John. On Tue, Sep 13, 2011 at 5:29 PM, John Novack jnov...@stromberg-carlson.orgwrote: Voip.ms has excellent support if you need it, which many do not. You log in to your account, then you can change from SIP to IAX, and if you click on the correct link they will give you your sample with your account information You need to set up a registration line in IAX, then a context in IAX that points to a context in extensions.conf Registration takes care of voip.ms finding you their web site setup is about as complete a site as I have seen, with many more options than I would ever need The only somewhat confusing issue is when using IAX they will not show you as registered Your Asterisk will, though. John Novack naren wrote: That's what I am hoping to do as well. Could you share some insight on how you set up the DID on the voip.ms web site to forward to Asterisk using IAX? In particular I am trying to find out where you set the url / ip address of your asterisk installation on the voip.ms web site. Thanks! On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.comwrote: I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] realtime goto/gotoif/dial
Hi all, I presume i made a silly mistake while filling a database But while googling on the results, i came across a lot of messages about the layout of app_data in case of goto and dial statements. (mostly about using the old | seperator instead of the , separator. So i was wondering, is this issue been solved? (I presume so, but can not find any confirmation about it) Hans -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Question about voip.ms service.
naren wrote: Ok that makes sense. I will take a look at my set up and see why it is not registering with voip.ms http://voip.ms. Understand that with IAX, voip.ms will not show you as registered. Your Asterisk should show you as registered from the CLI CLI iax2 show registry XX.XX.XX.XXX:4569 N Y ZZ.ZZ.ZZZ.ZZZ:4569 60 Registered X= Voip.ms server Y=your account number Z=Your IP address In IAX general section: register = Y:passw...@newyork.voip.ms ; change this to your server specified this is shown in their example, with your data filled in then your specific section for voipms: [voipms]; ; type=friend username=Y secret=PASSWORD context=from-voipms ; this points to your inbound context in extensions host=newyork.voip.ms disallow=all allow=ulaw ;Codec 1 supported allow=gsm ; Codec 2 supported insecure=port,invite ; from voip.ms example requirecalltoken=no ; required after 1.4.26 Hope this helps JN I opened a ticket with voip.ms http://voip.ms as well about an hour ago. I do like their service as well, that is why I want to try and get it working with them. Thanks John. On Tue, Sep 13, 2011 at 5:29 PM, John Novack jnov...@stromberg-carlson.org mailto:jnov...@stromberg-carlson.org wrote: Voip.ms has excellent support if you need it, which many do not. You log in to your account, then you can change from SIP to IAX, and if you click on the correct link they will give you your sample with your account information You need to set up a registration line in IAX, then a context in IAX that points to a context in extensions.conf Registration takes care of voip.ms http://voip.ms finding you their web site setup is about as complete a site as I have seen, with many more options than I would ever need The only somewhat confusing issue is when using IAX they will not show you as registered Your Asterisk will, though. John Novack naren wrote: That's what I am hoping to do as well. Could you share some insight on how you set up the DID on the voip.ms http://voip.ms web site to forward to Asterisk using IAX? In particular I am trying to find out where you set the url / ip address of your asterisk installation on the voip.ms http://voip.ms web site. Thanks! On Tue, Sep 13, 2011 at 4:19 PM, Robert-iPhone rhuddles...@gmail.com mailto:rhuddles...@gmail.com wrote: I'm using them for inbound and outbound on Asterisk and FreeSwitch Sent from my iPhone -- _ -- Bandwidth and Colocation Provided byhttp://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Dog is my Co-pilot -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] PRI Issues After Upgrade
I made the switch and everything seems to be working. It's hard to tell, since it never seems to fail for me, but fails once people get in. A question, though. When we moved the original box to our new office, they asked if we could support the dms100 setting, which worked. National seems to be working. Is there a big difference between the two? I'm just a simple programmer who happens to be the only IT guy in the office. - Stephen H. Gerstacker On Sep 13, 2011, at 10:23, Doug Lytle supp...@drdos.info wrote: Stephen H. Gerstacker wrote: Anything else I can try? Try switchtype=national just for testing. 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 Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] using variables in the shell function
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/custom/${TOPMENU})})
im trying to see if a file is available before playing the file
or does anybody have a different idea but not using agi
asterisk 1.6.2.20
thanks
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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Tue, Sep 13, 2011 at 6:47 PM, Matthew J. Roth mr...@imminc.com wrote: Kaushal, Your version of SoX does not have MP3 support. Since you have LAME installed, use it as a first step to produce an intermediate file that SoX supports. Then use SoX to convert the intermediate file to the desired format. Step 1 -- # lame --decode obd-demo.mp3 obd-demo.wav input: obd-demo.mp3 (8 kHz, 1 channel, MPEG-2.5 Layer III) output: obd-demo.wav (16 bit, Microsoft WAVE) skipping initial 1105 samples (encoder+decoder delay) Frame# 16818/16818 16 kbps # file obd-demo.wav obd-demo.wav: RIFF (little-endian) data, WAVE audio, Microsoft PCM, 16 bit, mono 8000 Hz Step 2 -- # sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.ulaw sox: Detected file format type: wav sox: WAV Chunk fmt sox: WAV Chunk data sox: Reading Wave file: Microsoft PCM format, 1 channel, 8000 samp/sec sox: 16000 byte/sec, 2 block align, 16 bits/samp, 19372126 data bytes sox: 9686063 Samps/chans sox: Input file obd-demo.wav: using sample rate 8000 size shorts, encoding signed (2's complement), 1 channel sox: Output file obd-demo.ulaw: using sample rate 8000 size bytes, encoding u-law, 1 channel sox: Output file: comment Processed by SoX Regards, Matthew Roth Thanks Matthew Roth. It worked. Also please let me know the difference between .ulaw and .alaw format and is there a way i can play this file formats. Regards Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using variables in the shell function
On Wed, 14 Sep 2011, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls /var/lib/asterisk/sounds/custom/${TOPMENU})})
im trying to see if a file is available before playing the file
or does anybody have a different idea but not using agi
Why not AGI?
