Re: [asterisk-users] custom automated meeting

[Thanasis]

on 11/02/2011 07:44 AM Sammy Govind wrote the following:
> core show application meetme

Thanks!
(I am new to asterisk, and just learning, so forgive my dumb questions)

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Re: [asterisk-users] custom automated meeting

[Sammy Govind]

Type in asterisk CLI>core show application meetme
or google "asterisk cmd meetme" simple?

On Tue, Nov 1, 2011 at 10:33 PM, Thanasis  wrote:

> on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
> > You need simple dialplan of four steps:
> > same =>n,Set(conf_name=conf-${RAND(1,1000)})
> > same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
> > same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
> > same =>n,MeetMe(${conf_name},dFI1xAC)
> > same =>n,Noop(do post conference stuff)
> >
>
> Thanks!
> What is the meaning of the options dFI1xAC passed to
> app,MeetMe,${conf_name} ?
> Where can I find them described please?
>
> >
> > 2011/10/31 Thanasis :
> >> I need your help in implementing the following scenario:
> >>
> >> A certain extension will ring two sip phones simultaneously and when one
> >> of them answers, the other keeps ringing until it answers too, and then
> >> all three (the caller and the other two) are immediately placed in a
> >> conference room (same room for all three).
> >>
> >> Can we do it?
> >>
>
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[asterisk-users] FFA - Asterisk 1.6.2.6

[Christian Tardif]

Hi,

I have a 1.6.2.6 fax installation with a FFA license which seems to be 
installed correctly (in fax show stats, I see that I have 1 Digium 
G.711  licensed channel, and 1 Digium T.38 licensed channel).


When trying to call my business line with a fax machine, it looks like 
it's ringing to my asterisk box, then transfer the call to my extension. 
In the logs, I see (after the line where it says that my extension is 
ringing): chan_sip.c: Fax detected but no fax extension.


How come does Asterisk even try to ring my phone? It seems that the 
detection (which should append BEFORE any phone ring) does not work, and 
I have no clue where to look at.


In case this helps, I'm configuring the installation with FreePBX 2.8.1.4

Thanks,

--
Christian Tardif


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Re: [asterisk-users] State of Asterisk+Virtualization+Timing

[Warren Selby]

Look at upgrading to at least 1.6.2 or 1.8, these both have newer timing 
sources that don't rely on dahdi. Also, if conferencing is a big deal, look at 
10, this contains a complete rewrite of ConfBridge which doesn't require dahdi 
for mixing at all. 

Thanks,
--Warren Selby, dCAP

On Nov 1, 2011, at 12:08 PM, Tim Nelson  wrote:

> Greetings-
> 
> I'm about to dive into the process of virtualizing some of my Asterisk 
> (primarily 1.4.x) infrastructure. In the past, when looking at virt 
> solutions, the primary issue preventing me from moving was the lack of proper 
> timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like 
> to use either OpenVZ or KVM, but each seem to have independent "issues" that 
> need to be addressed:
> 
> OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant 
> access to host node timing source (physical device, or dahdi_dummy in 
> /dev/dahdi/) to the containerized Asterisk process.
> 
> KVM - Higher overhead, easier installation, 'true virtualization'. Primary 
> issue is not timing per se, but KVM scheduling. Timing source, while present 
> from dahdi_dummy natively may still not get proper scheduling by KVM process. 
> This could also affect general call quality (even non IAX2 trunked voice), 
> DTMF, etc.
> 
> I have to believe there are others running virtualized Asterisk installations 
> with some degree of success on OpenVZ or KVM. Care to share your thoughts?
> 
> --Tim
> 
> --
> _
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Re: [asterisk-users] State of Asterisk+Virtualization+Timing

[Nick Khamis]

It would be nice if we can get it going with KVM. Cloud computing solutions
are moving towards the true linux based kernel vs. FreeBSD of XEN.

Cheers,

Nick.



