[asterisk-users] The plain old PBX functionality

[Gergo Csibra]

Hi,

back in the old analog telephony days there was "digital" PBX-es and
digital "system" phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user can transfer the call with one touch (pressing
one of this button).

I search this functionality in Asterisk. What versions, and what
extension functions (or other settings), and what VoIP phones can do
this?


-- 
Best regards,
 Gergomailto:csi...@gmail.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

Re: [asterisk-users] Calls not hanging up

[Markus]

Am 08.08.2014 05:13, schrieb D'Arcy J.M. Cain:

New data point - I just reverted to 11.10.2 without a single change to
the configuration and the problem has gone away.


Hmm. Could this have to do with session-timers (sip.conf)?

I remember when I went from 1.4 to 10.7 I had to manually mess with the 
session-timers because my peers who delivered incoming calls would 
always end the call after 30 minutes. But your problem is kind of the 
opposite. :)


Just a shot in the dark, without knowing much about SIP really, lol.

If you really wanna know, you should fire up tcpdump and see what's 
going on there.



--
_
-- 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] Calls not hanging up

[D'Arcy J.M. Cain]

On Thu, 7 Aug 2014 10:12:02 -0400
"D'Arcy J.M. Cain"  wrote:
> This just started after upgrading to 11.11.0.  After a call is
> completed (both ends hang up) the call still shows as active.

New data point - I just reverted to 11.10.2 without a single change to
the configuration and the problem has gone away.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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 too many files or memory leak???

[D'Arcy J.M. Cain]

On Thu, 7 Aug 2014 22:00:47 -0400
Jerry Geis  wrote:
> :[Aug  7 21:35:24] ERROR[19582] acl.c: Cannot create socket
> [Aug  7 21:35:24] WARNING[19582][C-0283] res_rtp_asterisk.c:
> Unable to allocate RTP socket: Too many open files
...
> I am running asterisk 11.11.0

Shot in the dark here but does "core show channels" show an inordinate
number of channels, especially channels that you know should be closed?

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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] asterisk too many files or memory leak???

[Jerry Geis]

I am seeing this in my log file

:[Aug  7 21:35:24] ERROR[19582] acl.c: Cannot create socket
[Aug  7 21:35:24] WARNING[19582][C-0283] res_rtp_asterisk.c: Unable to
allocate RTP socket: Too many open files
[Aug  7 21:35:24] NOTICE[19582][C-0283] chan_sip.c: Failed to
authenticate device "677";tag=3637370132313231383238343335
[Aug  7 21:35:24] WARNING[19734] manager.c: Failed to create temporary file
for command: Too many open files

Just wondering... was surprised to see this.
There are 999 open sockets in my asterisk process. That is not normal.
I know I can increase that ulimit - but I only have about 50 phones on the
system.
just wondering if there is an issue.
I am running asterisk 11.11.0

Jerry
-- 
_
-- 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] agi get_data noanswer

[Eric Wieling]

Generally the only thing you are allowed to do before answer is send audio.  
You can’t receive audio and can’t receive DTMF.   I assume it is to prevent 
people from doing exactly what you  are trying to do --- trying to have two way 
communications without paying for the call.


From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael Visser
Sent: Thursday, August 07, 2014 4:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] agi get_data noanswer

Hi Guys..
I am making an anoucement machine that is not allowed to "answer" the call due 
to a billing issue.
I found that Playback with "noanwser" is usefull in this case.

$AGI->exec('Playback',"$message","noanswer")}


But when i request some values to the user with get_data, i think there is an 
answer anywere.

Is there a way to get_data without answering the call?

Thanks in advance!!

rv
-- 
_
-- 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] agi get_data noanswer

[Rafael Visser]

Hi John.
I am making an inteligent annoucement resouce for a big ericsson switch. Is
just an ivr with agi applications.
The tricky thing try to make asterisk not to send answer. The perl
application with agi commands must be executed with out answering.