They both ('shelling out' or calling an AGI) have the same 'impact' on
system resources.
You can even write an AGI in shell if you lack the skills for other
languages like C, PHP, or Perl.
You should be able to cobble up an AGI in PHP (or Perl, but I'm not much
of a Perl coder myself) just by cutting and pasting from some of the
examples on voip-info.org.
This simple task would be a great way for you to 'get your feet wet.'
What will you do if the file is not available?
--
Thanks in advance,
-
Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST
Newline Fax: +1-760-731-3000
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Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Wed, 14 Sep 2011, Kaushal Shriyan wrote: Also please let me know the difference between .ulaw and .alaw format and is there a way i can play this file formats. alaw = Europe, ulaw = US Japan Wikipedia has articles on both algorithms if you are interested in the specifics. If you format your file correctly, Asterisk can play it. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Wed, Sep 14, 2011 at 6:42 AM, Steve Edwards asterisk@sedwards.com wrote: On Wed, 14 Sep 2011, Kaushal Shriyan wrote: Also please let me know the difference between .ulaw and .alaw format and is there a way i can play this file formats. alaw = Europe, ulaw = US Japan Wikipedia has articles on both algorithms if you are interested in the specifics. If you format your file correctly, Asterisk can play it. -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users Hi Steve, I have carried out the below steps [root@host0040 test]# lame --decode obd-demo.mp3 obd-demo.wav input: obd-demo.mp3 (44.1 kHz, 1 channel, MPEG-1 Layer III) output: obd-demo.wav (16 bit, Microsoft WAVE) skipping initial 1105 samples (encoder+decoder delay) Frame# 887/886256 kbps [root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.alaw sox: Detected file format type: wav sox: WAV Chunk fmt sox: WAV Chunk data sox: Reading Wave file: Microsoft PCM format, 1 channel, 44100 samp/sec sox: 88200 byte/sec, 2 block align, 16 bits/samp, 2041438 data bytes sox: 1020719 Samps/chans sox: Input file obd-demo.wav: using sample rate 44100 size shorts, encoding signed (2's complement), 1 channel sox: Output file obd-demo.alaw: using sample rate 8000 size bytes, encoding u-law, 1 channel sox: Output file: comment Processed by SoX sox: resample opts: Kaiser window, cutoff 0.80, beta 16.00 [root@host0040 test]# ls -ltr total 2932 -rwxr-xr-x 1 root root 741459 Sep 14 06:32 obd-demo.mp3 -rw-r--r-- 1 root root 2041482 Sep 14 06:32 obd-demo.wav -rw-r--r-- 1 root root 185165 Sep 14 06:32 obd-demo.alaw Am i doing it correctly ? Please comment Regards, Kaushal -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] sox: Failed reading obd-demo.mp3: Do not understand format type: mp3
On Wed, 14 Sep 2011, Kaushal Shriyan wrote: I have carried out the below steps [root@host0040 test]# sox -V obd-demo.wav -r 8000 -b -t ul -c 1 obd-demo.alaw sox: Output file obd-demo.alaw: using sample rate 8000 size bytes, encoding u-law, 1 channel Sox v14.x complains about the '-b.' You are encoding as u-law, but naming the file alaw. Since [a|u]law are 'headerless' file formats, this will probably confuse Asterisk. -rwxr-xr-x 1 root root 741459 Sep 14 06:32 obd-demo.mp3 -rw-r--r-- 1 root root 2041482 Sep 14 06:32 obd-demo.wav -rw-r--r-- 1 root root 185165 Sep 14 06:32 obd-demo.alaw Am i doing it correctly ? Please comment I've never used alaw (I'm in the US). I don't think an MP3 needs 'execute' permissions :) -- Thanks in advance, - Steve Edwards sedwa...@sedwards.com Voice: +1-760-468-3867 PST Newline Fax: +1-760-731-3000 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] using variables in the shell function
On 09/13/2011 07:49 PM, Israel Gottlieb wrote:
is it possible to pas variables to the shell function
Set(recordingavail=${SHELL(ls
/var/lib/asterisk/sounds/custom/${TOPMENU})})
im trying to see if a file is available before playing the file
or does anybody have a different idea but not using agi
asterisk 1.6.2.20
thanks
You should check out the STAT function.
core show function STAT
This should evaluate to 1
${STAT(e,/var/lib/asterisk/sounds/en/vm-goodbye.gsm)})
This should evaluate to 0
${STAT(e,/var/lib/asterisk/sounds/en/xyzzy.gsm)}
Dale
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Re: [asterisk-users] Reporting for Asterisk Call Center
Hi Tzafrir, On Sat, Sep 10, 2011 at 4:28 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote: On Fri, Sep 09, 2011 at 01:28:28PM -0500, Gerardo Barajas wrote: There are a lot of reporting tools. I have used: Asternic: http://www.asternic.biz/ Non of those are Free (Open Source). Clarification: Asternic Call Center Stats Lite is free (GPL3) and can be downloaded from the above link. The PRO version is commercial. Asternic CDR reports for FreePBX is also free and available for download on the asternic.biz site. -- Nicolás Gudiño I'm speaking at ElastixWorld: http://www.elastixworld.com/2011/ I'm speaking at 4K Conference: http://www.4kconf.com -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