On Tue, Nov 1, 2011 at 5:46 PM, Johan Wilfer  wrote:
> 2011-11-01 18:08, Tim Nelson skrev:
>
> Greetings-
>
> I'm about to dive into the process of virtualizing some of my Asterisk
> (primarily 1.4.x) infrastructure. In the past, when looking at virt
> solutions, the primary issue preventing me from moving was the lack of
> proper timing. We do not need it for MeetMe but rather for IAX2 trunking.
> I'd like to use either OpenVZ or KVM, but each seem to have independent
> "issues" that need to be addressed:
>
> OpenVZ - Better resource usage, lower overhead. Primary issue is how to
> grant access to host node timing source (physical device, or dahdi_dummy in
> /dev/dahdi/) to the containerized Asterisk process.
>
> KVM - Higher overhead, easier installation, 'true virtualization'. Primary
> issue is not timing per se, but KVM scheduling. Timing source, while present
> from dahdi_dummy natively may still not get proper scheduling by KVM
> process. This could also affect general call quality (even non IAX2 trunked
> voice), DTMF, etc.
>
> I have to believe there are others running virtualized Asterisk
> installations with some degree of success on OpenVZ or KVM. Care to share
> your thoughts?
>
>
> Hi Tim,
> I'm using OpenVZ, it works very well.
> Take a look at: http://wiki.openvz.org/Asterisk_from_source
>
> You will have to compile dahdi and install it on the HN, and then you
> compile and install asterisk in the containers.
> I've not done this with chan_iax, but with meetme. In the case with meetme
> you would have to move some files over to trick asterisk that dahdi is
> compiled on the machine. The wiki mentions copying user.h.
>
> I used this as a starting point, some years ago:
> http://www.telephreak.org/papers/vpa/
> This paper covers vserver, so it's not exactly the same - but the steps with
> tonezone was the same when I built the current server running this
> configuration.
>
> I'm in the process of building another server with openvz, so I'll need to
> refresh my memory and try to document the procedure.
>
> --
> Med vänlig hälsning
>
> Johan Wilfer email: jo...@jttech.se
> JT Tech | Utvecklare webb: http://jttech.se
> direkt: +46 31 380 91 01  support: +46 31 380 91 00
>
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Re: [asterisk-users] State of Asterisk+Virtualization+Timing

[Johan Wilfer]

2011-11-01 18:08, Tim Nelson skrev:
> Greetings-
>
> I'm about to dive into the process of virtualizing some of my Asterisk 
> (primarily 1.4.x) infrastructure. In the past, when looking at virt 
> solutions, the primary issue preventing me from moving was the lack of proper 
> timing. We do not need it for MeetMe but rather for IAX2 trunking. I'd like 
> to use either OpenVZ or KVM, but each seem to have independent "issues" that 
> need to be addressed:
>
> OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant 
> access to host node timing source (physical device, or dahdi_dummy in 
> /dev/dahdi/) to the containerized Asterisk process.
>
> KVM - Higher overhead, easier installation, 'true virtualization'. Primary 
> issue is not timing per se, but KVM scheduling. Timing source, while present 
> from dahdi_dummy natively may still not get proper scheduling by KVM process. 
> This could also affect general call quality (even non IAX2 trunked voice), 
> DTMF, etc.
>
> I have to believe there are others running virtualized Asterisk installations 
> with some degree of success on OpenVZ or KVM. Care to share your thoughts?
>

Hi Tim,
I'm using OpenVZ, it works very well.
Take a look at: http://wiki.openvz.org/Asterisk_from_source

You will have to compile dahdi and install it on the HN, and then you
compile and install asterisk in the containers.
I've not done this with chan_iax, but with meetme. In the case with
meetme you would have to move some files over to trick asterisk that
dahdi is compiled on the machine. The wiki mentions copying user.h.

I used this as a starting point, some years ago:
http://www.telephreak.org/papers/vpa/
This paper covers vserver, so it's not exactly the same - but the steps
with tonezone was the same when I built the current server running this
configuration.

I'm in the process of building another server with openvz, so I'll need
to refresh my memory and try to document the procedure.

-- 
Med vänlig hälsning

Johan Wilfer email: jo...@jttech.se
JT Tech | Utvecklare webb: http://jttech.se
direkt: +46 31 380 91 01  support: +46 31 380 91 00

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Re: [asterisk-users] State of Asterisk+Virtualization+Timing

[Nic Colledge]

Have you thought about using LXC rather than OpenVZ.

