[asterisk-users] [NEWBIE] Right dect to buy to use with asterisk

2013-12-11 Thread Mario Giammarco
Hello,
I need to setup this configuration:

- asterisk as IVR;
- dect phones.

So basically I need a standard set of features:

- each dect phone has its extension so I can call it directly;
- handover of a call with R key;
- if a call is not replied by someone ring all phones.

I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip
base station.

Which one should I buy?

Thanks,
Mario


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Re: [asterisk-users] [NEWBIE] Right dect to buy to use with asterisk

2013-12-11 Thread Leandro Dardini
Hello Mario,
nice to meet you on this mailing list!
Gigaset phones are a very high quality/price ratio, so I'll suggest you to
go with the dect ip models. Then you'll need to configure asterisk to act
as IVR, configure a queue and a failover to ring all hunt list.

Drop me a phone call and I'll be happy to help you

Leandro


2013/12/11 Mario Giammarco mgiamma...@gmail.com

 Hello,
 I need to setup this configuration:

 - asterisk as IVR;
 - dect phones.

 So basically I need a standard set of features:

 - each dect phone has its extension so I can call it directly;
 - handover of a call with R key;
 - if a call is not replied by someone ring all phones.

 I have little budget. I can choose to buy a fritz!box or a gigasect dect/ip
 base station.

 Which one should I buy?

 Thanks,
 Mario


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[asterisk-users] Asterisk 12 with sipml5+ PJNATH issue

2013-12-11 Thread Prakash N
Hi ,

 We trying to work on Asterisk 12 with sipml5+ PJNATH ,We can able register
but calling is not working  .The same configuration test with Xlite soft
phone it working fine

With Regards

Prakash
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[asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread Olivier
Hello,

I always thought that Energy Savings mode existed with ISDN Basic Rate
Interface in Point -to-multi-Point but it didn't with Point-to-point.

Is this correct ?
Have you ever met a public PSTN switch configured to cut B ISDN channels
in Point-to-point.signalling ?

Regards.
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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread jg
Yes, I can confirm that. P2P always requires a PBX and you can easily build up trunks of P2P 
connections. So all channels always need to work in a coordinated way, if you allow this sloppy 
formulation. Another reason for not allowing the energy saving mode is probably that in some 
countries part of the phone number resolution has to happen inside the pbx on the customer side. 
This could be a problem if it takes too much time to wake up a device, or if the state of the 
connection cannot be determined beforehand.


P2MP does have an energy saving mode which depends on the PSTN and on the TE device. Sometimes 
phone companies allow special sequences (like *121#) that explicitly allow to disable the energy 
saving mode, if the P2MP connection is connected up to a PBX. Some customer devices have 
configuration options like connect to a pbx which send exactly this sequence to the next 
telephone switch.


All the technicals docs are available online, but I have yet to find a person, myself included, 
who actually read them all. All of this applies to Western Europe, only.


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Re: [asterisk-users] Asterisk on Windows

2013-12-11 Thread Ryan Wagoner
On Wed, Dec 4, 2013 at 10:19 AM, CDR vene...@gmail.com wrote:

 Digium is 100% lost in the map. If they would come up with a Paid
 version of Asterisk, one that would use the .NET framework in Windows,
 something simple to install, they could go public on the product.
 Linux has a very steep learning curve. A Windows application that
 would do exactly the same would be a home run. Note: I am a Linux
 expert user, but it took me years to get here. And still, moving from
 regular RHEL 6.0 to Fedora 20 (RHEL 7) is a pain in the neck. The .NET
 framework and Windows server 2012 are miles away in terms of
 friendliness and on equal footing on performance. I don´t mean another
 slow cygwin port, I man a native Asterisk for windows. In fact, I
 would invest on the project if somebody wants to do it.


Windows and Linux should be able to coexist. I have had great success
setting up a VMware ESXi server with Windows VMs for AD and Exchange and
Linux VMs for Asterisk and Web / FTP. Asterisk with Exchange UM for
voicemail is a winning combination and works seamlessly. It is essentially
a private cloud of the customer. Why not use the OS that works for the task
at hand?

Ryan
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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread Olivier
2013/12/11 jg webaccou...@jgoettgens.de

 Yes, I can confirm that. P2P always requires a PBX and you can easily
 build up trunks of P2P connections. So all channels always need to work in
 a coordinated way, if you allow this sloppy formulation. Another reason for
 not allowing the energy saving mode is probably that in some countries part
 of the phone number resolution has to happen inside the pbx on the customer
 side. This could be a problem if it takes too much time to wake up a
 device, or if the state of the connection cannot be determined beforehand.


So if my understanding is correct, Energy Saving mode exists in P2P BRI
lines.
How do you monitor these lines ?
By monitoring, I mean getting an SNMP trap when the line is going down,
for instance.
Maybe I should just convince myself that I should not check anymore line
status looking a flag updated in real time, but simply trying to send an
outgoing call as I would with analog lines.


 P2MP does have an energy saving mode which depends on the PSTN and on the
 TE device. Sometimes phone companies allow special sequences (like *121#)
 that explicitly allow to disable the energy saving mode, if the P2MP
 connection is connected up to a PBX. Some customer devices have
 configuration options like connect to a pbx which send exactly this
 sequence to the next telephone switch.


Have you also heard about this special sequence with P2P lines or is it
something especially for P2MP ?




 All the technicals docs are available online, but I have yet to find a
 person, myself included, who actually read them all.