Something like

exten => 6009,1,Progress()
exten => 6009,n,Set(__INICIA=${EPOCH})
exten => 6009,n,Set(CHANNEL(language)=sc)
exten => 6009,n,AGI(anouncement.pl)
exten => 6009,n,Hangup()

Thanks anyway.
rv





2014-08-07 17:11 GMT-04:00 Tech Support :

> What you may want to check out is the PlayTones and Ringing
> applications in your dial plan. Asterisk will answer the call, but your
> users won't know that because all they hear is the call still ringing.
> After a certain amount of time passes, you can send them directly to
> voicemail, hangup, run your scripts, or anything else you want to do with
> the call. My dial plan snippet looks like this. Just an option.
>
>
>
> exten => s,n(ringing),Answer
>
> exten => s,n,PlayTones(ring)
>
> exten => s,n,Ringing
>
> exten => s,n,Wait(${TIMEOUT})
>
> exten => s,n,GotoIf($["${BLOCKDEST}" = "3"]?s-NA-VOICEMAIL,1)
>
> exten => s,n,GotoIf($["${CUSTCALLBLOCKACTION}" = "3"]?s-NA-VOICEMAIL,1) ;
>
> exten => s,n,PlayTones(congestion)
>
> exten => s,n,Congestion(10)
>
> exten => s,n,Hangup
>
>
>
> Regards;
>
> John V.
>
>
>
> *From:* asterisk-users-boun...@lists.digium.com [mailto:
> asterisk-users-boun...@lists.digium.com] *On Behalf Of *Rafael Visser
> *Sent:* Thursday, August 07, 2014 4:56 PM
> *To:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Subject:* [asterisk-users] agi get_data noanswer
>
>
>
> Hi Guys..
> I am making an anoucement machine that is not allowed to "answer" the call
> due to a billing issue.
> I found that Playback with "noanwser" is usefull in this case.
>
> $AGI->exec('Playback',"$message","noanswer")}
>
>
> But when i request some values to the user with get_data, i think there is
> an answer anywere.
>
> Is there a way to get_data without answering the call?
>
> Thanks in advance!!
>
> rv
>
> --
> _
> -- 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] agi get_data noanswer

[Tech Support]

What you may want to check out is the PlayTones and Ringing applications in 
your dial plan. Asterisk will answer the call, but your users won't know that 
because all they hear is the call still ringing. After a certain amount of time 
passes, you can send them directly to voicemail, hangup, run your scripts, or 
anything else you want to do with the call. My dial plan snippet looks like 
this. Just an option. 

 

exten => s,n(ringing),Answer

exten => s,n,PlayTones(ring)

exten => s,n,Ringing

exten => s,n,Wait(${TIMEOUT})

exten => s,n,GotoIf($["${BLOCKDEST}" = "3"]?s-NA-VOICEMAIL,1)

exten => s,n,GotoIf($["${CUSTCALLBLOCKACTION}" = "3"]?s-NA-VOICEMAIL,1) ; 

exten => s,n,PlayTones(congestion)

exten => s,n,Congestion(10)

exten => s,n,Hangup

 

Regards;

John V.

 

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Rafael Visser
Sent: Thursday, August 07, 2014 4:56 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] agi get_data noanswer

 

Hi Guys..
I am making an anoucement machine that is not allowed to "answer" the call due 
to a billing issue.
I found that Playback with "noanwser" is usefull in this case.

$AGI->exec('Playback',"$message","noanswer")}


But when i request some values to the user with get_data, i think there is an 
answer anywere.

Is there a way to get_data without answering the call?

Thanks in advance!!

rv

-- 
_
-- 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] agi get_data noanswer

[Rafael Visser]

Hi Guys..
I am making an anoucement machine that is not allowed to "answer" the call
due to a billing issue.
I found that Playback with "noanwser" is usefull in this case.