There are a few references to allowing guest access to timing hardware online.

I've only been playing with it recently and haven't used it in production yet 
but plan to soon.

As for general thoughts about virtualising asterisk, I tried it in the past 
(about a year ago) on KVM and VMWare and it didn't work too well for me. 
Regardless of whether you are using LXC / OpenVZ / KVM / Whatever, you should 
be careful not to have too much other stuff running on the box. If asterisk has 
to wait to get CPU time you will really notice it, this isn't a problem with 
other applications like say a webserver. 

Nic.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson
Sent: 01 November 2011 17:08
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] State of Asterisk+Virtualization+Timing

Greetings-

I'm about to dive into the process of virtualizing some of my Asterisk 
(primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, 
the primary issue preventing me from moving was the lack of proper timing. We 
do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either 
OpenVZ or KVM, but each seem to have independent "issues" that need to be 
addressed:

OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant 
access to host node timing source (physical device, or dahdi_dummy in 
/dev/dahdi/) to the containerized Asterisk process.

KVM - Higher overhead, easier installation, 'true virtualization'. Primary 
issue is not timing per se, but KVM scheduling. Timing source, while present 
from dahdi_dummy natively may still not get proper scheduling by KVM process. 
This could also affect general call quality (even non IAX2 trunked voice), 
DTMF, etc.

I have to believe there are others running virtualized Asterisk installations 
with some degree of success on OpenVZ or KVM. Care to share your thoughts?

--Tim

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Re: [asterisk-users] custom automated meeting

[Thanasis]

on 11/01/2011 05:41 PM Yaroslav Panych wrote the following:
> You need simple dialplan of four steps:
> same =>n,Set(conf_name=conf-${RAND(1,1000)})
> same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
> same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
> same =>n,MeetMe(${conf_name},dFI1xAC)
> same =>n,Noop(do post conference stuff)
> 

Thanks!
What is the meaning of the options dFI1xAC passed to
app,MeetMe,${conf_name} ?
Where can I find them described please?

> 
> 2011/10/31 Thanasis :
>> I need your help in implementing the following scenario:
>>
>> A certain extension will ring two sip phones simultaneously and when one
>> of them answers, the other keeps ringing until it answers too, and then
>> all three (the caller and the other two) are immediately placed in a
>> conference room (same room for all three).
>>
>> Can we do it?
>>

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[asterisk-users] State of Asterisk+Virtualization+Timing

[Tim Nelson]

Greetings-

I'm about to dive into the process of virtualizing some of my Asterisk 
(primarily 1.4.x) infrastructure. In the past, when looking at virt solutions, 
the primary issue preventing me from moving was the lack of proper timing. We 
do not need it for MeetMe but rather for IAX2 trunking. I'd like to use either 
OpenVZ or KVM, but each seem to have independent "issues" that need to be 
addressed:

OpenVZ - Better resource usage, lower overhead. Primary issue is how to grant 
access to host node timing source (physical device, or dahdi_dummy in 
/dev/dahdi/) to the containerized Asterisk process.

KVM - Higher overhead, easier installation, 'true virtualization'. Primary 
issue is not timing per se, but KVM scheduling. Timing source, while present 
from dahdi_dummy natively may still not get proper scheduling by KVM process. 
This could also affect general call quality (even non IAX2 trunked voice), 
DTMF, etc.

I have to believe there are others running virtualized Asterisk installations 
with some degree of success on OpenVZ or KVM. Care to share your thoughts?

--Tim

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Re: [asterisk-users] custom automated meeting

[Yaroslav Panych]

You need simple dialplan of four steps:
same =>n,Set(conf_name=conf-${RAND(1,1000)})
same =>n,Originate(SIP/dev1,app,MeetMe,${conf_name},dFI1x)
same =>n,Originate(SIP/dev2,app,MeetMe,${conf_name},dFI1x)
same =>n,MeetMe(${conf_name},dFI1xAC)
same =>n,Noop(do post conference stuff)


2011/10/31 Thanasis :
> I need your help in implementing the following scenario:
>
> A certain extension will ring two sip phones simultaneously and when one
> of them answers, the other keeps ringing until it answers too, and then
> all three (the caller and the other two) are immediately placed in a
> conference room (same room for all three).
>
> Can we do it?
>
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> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>               http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>   http://lists.digium.com/mailman/listinfo/asterisk-users
>

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Re: [asterisk-users] custom automated meeting

[Danny Nicholas]

Although if you dig through the archives you can find a good cross-section
of AGI samples.  Check the Asterisk Cookbook wikis as well.