Yes that's true.


 All of this applies to Western Europe, only.

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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread jg




So if my understanding is correct, Energy Saving mode exists in P2P BRI lines.

No, not for P2P lines, only for P2MP lines.


How do you monitor these lines ?

Most of the time I use a vendor specific utility, but pri show spans gets the 
status as well.



Have you also heard about this special sequence with P2P lines or is it something especially 
for P2MP ?
The sequence I gave is for P2MP only, but I don't know which telcos are using it (except for 
mine). P2P never sleeps as said before.


I haven't checked what happens when a P2P connection is not connected to anything at all. 
Theoretically it could go into the energy saving mode, but once it is connected, then it doesn't.


jg


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[asterisk-users] invalid From/Contact header values

2013-12-11 Thread Muhammad Faheem
Hi,
I'm observing wrong From/Contact header values. When I try to set
CallerID(num) it has no effect in the From and Contact Headers, and these
values are the same as the dialed number.
SIP Peers are defined using asterisk realtime. If I define the SIP Peers
using sip.conf then From/Contact header value are correct.

extentions.conf
[test]
exten= 1000, 1,NoOp()
same= n,Set(CALLERID(num)=)
same= n,Set(CALLERID(name)=)
same= n,Dial(SIP/1000)

exten= 2000, 1,NoOp()
same= n,Set(CALLERID(num)=)
same= n,Set(CALLERID(name)=)
same= n,Dial(SIP/2000)


Here is the sip trace...
--- Executing [2000@test:1] NoOp(SIP/1000-0014, ) in
new stack
-- Executing [2000@test:2] Set(SIP/1000-0014,
CALLERID(num)=) in new stack
-- Executing [2000@test:3] Set(SIP/1000-0014,
CALLERID(name)=) in new stack
-- Executing [2000@test:4] Dial(SIP/1000-0014, SIP/2000) in new
stack
  == Using SIP RTP CoS mark 5
Audio is at 16264
Adding codec 14 (alaw) to SDP
Adding codec 13 (ulaw) to SDP
Adding codec 12 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.10.7.218:5060:
INVITE sip:2000@10.10.7.218:5060 SIP/2.0
Via: SIP/2.0/UDP my-ip:5060;branch=z9hG4bK73e9c721
Max-Forwards: 70
From:  sip:2...@sipdev.mydomain.com;tag=as2a72da29
To: sip:2000@10.10.7.218:5060
Contact: sip:2000@my-ip:5060
Call-ID: 1f75fe937c6194227e6b5a5c29f41...@sipdev.mydomain.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.5.1
Date: Wed, 11 Dec 2013 16:23:07 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO,
PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 309

v=0
o=root 604923607 604923607 IN IP4 my-ip
s=Asterisk PBX 11.5.1
c=IN IP4 my-ip
t=0 0
m=audio 16264 RTP/AVP 8 0 3 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
uname -a
Linux 6g-asterisk-devel 2.6.32-279.el6.x86_64 #1 SMP Fri Jun 22 12:19:21
UTC 2012 x86_64 x86_64 x86_64 GNU/Linux

asterisk -rx core show version
Asterisk 11.5.1 built by root @ 6g-asterisk-devel on a x86_64 running Linux
on 2013-10-07 10:50:45 UTC

Please suggest me, either I put the issue in issue tracker or there is some
workaround.

Thank you!
Muhammad Faheem
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Re: [asterisk-users] Queue with linear strategy does not work

2013-12-11 Thread Rusty Newton
On Tue, Dec 10, 2013 at 10:14 PM, Thorben Jensen i...@thorben.dk wrote:
 I have a queue with linear strategy. When I add dynamic members it does NOT
 ring the members in the order they are added.

 I use the command AddQueueMember to add members but it seems to be random
 how it rings the members.

Does it ring them in the same order for every call(even if it is not
the expected order)?

Does the order change up for each call even when no new members have been added?

Can you provide a pastebin of a verbose log (see logger.conf)
demonstrating the problem?

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Check us out at: http://digium.com  http://asterisk.org

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Re: [asterisk-users] dahdi show channels no such command

2013-12-11 Thread Joseph Towery






 From: Tzafrir Cohen tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com 
Sent: Tuesday, December 10, 2013 4:06 PM
Subject: Re: [asterisk-users] dahdi show channels no such command
 

On Thu, Dec 05, 2013 at 07:19:56PM -0800, Joseph Towery wrote:
 Here are the commands
 
 asteriskpbx@pbxmain:/etc/dahdi$ dahdi_cfg -vvv
 DAHDI Tools Version - 2.7.0.2
 
 DAHDI Version: 2.7.0.1
 Echo Canceller(s): MG2
 Configuration
 ==
 
 
 Channel map:
 
 Channel 01: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 01)
 Channel 02: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 02)
 Channel 03: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 03)
 Channel 04: FXS Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 04)
 Channel 05: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 05)
 Channel 06: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 06)
 Channel 07: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 07)
 Channel 08: FXO Kewlstart (Default) (Echo Canceler: mg2) (Slaves: 08)
 
 8 channels to configure.
 