$AGI->exec('Playback',"$message","noanswer")}


But when i request some values to the user with get_data, i think there is
an answer anywere.

Is there a way to get_data without answering the call?

Thanks in advance!!

rv
-- 
_
-- 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] enable features

[Shishir Pokharel]

http://www.voip-info.org/wiki/view/PBX+Do+Not+Disturb

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of aris tsitras
Sent: Thursday, August 07, 2014 1:35 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] enable features

may i have an example of what you are describing?


On 7/8/2014 23:13, Shishir Pokharel wrote:
Uncommenting features.conf is not sufficient,  You got to have some logic on 
the dialplan to support what you are  asking for. If I were you, I would 
probably use some dial plan logic with asterisk internal DB .

From: 
asterisk-users-boun...@lists.digium.com
 [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aristeidis 
Tsitras
Sent: Thursday, August 07, 2014 12:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] enable features

i do have asterisk 1.8 (no gui, no distro based) and i would like to enable 
some features:
-call forward (conditional, unconditional,...)
-DND
-call waiting
-attended transfer
-follow me


all the features i would like to enable/disable them through digit codes such 
#45# and *45.
all these fetures should apply to asterisk only and not use the features from 
the service provider.

i have edited the /etc/asterisk/features.conf file and uncommented the option 
for attended transfer (*2). the thing is that it did not work. is there 
something else that  i have to write to sip/extensions.conf?






-- 
_
-- 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] enable features

[aris tsitras]

may i have an example of what you are describing?



On 7/8/2014 23:13, Shishir Pokharel wrote:


Uncommenting features.conf is not sufficient,  You got to have some 
logic on the dialplan to support what you are  asking for. If I were 
you, I would probably use some dial plan logic with asterisk internal DB .


*From:*asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] *On Behalf Of 
*Aristeidis Tsitras

*Sent:* Thursday, August 07, 2014 12:29 AM
*To:* asterisk-users@lists.digium.com
*Subject:* [asterisk-users] enable features

i do have asterisk 1.8 (no gui, no distro based) and i would like to 
enable some features:


-call forward (conditional, unconditional,...)

-DND

-call waiting

-attended transfer

-follow me

all the features i would like to enable/disable them through digit 
codes such #45# and *45.


all these fetures should apply to asterisk only and not use the 
features from the service provider.


i have edited the /etc/asterisk/features.conf file and uncommented the 
option for attended transfer (*2). the thing is that it did not work. 
is there something else that  i have to write to sip/extensions.conf?






-- 
_
-- 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] enable features

[Shishir Pokharel]

Uncommenting features.conf is not sufficient,  You got to have some logic on 
the dialplan to support what you are  asking for. If I were you, I would 
probably use some dial plan logic with asterisk internal DB .

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Aristeidis Tsitras
Sent: Thursday, August 07, 2014 12:29 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] enable features

i do have asterisk 1.8 (no gui, no distro based) and i would like to enable 
some features:
-call forward (conditional, unconditional,...)
-DND
-call waiting
-attended transfer
-follow me


all the features i would like to enable/disable them through digit codes such 
#45# and *45.
all these fetures should apply to asterisk only and not use the features from 
the service provider.

i have edited the /etc/asterisk/features.conf file and uncommented the option 
for attended transfer (*2). the thing is that it did not work. is there 
something else that  i have to write to sip/extensions.conf?



-- 
_
-- 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] Is it possible to set asterisk's VoIP authentication to be based on EAP-SIM auth of freeradius?

[Shishir Pokharel]

You can use sip proxy servers  on top of asterisk server to have a 
authentication from freeradius, at this point I don’t think asterisk supports 
what you are looking for.

Try this
http://www.opensips.org/Documentation/Tutorials-Radius

From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of rafa alfurqan
Sent: Thursday, August 07, 2014 12:30 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Is it possible to set asterisk's VoIP authentication 
to be based on EAP-SIM auth of freeradius?