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Sammy Govind
Sent: Tuesday, November 01, 2011 9:08 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] custom automated meeting

 

There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin
directory by developing it yourself.

On Tue, Nov 1, 2011 at 6:57 PM, Thanasis  wrote:

on 11/01/2011 03:25 PM Danny Nicholas wrote the following:

> One way to do this (there are probably more and better ways).  Incoming
call
> to 123456789 launches meetme(1234,b(connecta.agi))
> Connecta.agi calls lines B and C and connects them to meetme(1234).

Thanks, but could you be more elaborate please?
Where can I find connecta.agi ?


>
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis
> Sent: Tuesday, November 01, 2011 1:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] custom automated meeting
>
> I just want to make two specific sip phone sets to ring together, when
> someone dials a specific incoming extension. And then, when each of the
> ringed sets answers, to be placed immediately into meeting session with
the
> caller together with the other phone set.
>
> Here is exactly what I mean:
>
> Person A dials 123456789. Asterisk routes the incoming call and rings sip
> phones B and C. Person B answers phone B and starts talking with person A,
> while phone C keeps ringing. A minute later, and while A and B are still
> talking together, person C answers phone C, and starts talking with A and
B
> together (that is aromatically all being placed in the same conference
> session).
>
> Is that doable?

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Re: [asterisk-users] custom automated meeting

[Sammy Govind]

There is only one way to find out connecta.agi in /var/lib/asterisk/agi-bin
directory by developing it yourself.

On Tue, Nov 1, 2011 at 6:57 PM, Thanasis  wrote:

> on 11/01/2011 03:25 PM Danny Nicholas wrote the following:
> > One way to do this (there are probably more and better ways).  Incoming
> call
> > to 123456789 launches meetme(1234,b(connecta.agi))
> > Connecta.agi calls lines B and C and connects them to meetme(1234).
>
> Thanks, but could you be more elaborate please?
> Where can I find connecta.agi ?
>
> >
> > -Original Message-
> > From: asterisk-users-boun...@lists.digium.com
> > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis
> > Sent: Tuesday, November 01, 2011 1:58 AM
> > To: Asterisk Users Mailing List - Non-Commercial Discussion
> > Subject: Re: [asterisk-users] custom automated meeting
> >
> > I just want to make two specific sip phone sets to ring together, when
> > someone dials a specific incoming extension. And then, when each of the
> > ringed sets answers, to be placed immediately into meeting session with
> the
> > caller together with the other phone set.
> >
> > Here is exactly what I mean:
> >
> > Person A dials 123456789. Asterisk routes the incoming call and rings sip
> > phones B and C. Person B answers phone B and starts talking with person
> A,
> > while phone C keeps ringing. A minute later, and while A and B are still
> > talking together, person C answers phone C, and starts talking with A
> and B
> > together (that is aromatically all being placed in the same conference
> > session).
> >
> > Is that doable?
>
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> _
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Re: [asterisk-users] custom automated meeting

[Thanasis]

on 11/01/2011 03:25 PM Danny Nicholas wrote the following:
> One way to do this (there are probably more and better ways).  Incoming call
> to 123456789 launches meetme(1234,b(connecta.agi))
> Connecta.agi calls lines B and C and connects them to meetme(1234).

Thanks, but could you be more elaborate please?
Where can I find connecta.agi ?