 Setting echocan for channel 1 to mg2
 Setting echocan for channel 2 to mg2
 Setting echocan for channel 3 to mg2
 Setting echocan for channel 4 to mg2
 Setting echocan for channel 5 to mg2
 Setting echocan for channel 6 to mg2
 Setting echocan for channel 7 to mg2
 Setting echocan for channel 8 to mg2
 asteriskpbx@pbxmain:/etc/dahdi$
 
 
 
 pbxmain*CLI module show like chan_dahdi.so
 Module                         Description                              Use 
 Count
 chan_dahdi.so                  DAHDI Telephony Driver w/PRI             0    
    
 1 modules loaded

What's the output of lsdahdi? Note channels there that are open (by
Asterisk).

Try running in the asterisk CLI:

  module unload chan_dahdi.so
  module load chan_dahdi.so

The 'load' command will probably fail and give you the real error.
Though you should probably already have it in the logs.

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Thanks Tzafrir,


Here is the output of lsdahdi:
asteriskpbx@pbxmain:~$ sudo lsdahdi
### Span  1: WCTDM/0 Wildcard TDM800P (MASTER)
  1 FXO        FXSKS       (EC: MG2 - INACTIVE)
  2 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
  3 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
  4 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
  5 FXS        FXOKS       (EC: MG2 - INACTIVE)
  6 FXS        FXOKS       (EC: MG2 - INACTIVE)
  7 FXS        FXOKS       (EC: MG2 - INACTIVE)
  8 FXS        FXOKS       (EC: MG2 - INACTIVE)
asteriskpbx@pbxmain:~$


I figured it out.  I had to setup logger.conf to get console debugging to work 
even though I was starting asterisk -rv, doing the unload and load command 
I was able to figure out our my issue.  I had the dadhichan = variable set 
incorrectly for the FXO ports.  I appreciate the help.  


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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread Olivier
2013/12/11 jg webaccou...@jgoettgens.de



 So if my understanding is correct, Energy Saving mode exists in P2P BRI
 lines.

 No, not for P2P lines, only for P2MP lines.


At the moment, I'm having trouble with lines going up and down all day long.
I asked them to be created as P2P.
Mostly but not always calls are passing in and out correctly.

So, if I'm not mistaken, either :
- these lines are configured as P2MP (and not as P2P as required),
- I'm facing something else I've never met before.

As these lines are up most of the time, I'm having a hard time to get any
useful help by Telco beside level 1 basic checks.

For the record, these lines are terminated on my side with a Patton
smartnode (most of the time, I use Digium BRI boards).
During off hours, I'll try to check what happens when toggling to P2MP mode.




  How do you monitor these lines ?

 Most of the time I use a vendor specific utility, but pri show spans
 gets the status as well.



 Have you also heard about this special sequence with P2P lines or is it
 something especially for P2MP ?

 The sequence I gave is for P2MP only, but I don't know which telcos are
 using it (except for mine). P2P never sleeps as said before.

 I haven't checked what happens when a P2P connection is not connected to
 anything at all. Theoretically it could go into the energy saving mode, but
 once it is connected, then it doesn't.

 jg



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Re: [asterisk-users] Call Queue advise

2013-12-11 Thread Chad Wallace

On Mon, 9 Dec 2013 16:15:14 -0800
Bryan Anderson shadow...@gmail.com wrote:

 On Mon, Dec 9, 2013 at 4:11 PM, Chad Wallace
 cwall...@lodgingcompany.comwrote:
 
  On Mon, 9 Dec 2013 15:47:57 -0800
  Bryan Anderson shadow...@gmail.com wrote:
 
   I have a call queue that rings about 15 users and they are
   wanting to set it up so that the last person to answer a call
   doesn't ring on the next incoming call.
 
  Wouldn't the leastrecent strategy work for that?  It wouldn't
  absolutely forbid an agent from taking the next call, but it would
  make sure every other agent had priority.  You could also add a
  large wrap up time, to ensure they never get a second call within a
  certain time period.

 yes but I believe that least recent would ring one agent at a time?
 If my understanding is incorrect please correct it.  We are wanting
 to keep with multiple phones ring to ensure coverage.

Yes, you're right.  It seems your solution (setting a penalty after
they get a call) is probably the only one--unless, as has been
suggested, you rethink your requirements. You could also remove them
from the queue or pause them instead of changing their penalty.


-- 

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The Lodging Company
http://www.lodgingcompany.com/
OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] Asterisk on Windows

2013-12-11 Thread Tzafrir Cohen
On Tue, Dec 10, 2013 at 08:43:01PM -0500, Brian wrote:
 On Tue, 10 Dec 2013 23:02:45 +0200
 Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
 
  On Wed, Dec 04, 2013 at 02:12:41PM -0500, Ruddy Gbaguidi wrote:
   I never tought this is become a Linux vs Windows fight.
   We have been using asterisk on linux from a long time now and happy
   with it.
   But some of our customers who has windows in their environment want
   to use our call center software we developed on top of asterisk.
   So, the question was :
   Did anybody ever tried to isolate the asterisk SIP server/module and
   make it run under Windows ?
   Since, asterisk 12 is using pjsip (which is cross platform already),
   I tought it may be possible and wanted advices.
   
   I would love that every single customer switch to Linux and Ubuntu
   tomorrow morning but at the moment, that's not the case.
  
  There was an old half-working port of Asterisk to Cygwin which does
  run on Windows. It has not worked since at least 1.6.0 .
 
 That's just a unix-like interface which won't address the issues the OP
 has/had with running/configuring asterisk. IMHO it would probably be
 even more challenging. And IIRC the OP was looking for a non emulated
 solution anyway.