Hi all,

I want to make initial VoIP authentication process from asterisk server to be 
based on EAP-SIM authentication of Freeradius server (so it will be not 
necessary to insert account datas in the asterisk database). Is there any way 
of doing that from Freeradius and Asterisk? Or at least, is there any way to 
sync the EAP-SIM data on Freeradius to asterisk server?

thank you
-- 
_
-- 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] The plain old PBX functionality

[Kevin Larsen]

> back in the old analog telephony days there was "digital" PBX-es and
> digital "system" phonesets. This phonesets have had many individual
> illuminatable buttons connected with extensions. The PBX can show on
> the buttons if some extension is ringing (blinks) or busy (constant
> light), and the user can transfer the call with one touch (pressing
> one of this button).
> 
> I search this functionality in Asterisk. What versions, and what
> extension functions (or other settings), and what VoIP phones can do
> this?

Asterisk has had this functionality for a long time. The terms you want to 
search for are BLF (Busy Lamp Field) and Subscribe. I imagine that most 
sip phones have the necessary features to do BLF. I know the Polycom 
phones I use certainly do. The Digium branded phones do as well.-- 
_
-- 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] The plain old PBX functionality

[Gergo Csibra]

Hi,

back in the old analog telephony days there was "digital" PBX-es and
digital "system" phonesets. This phonesets have had many individual
illuminatable buttons connected with extensions. The PBX can show on
the buttons if some extension is ringing (blinks) or busy (constant
light), and the user can transfer the call with one touch (pressing
one of this button).

I search this functionality in Asterisk. What versions, and what
extension functions (or other settings), and what VoIP phones can do
this?

-- 
Best regards,
 Gergo  mailto:csi...@gmail.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

[asterisk-users] multicastRTp

[Jerry Geis]

I am using a cyberdata "sip paging adapter" and with the
Dial(MulticastRTP/basic/IP:port) and with
tshark I see the RTP data, my device looks like its accepting the data
and I hear a click for my relay on my device so it would seem its accepting
the call,
however - I hear no audio...

Asterisk 11.11.0 is what I am using.
What might be wrong here?
Thanks,

jerry
-- 
_
-- 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] Calls not hanging up

[Andres]

On 8/7/14, 12:14 PM, D'Arcy J.M. Cain wrote:

On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad  wrote:

Your call is up on VoiceMail you should check dialstatus before
sending user to VoiceMail.

I removed the voicemail command from the dialplan and it was exactly
the same behaviour.


You have 3 ways to automatically hang up a call.
1)  Caller hangs up
2)  Callee hangs up
3)  Timeout hangs up call

I suggest you capture the SIP messages to see if the hangup messages are 
not reaching Asterisk (caller or callee).  It is also a good idea to 
place a hard limit on calls so they hangup by timeout and not stay there 
forever.  From the DIAL comand:


L(x[:y[:z]]):
x - Maximum call time, in milliseconds
y - Warning time, in milliseconds
z - Repeat time, in milliseconds
Limit the call to  milliseconds. Play a warning when  mill
iseconds are left. Repeat the warning every  milliseconds until time
expires.


--
Technical Support
http://www.cellroute.net


--
_
-- 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] Calls not hanging up

[D'Arcy J.M. Cain]

On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad  wrote:
> Your call is up on VoiceMail you should check dialstatus before
> sending user to VoiceMail.

I removed the voicemail command from the dialplan and it was exactly
the same behaviour.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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] *SOLVED* Re: Anyone have any experience with inbound SIP trunks from Simwood?

[Steve Edwards]

On Thu, 7 Aug 2014, A J Stiles wrote:


.  And my mistake was in sip.conf.  The configuration stanza I had named
"simwood_in_slough" should, of course, have been named after the number I had
programmed in at the other end of the trunk .

*hangs head in shame*


It's OK. We're all a little 'slow' from time to time.