> 
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis
> Sent: Tuesday, November 01, 2011 1:58 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] custom automated meeting
> 
> I just want to make two specific sip phone sets to ring together, when
> someone dials a specific incoming extension. And then, when each of the
> ringed sets answers, to be placed immediately into meeting session with the
> caller together with the other phone set.
> 
> Here is exactly what I mean:
> 
> Person A dials 123456789. Asterisk routes the incoming call and rings sip
> phones B and C. Person B answers phone B and starts talking with person A,
> while phone C keeps ringing. A minute later, and while A and B are still
> talking together, person C answers phone C, and starts talking with A and B
> together (that is aromatically all being placed in the same conference
> session).
> 
> Is that doable?

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Re: [asterisk-users] Issues with Xorcom astribanks when running Ubuntu 11.10 (oneiric) as a quemu/libvirt guest

[Eric van der Vlist]

Tzafrir,

I am in front of the server.

Le dimanche 30 octobre 2011 à 22:13 +0100, Eric van der Vlist a écrit :
> Tzafrir,
> 
> Le dimanche 30 octobre 2011 à 10:30 +0200, Tzafrir Cohen a écrit :

> > The problem is elsewhere. What happens if
> > you manually run:
> > 
> >   /usr/share/dahdi/xpp_fxloader load #?

vdv@asterisk-rg:~$ lsusb
Bus 001 Device 001: ID 1d6b:0002 Linux Foundation 2.0 root hub
Bus 001 Device 003: ID e4e4:1161 Xorcom Ltd. Astribank 2 series
vdv@asterisk-rg:~$ sudo /usr/share/dahdi/xpp_fxloader load
[sudo] password for vdv: 
'xpp_fxloader'[850]: - FIRMWARE LOADING: (load) [1 devices]
Got all 1 devices
INFO: usb:001/003: ID=E4E4:1161 [Xorcom LTD / Astribank / X1047833]
INFO: Loading hexfile to FPGA: /usr/share/dahdi/FPGA_1161.hex (version
7276)
mpp_funcs.c:308: ERROR(recv_command): Receive from usb failed.
mpp_funcs.c:359: ERROR(process_command): recv_command failed
mpp_funcs.c:710: ERROR(mpp_send_start): process_command failed: -71
astribank_hexload.c:99: ERROR(load_hexfile): Failed hexfile send start:
-71
astribank_hexload.c:218: ERROR(main): Loading firmware to FPGA failed
'xpp_fxloader'[898]: /usr/sbin/astribank_hexload failed with status 1

> > Or:
> > 
> >   astribank_tool -D 001/005 -Q
> 
> I'll test that as soon as I can!
> 
> > If you have dahdi-tools < 2.5, you'll need:
> > 
> >   astribank_tool -D /dev/bun/usb/001/005 -Q

vdv@asterisk-rg:~$ sudo astribank_tool -D /dev/bus/usb/001/003 -Q
mpp_funcs.c:308: ERROR(recv_command): Receive from usb failed.
mpp_funcs.c:359: ERROR(process_command): recv_command failed
mpp_funcs.c:454: ERROR(mpp_proto_query): process_command failed: -71
mpp_funcs.c:1001: ERROR(mpp_init): Protocol handshake failed: -71
astribank_tool.c:209: ERROR(main): Failed initializing MPP

> > What version of dahdi-tools is it?
> > 
> 2.4.1, and I see that dahdi-firmware-nonfree (that includes your
> firmware) is 2.2.1.1-1:
> 
> vdv@lrt-rg:~$ dpkg -l "*dahdi*"
> Souhait=inconnU/Installé/suppRimé/Purgé/H=à garder
> |
> État=Non/Installé/fichier-Config/dépaqUeté/échec-conFig/H=semi-installé/W=attend-traitement-déclenchements
> |/ Err?=(aucune)/besoin Réinstallation (État,Err: majuscule=mauvais)
> ||/ Nom  Version  Description
> +++---
> ii  asterisk-dahdi   1:1.8.4.4~dfsg-2ubuntu1  DAHDI devices
> support for the Asterisk PBX
> ii  dahdi1:2.4.1-1ubuntu1 utilities for
> using the DAHDI kernel modules
> ii  dahdi-dkms   1:2.4.1+dfsg-1ubuntu2DAHDI telephony
> interface (dkms kernel driver)
> ii  dahdi-firmware-nonfree   2.2.1.1-1DAHDI non-free
> firmware
> ii  dahdi-linux  1:2.4.1+dfsg-1ubuntu2DAHDI telephony
> interface - Linux userspace parts
> un  dahdi-source(aucune
> description n'est disponible)
> 
> That being said, the host (in which the firmware loads fine) has exactly
> the same versions installed.
> 
Thanks for your help,