It's not emulated. It uses a compatibility layer library. If emulation
were such a major issue for you, I guess you'd never consider using a
language such as Java, Perl or Python, where the program runs in its own
virtual machine.

 
  And for others: the name is [MS-]Windows. Not 'wind-blows or whatever
  name you find for it. Please respect this list. If you don't have
  anything useful to add to the thread, please refrain from replying.
  
 
 I have to agree with the name calling part but the OP did imply that
 Windows was superior and that a Windows port would be profitable. You
 can't really expect to get away with that on a list devoted to an open
 source application without making a complete fool out of yourself.
 
 If it was a post regarding one of the many proprietary closed source
 applications/games without a native port to Linux/BSD/OS X then it
 would be a valid complaint. Having access to the source as well as
 liberal licensing terms which allow porting isn't a valid complaint and
 never will be.

I did not reply to that troll. I replied an OP who said that MS-Windows
was a requirement for his case.

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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread jg


At the moment, I'm having trouble with lines going up and down all day long.
I asked them to be created as P2P.
Mostly but not always calls are passing in and out correctly.
I doubt that the energy-saving is causing the trouble. How do you get the ISDN-signal? Is it an 
S0-bus? If yes, I may be able to be of further help. Even if the connection goes down, it should 
not matter for P2MP connections. But S0-bus problems can be nasty (though seen rarely these days).


So, if I'm not mistaken, either :
- these lines are configured as P2MP (and not as P2P as required),
- I'm facing something else I've never met before.

I would bet on the 2nd choice.


As these lines are up most of the time, I'm having a hard time to get any useful help by Telco 
beside level 1 basic checks.


For the record, these lines are terminated on my side with a Patton smartnode (most of the 
time, I use Digium BRI boards).

During off hours, I'll try to check what happens when toggling to P2MP mode.

I doubt that you do get anything useful if you pick the wrong mode.

Are you sure that your are not dealing with termination issues? Try sending a T.30 fax with fine 
resolution and ecm enabled when your connection is up. If it takes too long or fails, this is 
something to look at.


jg


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Re: [asterisk-users] dahdi show channels no such command

2013-12-11 Thread Tzafrir Cohen
On Wed, Dec 11, 2013 at 09:22:13AM -0800, Joseph Towery wrote:

 Here is the output of lsdahdi:
 asteriskpbx@pbxmain:~$ sudo lsdahdi
 ### Span  1: WCTDM/0 Wildcard TDM800P (MASTER)
   1 FXO        FXSKS       (EC: MG2 - INACTIVE)
   2 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
   3 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
   4 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
   5 FXS        FXOKS       (EC: MG2 - INACTIVE)
   6 FXS        FXOKS       (EC: MG2 - INACTIVE)
   7 FXS        FXOKS       (EC: MG2 - INACTIVE)
   8 FXS        FXOKS       (EC: MG2 - INACTIVE)

For the record: this output shows that the DAHDI channels have been
configured (they got the FXSKS signalling, which means it was applied by
running dahdi_cfg) but they are not '(in use)', which means that no
channel has been opened by Asterisk's chan_dahdi.

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[asterisk-users] Ctrl-W killing entire line, not just last word

2013-12-11 Thread Dotan Cohen
On the Asterisk CLI, I notice that pressing Ctrl-W deletes the entire
line of text, as opposed to just the last word. Is this an artifact of
using editline instead of readline? I'll note that other Emacs-style
keypresses work just fine on the Asterisk CLI, but Ctrl-W is a real
pain. I would go so far as to call this a bug as it is certainly
behaviour that differs from what the user expects.

Here are my system details:

*CLI core show version
Asterisk 11.5.1 built by dotancohen @ eris on a x86_64 running Linux
on 2013-10-07 19:20:49 UTC
*CLI Asterisk cleanly ending (0).
Executing last minute cleanups
 - eris():~$ uname -a
Linux eris 3.8.0-34-generic #49-Ubuntu SMP Tue Nov 12 18:00:10 UTC
2013 x86_64 x86_64 x86_64 GNU/Linux
 - eris():~$ cat /etc/issue
Ubuntu 13.04 \n \l



Interestingly, I don't even seem to have editline installed and thus
Asterisk is likely using its own packaged editline:

 - eris():~$ aptitude search editline
p   libeditline-dev- development files for
libeditline
p   libeditline-dev:i386   - development files for
libeditline
p   libeditline0   - line editing library
similar to readline
p   libeditline0:i386  - line editing library
similar to readline
p   libghc-editline-dev- Haskell bindings to
the BSD editline library (libedit
p   libghc-editline-dev:i386   - Haskell bindings to
the BSD editline library (libedit
v   libghc-editline-dev-0.2.1.1-b3dee  -
v   libghc-editline-dev-0.2.1.1-c95e1:i386 -
p   libghc-editline-doc- Haskell bindings to
the BSD editline library (libedit



Thank you!


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Re: [asterisk-users] Call Queue advise

2013-12-11 Thread Bryan Anderson
ok thanks.  The problem isn't agents not wanting to get calls.  Then they
just wont answer.  Some slower to answer users are complaining the people
are not working to be able to answer the call quicker.

@Paul Belanger  - Option two is where I am thinking but I am trying to
figure out the best way to do so.  my current thought is a Macro the queue
runs at the answer of a call.

Currently agents do not log in and out.  That is slatted for roll out after
a few more hires to that team.