--
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] Calls not hanging up

[D'Arcy J.M. Cain]

On Thu, 7 Aug 2014 17:12:40 +0200
Asghar Mohammad  wrote:
> Your call is up on VoiceMail you should check dialstatus before
> sending user to VoiceMail.

so
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Application_Dial
is incorrect now?  That page says:

"Unless there is a timeout specified, the Dial application will wait
indefinitely until one of the called channels answers, the user hangs
up, or if all of the called channels are busy or unavailable. Dialplan
executing will continue if no requested channels can be called, or if
the timeout expires. This application will report normal termination if
the originating channel hangs up, or if the call is bridged and either
of the parties in the bridge ends the call."

The second sentence implies that the dialplan will not continue, i.e.
will not go to VM, if the call is answered.  The third sentence
reinforces that interpretation. That's certainly what happened in 11.10.
I didn't see anything in the change logs that would suggest such a
drastic change in behaviour.

-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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] Calls not hanging up

[Asghar Mohammad]

Your call is up on VoiceMail you should check dialstatus before sending
user to VoiceMail.


On Thu, Aug 7, 2014 at 4:12 PM, D'Arcy J.M. Cain  wrote:

> This just started after upgrading to 11.11.0.  After a call is
> completed (both ends hang up) the call still shows as active.
>
> # asterisk -x "core show channels"
> Channel  Location State   Application(Data)
> SIP/thinktel-000 (None)   Up  AppDial((Outgoing
> Line)) SIP/4164251212-0 416555@LocalSets Up
> Dial(SIP/thinktel/416555) 2 active channels
> 1 active call
> 1 call processed
>
> The 1212 number is mine and is hung up.  I even rebooted my ATA to make
> sure that it wasn't holding the line.  My dialplan is extremely
> simple.  In fact, I even simplified it from what it was for this
> testing.  Here it is.
>
> exten => 4164251212,1,Verbose(0, ${CALLERID(all)} Calling ${EXTEN})
> same => n,Dial(SIP/4164251212,30)
> same => n,VoiceMail(4164251212@LocalSets,u)
> same => n,Hangup()
>
> I can post any other log or config excerpts if someone thinks that they
> are relevant but all of this was working under 11.10.2.
>
> Thanks.
>
>
> --
> D'Arcy J.M. Cain
> System Administrator, Vex.Net
> http://www.Vex.Net/ IM:da...@vex.net
> VoIP: sip:da...@vex.net
>
> --
> _
> -- 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] *SOLVED* Re: Anyone have any experience with inbound SIP trunks from Simwood?

[A J Stiles]

On Wednesday 06 Aug 2014, I wrote:
> I'm trying -- unsuccessfully! -- to configure an inbound trunk with
> Simwood, and I was hoping someone on this list might have managed to do
> this.
> 
> I have configured some numbers to route to a SIP endpoint
>   %e164@customer's server
> and convinced the customer to open up UDP ports 5060 and 1 - 2.
> 
> Calling the number gets a SIP request from Simwood.  The customer's machine
> then sends a SIP 401 response.  Simwood send an ACK .  and then
> nothing. Nothing appears in the Asterisk CLI; to get the SIP trace I used
> the command
> 
> # ngrep -t -q -n -q -Wbyline -deth0 1283 port 5060
> 
> (note that 1283 = the STD code from which the call is originating, so it
> should show up in any related packets.)
> 
> 
> ##  sip.conf  ##
> [simwood_in_slough]
> type=friend
> host=178.22.140.34
> fromdomain=178.22.140.34
> permit=178.22.140.34/255.255.255.255
> qualify=no
> context=from-simwood
> dtmfmode=rfc2833
> insecure=invite,port
> disallow=all
> allow=alaw
> nat=yes
> directmedia=no

.  And my mistake was in sip.conf.  The configuration stanza I had named 
"simwood_in_slough" should, of course, have been named after the number I had 
programmed in at the other end of the trunk .