Eric


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Re: [asterisk-users] custom automated meeting

[Danny Nicholas]

One way to do this (there are probably more and better ways).  Incoming call
to 123456789 launches meetme(1234,b(connecta.agi))
Connecta.agi calls lines B and C and connects them to meetme(1234).

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Thanasis
Sent: Tuesday, November 01, 2011 1:58 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] custom automated meeting

I just want to make two specific sip phone sets to ring together, when
someone dials a specific incoming extension. And then, when each of the
ringed sets answers, to be placed immediately into meeting session with the
caller together with the other phone set.

Here is exactly what I mean:

Person A dials 123456789. Asterisk routes the incoming call and rings sip
phones B and C. Person B answers phone B and starts talking with person A,
while phone C keeps ringing. A minute later, and while A and B are still
talking together, person C answers phone C, and starts talking with A and B
together (that is aromatically all being placed in the same conference
session).

Is that doable?

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Re: [asterisk-users] Nat Phone in Asterisk 10

[Danny Nicholas]

IP outputs

ip 1

Object "1" is unknown, try "ip help".

ip -4 a

1: lo:  mtu 16436 qdisc noqueue state UNKNOWN

inet 127.0.0.1/8 brd 127.255.255.255 scope host lo

inet 127.0.0.2/8 brd 127.255.255.255 scope host secondary lo

2: eth0:  mtu 1500 qdisc pfifo_fast state
UNKNOWN qlen 1000

inet 192.168.23.97/24 brd 192.168.23.255 scope global eth0

ip ro

192.168.23.0/24 dev eth0  proto kernel  scope link  src 192.168.23.97

169.254.0.0/16 dev eth0  scope link

127.0.0.0/8 dev lo  scope link

default via 192.168.23.150 dev eth0

 

phone that doesn't connect has static IP of 192.168.33.90

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Anton
Kvashenkin
Sent: Tuesday, November 01, 2011 4:12 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Nat Phone in Asterisk 10

 

Not info about networl settings. Please give output of

ip l

ip -4 a

ip ro

2011/11/1 Danny Nicholas 

Hello listers,

Another opportunity presents itself in my 1.4 to
10.0 conversion.   My asterisk is set up for 192.168.23.xx and most of my
phones are 192.168.23.yy peers.  I work on two subnets so I have one phone
defined as 192.168.33.xx.  This phone comes up and registers and accepts
calls and calls out in 1.4.41 but shows unreachable in 10.0.  What changed
that is killing my "off-network" phone?

 

Thanks in Advance

Danny Nicholas

 


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Re: [asterisk-users] Problem with Atxfer for the calling party

[Antonio Modesto]

Good morning,

I have not solved this problem yet, but, I found that the source of
the problem are my macros. For example, I have this context:

context ramais {
101 => &dial_sip(exten1);
102 => &dial_sip(exten2);
103 => &dial_sip(exten3);
};

All these extensions use the dial_sip macro, I have changed this context
to use the Dial application instead of dial_sip macro, it worked fine.
The problem is that when i use the macro, the current context is changed
to the dial_sip context, the dial_sip context is automatically created
by asterisk when i use any macro and of fact this context doesn't have
the ramais context included. Is there some way to specify on which
context the macro will run?