Thanks,

-Bryan Anderson


On Wed, Dec 11, 2013 at 10:22 AM, Chad Wallace
cwall...@lodgingcompany.comwrote:


 On Mon, 9 Dec 2013 16:15:14 -0800
 Bryan Anderson shadow...@gmail.com wrote:

  On Mon, Dec 9, 2013 at 4:11 PM, Chad Wallace
  cwall...@lodgingcompany.comwrote:
 
   On Mon, 9 Dec 2013 15:47:57 -0800
   Bryan Anderson shadow...@gmail.com wrote:
  
I have a call queue that rings about 15 users and they are
wanting to set it up so that the last person to answer a call
doesn't ring on the next incoming call.
  
   Wouldn't the leastrecent strategy work for that?  It wouldn't
   absolutely forbid an agent from taking the next call, but it would
   make sure every other agent had priority.  You could also add a
   large wrap up time, to ensure they never get a second call within a
   certain time period.

  yes but I believe that least recent would ring one agent at a time?
  If my understanding is incorrect please correct it.  We are wanting
  to keep with multiple phones ring to ensure coverage.

 Yes, you're right.  It seems your solution (setting a penalty after
 they get a call) is probably the only one--unless, as has been
 suggested, you rethink your requirements. You could also remove them
 from the queue or pause them instead of changing their penalty.


 --

 C. Chad Wallace, B.Sc.
 The Lodging Company
 http://www.lodgingcompany.com/
 OpenPGP Public Key ID: 0x262208A0


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Re: [asterisk-users] dahdi show channels no such command

2013-12-11 Thread Joseph Towery





 From: Tzafrir Cohen tzafrir.co...@xorcom.com
To: asterisk-users@lists.digium.com 
Sent: Wednesday, December 11, 2013 1:37 PM
Subject: Re: [asterisk-users] dahdi show channels no such command
 

On Wed, Dec 11, 2013 at 09:22:13AM -0800, Joseph Towery wrote:

 Here is the output of lsdahdi:
 asteriskpbx@pbxmain:~$ sudo lsdahdi
 ### Span  1: WCTDM/0 Wildcard TDM800P (MASTER)
   1 FXO        FXSKS       (EC: MG2 - INACTIVE)
   2 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
   3 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
   4 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
   5 FXS        FXOKS       (EC: MG2 - INACTIVE)
   6 FXS        FXOKS       (EC: MG2 - INACTIVE)
   7 FXS        FXOKS       (EC: MG2 - INACTIVE)
   8 FXS        FXOKS       (EC: MG2 - INACTIVE)

For the record: this output shows that the DAHDI channels have been
configured (they got the FXSKS signalling, which means it was applied by
running dahdi_cfg) but they are not '(in use)', which means that no
channel has been opened by Asterisk's chan_dahdi.

 
Okay, I did run dahdi_cfg in an attempt to get the show channels working.  Is 
this a problem?  

Thanks,
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[asterisk-users] A Question about Management/Control Protocol Licensing

2013-12-11 Thread Steve Murphy
I see the following paragraph in the Asterisk trunk LICENSE file:

In addition, Asterisk implements two management/control protocols: the
Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
(AGI). It is our belief that applications using these protocols to
manage or control an Asterisk instance do not have to be licensed
under the GPL or a compatible license, as we believe these protocols
do not create a 'derivative work' as referred to in the GPL. However,
should any court or other judiciary body find that these protocols do
fall under the terms of the GPL, then we hereby grant you a license to
use these protocols in combination with Asterisk in external
applications licensed under any license you wish.

This probably originated some years ago, and I wonder if Digium or the
Asterisk
community might consider adding the OTHER management/control protocols to
this
list: ARI, and the ExternalIVR interface.

If not, it might be instructive to learn why!

murf



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Re: [asterisk-users] Ctrl-W killing entire line, not just last word

2013-12-11 Thread Tzafrir Cohen
On Wed, Dec 11, 2013 at 09:22:54PM +0200, Dotan Cohen wrote:

 Interestingly, I don't even seem to have editline installed and thus
 Asterisk is likely using its own packaged editline:

You need libedit-dev, not libeditline-dev.

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Re: [asterisk-users] dahdi show channels no such command

2013-12-11 Thread Tzafrir Cohen
On Wed, Dec 11, 2013 at 12:09:17PM -0800, Joseph Towery wrote:
 
 
 
 
 
  From: Tzafrir Cohen tzafrir.co...@xorcom.com
 To: asterisk-users@lists.digium.com 
 Sent: Wednesday, December 11, 2013 1:37 PM
 Subject: Re: [asterisk-users] dahdi show channels no such command
  
 
 On Wed, Dec 11, 2013 at 09:22:13AM -0800, Joseph Towery wrote:
 
  Here is the output of lsdahdi:
  asteriskpbx@pbxmain:~$ sudo lsdahdi
  ### Span  1: WCTDM/0 Wildcard TDM800P (MASTER)
    1 FXO        FXSKS       (EC: MG2 - INACTIVE)
    2 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
    3 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
    4 FXO        FXSKS       (EC: MG2 - INACTIVE)  RED
    5 FXS        FXOKS       (EC: MG2 - INACTIVE)
    6 FXS        FXOKS       (EC: MG2 - INACTIVE)
    7 FXS        FXOKS       (EC: MG2 - INACTIVE)
    8 FXS        FXOKS       (EC: MG2 - INACTIVE)
 
 For the record: this output shows that the DAHDI channels have been
 configured (they got the FXSKS signalling, which means it was applied by
 running dahdi_cfg) but they are not '(in use)', which means that no
 channel has been opened by Asterisk's chan_dahdi.
 
