Carlos Rojas | 11 Feb 03:18
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Virtual Server

Hello everybody

someone in this list, has installed asterisk, in a virtual server like  proxmox? I'm thinking  install some asterisk servers in a machine dell xeon 64 processor, but I'm not sure, about virtual Server software.

I heard, about proxmox, but I don't know if works fine.

Regards

Carlos
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Mike | 10 Feb 23:30
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Polycom firmware 4.0.1 and paging

Hi,

 

I just moved many Polycom phones from firmware v3 to 4.0.1b.  Anto-Answer simply stopped functioning. I can downgrade and make it work, upgrading kills it again. There obviously is a difference in how the newer firmware is treating this auto answer sip header.

 

Can anybody tell me if they have Polycom firmware 4.x.x working with auto-answer/paging? Just so I know it’s worth my time to investigate, as opposed to knowing it`s a Polycom firmware bug? If so, did you have to make any changes to the SIP header sent to make Polycom phones auto answer?

 

Regards,

 

Mike

 

 

 

 

 

 

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Phil Frost | 10 Feb 20:37
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Call queuing behavior

I'm trying to implement a very simple call queue for a small, low volume helpdesk. We have 2-5 agents, and
rarely does the queue get more than 1 or 2 callers deep. I'm using the ringall strategy and I want calls
answered in FIFO order.

Say caller A calls the queue, and there is one member logged in. Asterisk rings the member.

Now, caller B calls. Asterisk rings the member. Now the member's handset is showing two incoming calls.

This particular member is a bit lazy or busy, so he waits 30 seconds, and the first call times out. Asterisk
says, "Nobody picked up in 30000 ms", the caller hears the periodic announcement, and Asterisk stops
ringing the member.

Now, the member is unbusy, so he answers a call. But, he's connected to caller B, even though caller A called
first. That's not what I'd expect - I want callers to be answered in FIFO order.

I suspect there's some interaction with the "ringinuse" and "timeout" settings here. I had thought, maybe
I'll make the timeout very long. Since I'm using ringall, I don't have to worry about a lazy/dead member not
answering and thus preventing the caller from being presented to the next member. However, if I do this, I
can't seem to make it longer than 60 seconds, and also the caller seems to only be presented with
announcements when the timeout expires. I'd like to tell the caller every 30 seconds that they can press 0
to leave a voicemail, regardless of any other queue activity.

ringinuse=no might be nice, also so if there are more than three callers in the queue I don't eat up all the
call appearance buttons on my member's handsets. However, I read that only SIP channels can report "in
use", and my members are on OOH323 channels. So, that's out. Coincidentally, I could make my "members" be
just one, which is a hunt group implemented in another PBX. I'd then want Asterisk to present one caller to
this one member, and keep presenting that caller to the one member until it's answered, or the caller has
been waiting over five minutes, when he's sent to voicemail. Only then is the next caller presented. Even
though I'd think it would be easy for app_queue to know that the member is busy (after all, it's calling
them), there doesn't seem to be any way to direct app_que
 ue to not throw every caller in the queue at the one member.

Any ideas on how I might approach a better solution?

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bakko | 10 Feb 18:32
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Call Completion

Hello,

I'm trying the Call Completion system. All work fine.

I still don't undesrtand how Asterisk work.

Does Asterisk use sip signaling or other protocol to send notifications?

On the Asterisk Wiki seems that the system is based on 
draft-ietf-bliss-call-completion-04 but this draft talk about SUBSCRIBE and 
NOTIFY.

I see nothing in sip capture.

Can you help me with this question?

Thank's

Regards

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James Wystead | 10 Feb 17:34
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Question for the group

Hello Folks;

I know this is a non-commercial discussion group, but I am looking for some open-source software suggestions


We are going to be setting up a prepaid PBX service with the following features:


  • Email to Fax and  Fax to Email
  • Inward DID local and 800 services
  • Calling card SIP based and ANI authenticated

I see there are many types of software that can be addons/installs/etc to Asterisk. 

So, the question that I ask is which one would be best suited for these needs? Of course, it needs to be scalable and work well (most opensource software does)

So, any thoughts? 

Thanks

G
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Vieri | 10 Feb 14:40
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distributed queue information over several Asterisk nodes

Is it possible to distribute QUEUE information among several Asterisk nodes in a "multimaster" or "load
balancing" setup?

I haven't tried this yet but if one uses realtime with a clustered multimaster database and the queue
agents/members are fixed SIP channels (eg. SIP/100) then I guess that the Queue app will be able to contact
the member no matter to which Asterisk node it registered.
However, what happens if incoming calls enter more than one queue (a queue on any Asterisk node, as it would
be expected in a fully load-balanced setup)?
Let's say QUEUE1 on ASTNODE1 has 1 incoming call waiting to be picked up and a second call comes in but enters
QUEUE1 on ASTNODE2 which was previously empty.
So for example, how can the caller in QUEUE1 on ASTNODE2 be placed in position 2 instead of 1?

In other words, can the same QUEUE work/collaborate over different Asterisk nodes?

Thanks,

Vieri

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Matteo Fortini | 10 Feb 12:30
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DTMF forwarding and Page

Hi,
I'd like to implement some way of controlling remote SIP clients while 
in a call, to execute remote commands.

The call topology (think of a PA system) is this:
* the caller is in a MeetMe() conference room
* the callees are Page()d, then the dynamic conference room is connected 
to the previous one

I'm wondering if Asterisk is relaying DTMF (SIP info or RTP) from the 
caller to the callees. I found option 'F' for MeetMe, but I have no 
control on Page().

TIA,
Matteo

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ing.Achim Alexandru | 10 Feb 12:13
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dial plan with hangup cause 34

Dear Asterisk Users,

I have a question. I use asterisk 1.6 withh freepbx on ubuntu ,
compiled manually.
I want to change the route congestion message ( all-circuit-bussy....)
wiyh a hangup cause 34 ( something like that in dialplan
s,n,GotoIf($[${HANGUPCAUSE} = 34]?failover,1). Have any ideas?

Thanks
Alexandru Achim

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Asterisk 10.1.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 10.1.2. This
release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 10.1.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* --- Fix SIP INFO DTMF handling for non-numeric codes ---
  (Closes issue ASTERISK-19290. Reported by: Ira Emus)

* --- Fix crash in ParkAndAnnounce ---
  (Closes issue ASTERISK-19311. Reported-by: tootai)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-10.1.2

Thank you for your continued support of Asterisk!

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Asterisk 1.8.9.2 Now Available

The Asterisk Development Team has announced the release of Asterisk 1.8.9.2.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/

The release of Asterisk 1.8.9.2 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!

The following are the issues resolved in this release:

* --- Fix SIP INFO DTMF handling for non-numeric codes ---
  (Closes issue ASTERISK-19290. Reported by: Ira Emus)

* --- Fix crash in ParkAndAnnounce ---
  (Closes issue ASTERISK-19311. Reported-by: tootai)

For a full list of changes in this release, please see the ChangeLog:

http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.2

Thank you for your continued support of Asterisk!

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Mike Diehl | 9 Feb 22:11

Turning off splash ring on PAP2T

Hi all,

I'd like to know how I can turn off the "splash ring" voicemail waiting 
indication on a PAP2T from the provisioning XML file.  I can do it from the web 
interface, but I need to do it on "a lot" of machines....

TIA,

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Take care and have fun,
Mike Diehl.

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Gmane