On Mon, 2011-10-31 at 09:09 -0200, Antonio Modesto wrote:

> Good Morning,
> 
>   I have an asterisk18-1.8.7.1 running on a FreeBSD 8.2-STABLE, and it
> is working well so far, i'm just having some problems with atxfer.
> 
> I have written this macro to dial sip extensions:
> 
> macro dial_sip(exten) {
> Verbose(2,"==> Chamando a MACRO dial_sip - ponto 1 macros.ael
> <==");
> Verbose(4,"> Macro dial_sip iniciada.");
> ChanIsAvail(SIP/${exten});
> Verbose(2,"==> ${AVAILORIGCHAN}");
> 
> if ("${AVAILORIGCHAN}" != "")
> {
> Verbose(4,"> SIP/${exten} parece estar disponivel,
> vou disca-lo agora.");
> Set(FromExt=${CALLERID(num)});
> System(/bin/sh /var/spool/asterisk/calllog/log.sh
> SIP/${FromExt} SIP/${exten} SIP-TO-SIP);
> Verbose(4,"> System status: ${SYSTEMSTATUS}");
> Dial(SIP/${exten},${SIP_DIAL_TIMEOUT},Ttr);
> Hangup();
> }
> else
> {
> Verbose(2,"> SIP/${exten} nao esta disponivel.");
> Hangup();
> };
> 
> 
> NoOp("From ${MACRO_EXTEN} to ${exten});
> System(${CALLLOGDIR}/log.sh ${exten});
> 
> return;
> };
> 
> It is working, but the calling party is not able to transfer the calls
> because asterisk doesn't wait all the digits be typed, it tries to
> transfer the call when the first digit is pressed (We use 3 digits
> extensions):
> 
> [Oct 31 09:04:01] WARNING[2926]: features.c:2315 builtin_atxfer:
> Extension '1' does not exist in context 'dial_sip'
>   == Spawn extension (dial_sip, ~~s~~, 11) exited non-zero on
> 'SIP/modesto-000d'
> [Oct 31 09:04:03] WARNING[2926]: features.c:2319 builtin_atxfer: No
> digits dialed for atxfer.
> 
> Does anyone have suggestions?
> 
> Regards. 
> 
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Re: [asterisk-users] bug in queuemanager?

[Henry Dogger]

Sorry it took me a while, but I was ill for a few days J

 

Part1: http://pastebin.com/SZqgxh7B

Part2: http://pastebin.com/gfJtVVRE

 

In this log a call from extension 346 is made to queue 900.

Queue 900 has 1 agent namely agent 300 which is logged on at extension
204.

Queue 901 has 1 agent namely agent 301 which is logged on at extension
203.

Agent 300 answers call from 346 and transfers this call to queue 901.

After agent 301 has answered this forwarded call (caller 346) a new call
from 346 arrives at queue 901.

After agent 301 hangs up the call, the new call from 346 is presented
immediately without any wrap-up time.

Hope this logging helps...

 

Greetings,

Henry

 

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Warren
Selby
Sent: dinsdag 25 oktober 2011 18:07
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] bug in queuemanager?

 

On Tue, Oct 25, 2011 at 7:25 AM, Henry Dogger 
wrote:

Customer 200 calls to queue 900, Agent 300 answers but tells Customer
200 that he should be at Queue 901 and transfers Customer 200 (using *2)
to Queue 901. Agent 301 now gets the call from Queue 901 with Customer
200, answers the calls etc. After disconnect a new call arrivers
immediately from Queue 901, without any wrap-up time. This should be
considered as a bug IMO.

Any ideas on how to fix, workaround this problem?

 

 


Please share the CLI output of such a situation, with the verbosity and
debugging both set to 10 ('core set verbose 10' and 'core set debug 10'
from the asterisk CLI), it may shed some light on whether this is a bug
or a "feature".  


-- 
Thanks,
--Warren Selby, dCAP
http://www.SelbyTech.com

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Re: [asterisk-users] Nat Phone in Asterisk 10

[Anton Kvashenkin]

Not info about networl settings. Please give output of
ip l
ip -4 a
ip ro

2011/11/1 Danny Nicholas 

> Hello listers,
>
> Another opportunity presents itself in my 1.4 to
> 10.0 conversion.   My asterisk is set up for 192.168.23.xx and most of my
> phones are 192.168.23.yy peers.  I work on two subnets so I have one phone
> defined as 192.168.33.xx.  This phone comes up and registers and accepts
> calls and calls out in 1.4.41 but shows unreachable in 10.0.  What changed
> that is killing my “off-network” phone?
>
> ** **
>
> Thanks in Advance
>
> Danny Nicholas
>
> ** **
>
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>
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