  
 Okay, I did run dahdi_cfg in an attempt to get the show channels working.  Is 
 this a problem?  

Certainly not.

-- 
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[asterisk-users] Language Coverage in Asterisk

2013-12-11 Thread Steve Murphy
I see that Asterisk distributes soundsets for
English/English-AU/Spanish/French, and Russian.

There is code for several other languages inside Asterisk;
how does one obtain the other soundsets?

Also, I noted that the source sound files don't seem to be publicly
available for the sound sets that Asterisk distributes. I would assume that
the source sounds are all 44khz (or more) cd quality sounds, probably in
pcm wave format, maybe even in stereo. Are these source
sound sets indeed withheld from the public? Or am I mistaken in my
impression?


murf


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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread Olivier
2013/12/11 jg webaccou...@jgoettgens.de


 At the moment, I'm having trouble with lines going up and down all day
 long.
 I asked them to be created as P2P.
 Mostly but not always calls are passing in and out correctly.

 I doubt that the energy-saving is causing the trouble. How do you get the
 ISDN-signal? Is it an S0-bus?


I'm not 100% sure I'm correctly understanding these latter two questions.

Anyway, refering to
http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network / Basic
rate Interface paragraph, my setup includes :
- a single network termination unit, installed and managed by telco, is
connected to public network through 2 twisted pairs,
- this unit includes two small electronic cards,
- each card has two RJ45 sockets,
- each card is connected through a single Cat5 straight, 1m long patch cord
to a single BRI port from my Patton box,
- both Patton's BRI ports are managed as a hunt group (for outbound
calls),
- for inbound, I didn't checked but usually, Telco sends one call on one
interface, the next one and the next interface.

So to summarize, (I think):
- I'm getting ISDN signal from two distinct copper pairs,
- it is not an S0-bus.


 If yes, I may be able to be of further help. Even if the connection goes
 down, it should not matter for P2MP connections.


You mean: layer1 is down but layer3 remains up ?
I'm seeing layer3 going down for 4 or 5 seconds, then up again. This
happens for both interfaces.


 But S0-bus problems can be nasty (though seen rarely these days).


 So, if I'm not mistaken, either :
 - these lines are configured as P2MP (and not as P2P as required),
 - I'm facing something else I've never met before.

 I would bet on the 2nd choice.



From memory, with a misconfiguration between P2P and P2MP, you get no call
at all which is not the case.



 As these lines are up most of the time, I'm having a hard time to get any
 useful help by Telco beside level 1 basic checks.

 For the record, these lines are terminated on my side with a Patton
 smartnode (most of the time, I use Digium BRI boards).
 During off hours, I'll try to check what happens when toggling to P2MP
 mode.

 I doubt that you do get anything useful if you pick the wrong mode.

 Are you sure that your are not dealing with termination issues?

Try sending a T.30 fax with fine resolution and ecm enabled when your
 connection is up. If it takes too long or fails, this is something to look
 at.


Can you elaborate a bit ?
I'm using an old Asterisk 1.6.1 version there so I can't rely on T.38/T.30
gateway feature at the moment.



 jg


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Re: [asterisk-users] A Question about Management/Control Protocol Licensing

2013-12-11 Thread Paul Belanger

On 13-12-11 03:15 PM, Steve Murphy wrote:

I see the following paragraph in the Asterisk trunk LICENSE file:

In addition, Asterisk implements two management/control protocols: the
Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
(AGI). It is our belief that applications using these protocols to
manage or control an Asterisk instance do not have to be licensed
under the GPL or a compatible license, as we believe these protocols
do not create a 'derivative work' as referred to in the GPL. However,
should any court or other judiciary body find that these protocols do
fall under the terms of the GPL, then we hereby grant you a license to
use these protocols in combination with Asterisk in external
applications licensed under any license you wish.

This probably originated some years ago, and I wonder if Digium or the
Asterisk
community might consider adding the OTHER management/control protocols to
this
list: ARI, and the ExternalIVR interface.

If not, it might be instructive to learn why!

Would also like to see this update to include ARI. We talked a little 
about it at astridevcon, and I think it is likely an oversight.


--
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Github: https://github.com/pabelanger | Twitter: 
https://twitter.com/pabelanger


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[asterisk-users] Asterisk Language Status

2013-12-11 Thread Steve Murphy
In putting together the SoundPack code, I am looking at the
various language/locale specific code, and wondering how it all
really stands...

So, share with me, non-English speakers, what is your experience
and impression?

I heard a few comments during AstriDevCon, that some of the languages are
not quite right; some said their language was understandable, but...

Would anyone be willing to share with me, the problems they have with
various translations?

Do they pronounce numbers in a grammatically correct fashion? Any issues
with gender/tense/whatever?