*hangs head in shame*

-- 
AJS

Note:  Originating address only accepts e-mail from list!  If replying off-
list, change address to asterisk1list at earthshod dot co dot uk .

-- 
_
-- 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] Calls not hanging up

[D'Arcy J.M. Cain]

This just started after upgrading to 11.11.0.  After a call is
completed (both ends hang up) the call still shows as active.

# asterisk -x "core show channels"
Channel  Location State   Application(Data)
SIP/thinktel-000 (None)   Up  AppDial((Outgoing
Line)) SIP/4164251212-0 416555@LocalSets Up
Dial(SIP/thinktel/416555) 2 active channels
1 active call
1 call processed

The 1212 number is mine and is hung up.  I even rebooted my ATA to make
sure that it wasn't holding the line.  My dialplan is extremely
simple.  In fact, I even simplified it from what it was for this
testing.  Here it is.

exten => 4164251212,1,Verbose(0, ${CALLERID(all)} Calling ${EXTEN})
same => n,Dial(SIP/4164251212,30)
same => n,VoiceMail(4164251212@LocalSets,u)
same => n,Hangup()

I can post any other log or config excerpts if someone thinks that they
are relevant but all of this was working under 11.10.2.

Thanks.


-- 
D'Arcy J.M. Cain
System Administrator, Vex.Net
http://www.Vex.Net/ IM:da...@vex.net
VoIP: sip:da...@vex.net

-- 
_
-- 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] enable features

[Scott Griepentrog]

To enable transfers using in-call DTMF sequences, you'll need to use the t
and/or T options in the Dial() command that initiates the call.  For
details see:

https://wiki.asterisk.org/wiki/display/AST/Application_Dial




On Thu, Aug 7, 2014 at 2:29 AM, Aristeidis Tsitras 
wrote:

> i do have asterisk 1.8 (no gui, no distro based) and i would like to
> enable some features:
> -call forward (conditional, unconditional,...)
> -DND
> -call waiting
> -attended transfer
> -follow me
>
>
> all the features i would like to enable/disable them through digit codes
> such #45# and *45.
> all these fetures should apply to asterisk only and not use the features
> from the service provider.
>
> i have edited the /etc/asterisk/features.conf file and uncommented the
> option for attended transfer (*2). the thing is that it did not work. is
> there something else that  i have to write to sip/extensions.conf?
>
>
>
>
> --
> _
> -- 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
>



-- 
[image: Digium logo]
Scott Griepentrog
Digium, Inc · Software Developer
445 Jan Davis Drive NW · Huntsville, AL 35806 · US
direct/fax: +1 256 428 6239 · mobile: +1 317 507 4029
Check us out at: http://digium.com · http://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] Dahdi > CAPI migration

[Patrick Laimbock]

Hi Toney,

Comments inline.

On 07-08-14 12:10, Toney Mareo wrote:

Hello Folks,
I looking to migrate a pbx from one server to another. The original server has 
this ISDN card:

00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller 
[HFC-4S] (rev 01)

The new server:
00:00.0 I2O: Digital Equipment Corporation StrongARM DC21285 (rev 04) << AVM 
ISDN Controller C4


Now I'm not fully aware of both's cards functions since the manuals are very brief. What 
I heard is that the later card does not support some NT commands what I might going to 
need at this migration. I can't really say if one card is "better" than the 
other one.
Unfortunately the two uses completely different drivers. The first card uses 
dahdi, the second uses capi. I want this migration go as smooth as possible 
with the least downtime so I looking for some help
maybe someone has more experience with these cards.


In the past I have used Eicon Diva Server cards with chan_capi for years 
and they worked great. I



I also read something about CAPI uses completely different dial plans.
So all the Asterisk configurations are migrated from the old server to the new 
one.


That is correct although the differences/changes you need to make are 
limited. Adjusting the Dial command will go a long way.