Are the sound sets in other languages easily findable and downloadable?
I see a good selection on
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international

murf


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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread Olivier
I double-checked with other locations (also using Patton devices and I
didn't get any complain). I can find the same pattern everywhere:

2013-12-11T20:17:39 : LOGINFO: Link down on interface bri 0 0 0
2013-12-11T20:17:39 : LOGINFO: Link down on interface ISDN 0 0
2013-12-11T20:18:11 : LOGINFO: Link up on interface bri 0 0 0
2013-12-11T20:18:11 : LOGINFO: Link up on interface ISDN 0 0
2013-12-11T20:18:14 : LOGINFO: Link down on interface bri 0 0 0
2013-12-11T20:18:15 : LOGINFO: Link down on interface ISDN 0 0
2013-12-11T20:18:18 : LOGINFO: Link up on interface bri 0 0 0
2013-12-11T20:18:18 : LOGINFO: Link up on interface ISDN 0 0
2013-12-11T20:21:14 : LOGINFO: Link down on interface bri 0 0 1
2013-12-11T20:21:15 : LOGINFO: Link down on interface ISDN 0 1
2013-12-11T20:21:46 : LOGINFO: Link up on interface bri 0 0 1
2013-12-11T20:21:46 : LOGINFO: Link up on interface ISDN 0 1
2013-12-11T20:21:49 : LOGINFO: Link down on interface bri 0 0 1
2013-12-11T20:21:49 : LOGINFO: Link down on interface ISDN 0 1
2013-12-11T20:21:53 : LOGINFO: Link up on interface bri 0 0 1
2013-12-11T20:21:53 : LOGINFO: Link up on interface ISDN 0 1
2013-12-11T20:37:39 : LOGINFO: Link down on interface bri 0 0 0
2013-12-11T20:37:39 : LOGINFO: Link down on interface ISDN 0 0
2013-12-11T20:38:11 : LOGINFO: Link up on interface bri 0 0 0
2013-12-11T20:38:11 : LOGINFO: Link up on interface ISDN 0 0
2013-12-11T20:38:13 : LOGINFO: Link down on interface bri 0 0 0
2013-12-11T20:38:13 : LOGINFO: Link down on interface ISDN 0 0
2013-12-11T20:38:13 : LOGINFO: Link up on interface bri 0 0 0
2013-12-11T20:38:14 : LOGINFO: Link up on interface ISDN 0 0
2013-12-11T20:41:14 : LOGINFO: Link down on interface bri 0 0 1
2013-12-11T20:41:14 : LOGINFO: Link down on interface ISDN 0 1
2013-12-11T20:41:46 : LOGINFO: Link up on interface bri 0 0 1
2013-12-11T20:41:46 : LOGINFO: Link up on interface ISDN 0 1
2013-12-11T20:41:49 : LOGINFO: Link down on interface bri 0 0 1
2013-12-11T20:41:49 : LOGINFO: Link down on interface ISDN 0 1
2013-12-11T20:41:53 : LOGINFO: Link up on interface bri 0 0 1
2013-12-11T20:41:53 : LOGINFO: Link up on interface ISDN 0 1

So it seems common from Telco to turn ISDN lines going down and up, once in
while and this is not for energy savings.


2013/12/11 jg webaccou...@jgoettgens.de


 At the moment, I'm having trouble with lines going up and down all day
 long.
 I asked them to be created as P2P.
 Mostly but not always calls are passing in and out correctly.

 I doubt that the energy-saving is causing the trouble. How do you get the
 ISDN-signal? Is it an S0-bus? If yes, I may be able to be of further help.
 Even if the connection goes down, it should not matter for P2MP
 connections. But S0-bus problems can be nasty (though seen rarely these
 days).


 So, if I'm not mistaken, either :
 - these lines are configured as P2MP (and not as P2P as required),
 - I'm facing something else I've never met before.

 I would bet on the 2nd choice.


 As these lines are up most of the time, I'm having a hard time to get any
 useful help by Telco beside level 1 basic checks.

 For the record, these lines are terminated on my side with a Patton
 smartnode (most of the time, I use Digium BRI boards).
 During off hours, I'll try to check what happens when toggling to P2MP
 mode.

 I doubt that you do get anything useful if you pick the wrong mode.

 Are you sure that your are not dealing with termination issues? Try
 sending a T.30 fax with fine resolution and ecm enabled when your
 connection is up. If it takes too long or fails, this is something to look
 at.


 jg


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Re: [asterisk-users] Asterisk Language Status

2013-12-11 Thread Administrator TOOTAI

Hi Steve,

Le 11/12/2013 22:23, Steve Murphy a écrit :


In putting together the SoundPack code, I am looking at the
various language/locale specific code, and wondering how it all
really stands...

So, share with me, non-English speakers, what is your experience
and impression?

I heard a few comments during AstriDevCon, that some of the languages 
are not quite right; some said their language was understandable, but...


Would anyone be willing to share with me, the problems they have with 
various translations?


For instance, beside French, you can also have Canadian French, Belgium 
French, Switzerland French, ...


Some words are not the same or not existing, accent is different, saying 
numbers (90 is quatre ving dix in French but nonente in Belgium or 
Switzerland) aso.




Do they pronounce numbers in a grammatically correct fashion? Any 
issues with gender/tense/whatever?


Are the sound sets in other languages easily findable and downloadable?
I see a good selection on
http://www.voip-info.org/wiki/view/Asterisk+sound+files+international



Yes, valid selection. FYI Debian has ready packages for languages.

Regards

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Daniel

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Re: [asterisk-users] A Question about Management/Control Protocol Licensing

2013-12-11 Thread Matthew Jordan
On Wed, Dec 11, 2013 at 3:15 PM, Paul Belanger paul.belan...@polybeacon.com
 wrote:

 On 13-12-11 03:15 PM, Steve Murphy wrote:

 I see the following paragraph in the Asterisk trunk LICENSE file:

 In addition, Asterisk implements two management/control protocols: the
 Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
 (AGI). It is our belief that applications using these protocols to
 manage or control an Asterisk instance do not have to be licensed
 under the GPL or a compatible license, as we believe these protocols
 do not create a 'derivative work' as referred to in the GPL. However,
 should any court or other judiciary body find that these protocols do
 fall under the terms of the GPL, then we hereby grant you a license to
 use these protocols in combination with Asterisk in external
 applications licensed under any license you wish.

 This probably originated some years ago, and I wonder if Digium or the
 Asterisk
 community might consider adding the OTHER management/control protocols to
 this
 list: ARI, and the ExternalIVR interface.

 If not, it might be instructive to learn why!

  Would also like to see this update to include ARI. We talked a little
 about it at astridevcon, and I think it is likely an oversight.


It isn't an oversight. It's on my ToDo list (and this item is an action
item on the wiki as well). We had the Thanksgiving holiday; then I was out
last week at AdhearsionConf (great conference!). The licensing file will
get updated before 12 is released.

As an aside, we also had conversations about it on the asterisk-app-dev
list [1], where I responded that I would get answers to the licensing
questions. Granted, it has been much longer than a week or two - mea culpa
on a bad time estimate.

[1]
http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000127.html

Matt

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445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com  http://asterisk.org
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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread jg


So it seems common from Telco to turn ISDN lines going down and up, once in while and this is 
not for energy savings.

I don't think so. There is another reason for you problem.


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Re: [asterisk-users] A Question about Management/Control Protocol Licensing

2013-12-11 Thread Steve Murphy
On Wed, Dec 11, 2013 at 3:29 PM, Matthew Jordan mjor...@digium.com wrote:


 On Wed, Dec 11, 2013 at 3:15 PM, Paul Belanger 
 paul.belan...@polybeacon.com wrote:

 On 13-12-11 03:15 PM, Steve Murphy wrote:

 I see the following paragraph in the Asterisk trunk LICENSE file:

 In addition, Asterisk implements two management/control protocols: the
 Asterisk Manager Interface (AMI) and the Asterisk Gateway Interface
 (AGI). It is our belief that applications using these protocols to
 manage or control an Asterisk instance do not have to be licensed
 under the GPL or a compatible license, as we believe these protocols
 do not create a 'derivative work' as referred to in the GPL. However,
 should any court or other judiciary body find that these protocols do
 fall under the terms of the GPL, then we hereby grant you a license to
 use these protocols in combination with Asterisk in external
 applications licensed under any license you wish.

 This probably originated some years ago, and I wonder if Digium or the
 Asterisk
 community might consider adding the OTHER management/control protocols to
 this
 list: ARI, and the ExternalIVR interface.

 If not, it might be instructive to learn why!

  Would also like to see this update to include ARI. We talked a little
 about it at astridevcon, and I think it is likely an oversight.


 It isn't an oversight. It's on my ToDo list (and this item is an action
 item on the wiki as well). We had the Thanksgiving holiday; then I was out
 last week at AdhearsionConf (great conference!). The licensing file will
 get updated before 12 is released.

 As an aside, we also had conversations about it on the asterisk-app-dev
 list [1], where I responded that I would get answers to the licensing
 questions. Granted, it has been much longer than a week or two - mea culpa
 on a bad time estimate.

 [1]
 http://lists.digium.com/pipermail/asterisk-app-dev/2013-October/000127.html

 Matt


​Many thanks, Matt for this info!

-- 

Steve Murphy
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Re: [asterisk-users] [OT] Does Energy Savings mode exist with BRI/PtP ?

2013-12-11 Thread jg




Anyway, refering to http://en.wikipedia.org/wiki/Integrated_Services_Digital_Network / Basic 
rate Interface paragraph, my setup includes :
- a single network termination unit, installed and managed by telco, is connected to public 
network through 2 twisted pairs,

- this unit includes two small electronic cards,
- each card has two RJ45 sockets,
- each card is connected through a single Cat5 straight, 1m long patch cord to a single BRI 
port from my Patton box,

- both Patton's BRI ports are managed as a hunt group (for outbound calls),
- for inbound, I didn't checked but usually, Telco sends one call on one interface, the next 
one and the next interface.


So to summarize, (I think):
- I'm getting ISDN signal from two distinct copper pairs,
- it is not an S0-bus.
Termination Unit probably means you get the T reference point and behind the unit you do have 
the S reference point with a 4 wire signal, which would be a standard setup.



Are you sure that your are not dealing with termination issues?

Try sending a T.30 fax with fine resolution and ecm enabled when your 
connection is up. If
it takes too long or fails, this is something to look at.


Can you elaborate a bit ?
Maybe it is easier if you could  mention your country and your telco. I would guess you are 
either in France or Belgium, but I could be wrong. Then we could simply lookup what is relevant 
for you.
I'm using an old Asterisk 1.6.1 version there so I can't rely on T.38/T.30 gateway feature at 
the moment.
I said T.30 and not T.38 and the complete explanation would be rather technical. In essence a 
T.30 fax with ecm enabled does not react nicely to certain signaling errors and pertubations and 
this is easy to detect. Most of the time I can guess what the problem is when I see the type of 
error. Don't ask me how I learned this. Once you know what to look for, Asterisk 1.whatever is 
very reliable as far as facsimiles go.


jg
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