By following this guide:
http://www.asteriskguru.com/tutorials/avm_c4.html

I have the C4 modul loaded. My asterisk box is Asterisk 1.6.0.26-FONCORE-r78 
(Trixbox).


That Asterisk version is rather old and assuming that's the old Trixbox 
CE it is without security updates since 2012. I recommend you use a 
fresh install of something like CentOS 6.5 + all updates and Asterisk 
11.11.0 (or later if available) with the latest dahdi-linux, dahdi-tools 
and libpri releases. Also get the latest chan_capi from here: 
ftp://ftp.chan-capi.org/chan-capi/ The version with -HEAD in the name 
has the latest fixes and is the one I always used.



How can I see that this C4 card is really working from asterisk?


Once you have the AVM C4 kernel module loaded and Asterisk with 
chan_capi is installed and capi is enabled in the Asterisk config, start 
Asterisk and then you should have a 'capi' command in the CLI. Executing 
it should show you info about the status of your ISDN channels.



What is the difference between the chan_dahdi.conf and chan_capi.conf?


One is for DAHDI supported cards and the other is for CAPI cards like 
the Eicon Diva Server and the AVM C4.



Can't I just tell it somewhere to use the new card and I don't have to touch 
the existing dialplans etc?


Nope.

HTH,
Patrick

--
_
-- 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] Dahdi > CAPI migration

[Toney Mareo]

Hello Folks,
I looking to migrate a pbx from one server to another. The original server has 
this ISDN card:
 
00:00.0 ISDN controller: Cologne Chip Designs GmbH ISDN network Controller 
[HFC-4S] (rev 01)
 
The new server:
00:00.0 I2O: Digital Equipment Corporation StrongARM DC21285 (rev 04) << AVM 
ISDN Controller C4


Now I'm not fully aware of both's cards functions since the manuals are very 
brief. What I heard is that the later card does not support some NT commands 
what I might going to need at this migration. I can't really say if one card is 
"better" than the other one.
Unfortunately the two uses completely different drivers. The first card uses 
dahdi, the second uses capi. I want this migration go as smooth as possible 
with the least downtime so I looking for some help
maybe someone has more experience with these cards.
 
I also read something about CAPI uses completely different dial plans.
So all the Asterisk configurations are migrated from the old server to the new 
one.

By following this guide:
http://www.asteriskguru.com/tutorials/avm_c4.html

I have the C4 modul loaded. My asterisk box is Asterisk 1.6.0.26-FONCORE-r78 
(Trixbox).

How can I see that this C4 card is really working from asterisk?

What is the difference between the chan_dahdi.conf and chan_capi.conf?

Can't I just tell it somewhere to use the new card and I don't have to touch 
the existing dialplans etc?


Thanks!
 

-- 
_
-- 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] Is it possible to set asterisk's VoIP authentication to be based on EAP-SIM auth of freeradius?

[rafa alfurqan]

Hi all,

I want to make initial VoIP authentication process from asterisk server to
be based on EAP-SIM authentication of Freeradius server (so it will be not
necessary to insert account datas in the asterisk database). Is there any
way of doing that from Freeradius and Asterisk? Or at least, is there any
way to sync the EAP-SIM data on Freeradius to asterisk server?

thank you
-- 
_
-- 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] enable features

[Aristeidis Tsitras]

i do have asterisk 1.8 (no gui, no distro based) and i would like to enable 
some features:-call forward (conditional, unconditional,...)-DND-call 
waiting-attended transfer-follow me

all the features i would like to enable/disable them through digit codes such 
#45# and *45.all these fetures should apply to asterisk only and not use the 
features from the service provider.
i have edited the /etc/asterisk/features.conf file and uncommented the option 
for attended transfer (*2). the thing is that it did not work. is there 
something else that  i have to write to sip/extensions.conf?


  -- 
_
-- 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