[asterisk-users] Queuing: calls stay in queue and agents are ready !!

[bilal ghayyad]

Hi All;

I configured some queues, and I configured the dialed numbers for login and 
logout for the agents. 

Two agents are logged in, the first two calls are received at the agents and 
they answered and hangup. Again, the two agents are idle and ready to receive 
calls. The third and call goes to queue and stay in waiting although the agents 
are ready !!! We disconnect the call and re call again, the same thing (the 
call goes for the queue in the waiting and does not go for the agents who are 
ready to receive calls).

I tried to change something in the settings, I made the autofill=yes and the 
autopause=no without any success (the same problem). What could cause for such 
behaviour?

What is the parameter or the settings that make such thing happen?

Regards
Bilal

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Re: [asterisk-users] Queuing: calls stay in queue and agents are ready !!

[Tarek Sawah]

Bilal , 
if you can do a "core show queue  QUEUENUMBER" and paste the output here at the 
moment of this problem it will be helpful to see what is the status of your 
agents at that moment for the queue.
does it help if the agents logout then back in instead of "disconnecting the 
call and calling back again"?
what is the timeout for the agent setup in the queue settings? or more helpful 
if you paste your queue settings



Tarek Sawah

Information Technology  Adviser

Integrated Digital Systems

CCNP, MCSE, RHCE, TELECOM

USA: +1 386 492 9993




> Date: Mon, 19 Sep 2011 02:46:59 -0700
> From: bilmar...@yahoo.com
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-users] Queuing: calls stay in queue and agents are ready !!
>
> Hi All;
>
> I configured some queues, and I configured the dialed numbers for login and 
> logout for the agents.
>
> Two agents are logged in, the first two calls are received at the agents and 
> they answered and hangup. Again, the two agents are idle and ready to receive 
> calls. The third and call goes to queue and stay in waiting although the 
> agents are ready !!! We disconnect the call and re call again, the same thing 
> (the call goes for the queue in the waiting and does not go for the agents 
> who are ready to receive calls).
>
> I tried to change something in the settings, I made the autofill=yes and the 
> autopause=no without any success (the same problem). What could cause for 
> such behaviour?
>
> What is the parameter or the settings that make such thing happen?
>
> Regards
> Bilal
>
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Re: [asterisk-users] [1.6.2.9] Echo even when using headset?

[Gilles]

On Sun, 18 Sep 2011 22:28:32 +0200, Gilles 
wrote:
>For some reason, even through I'm using a headset, there's a lot of
>echo and after a few seconds, it sounds like it enters a very fast
>loop before the echo stops somewhat. IOW, unusable sound.

Problem solved: Tried XLite 4.1 on another test, and sound is OK, so I
guess it's something in my work PC.


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[asterisk-users] Anyone got a working SCCP configuration for a Cisco 6945?

[ft...@mindspring.com]

I'm trying to set-up a Cisco 6945 with SCCP firmware under AsterisK 1.6 
with sccp-b 3.0.4.  Does anyone have a working example they can share: 
the TFTP root directory files and the sccp.conf file?


Thanks.

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[asterisk-users] NC DATA FINDOUT IN AUTO DIALER

[mahesh katta]

Hi List,

I have one query, I am using Go autodial in this using auto dialing.
autodial can do only whenever customer pick the call that call will go to
agents.
but problem autodial dialing the database in that I am not getting NC data
means, not reachable,switch off ,outofservice data. how can I get this data.
is there any software get the telcovoice and give report ?

Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com
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[asterisk-users] /usr/sbin/asterisk -rx and AMI

[Jonas Kellens]

Hello,

currently I run a php script in cron which polls for information on SIP 
peers using the /usr/sbin/asteirsk -rx method.


I notice that after some time the Asterisk interface freezes, SIP Peer 
registrations become unreachable and sip reload or any other command on 
the CLI does not respond !

Solution : /sbin/service asterisk restart

Is it possible that the 2 minute polling interval is to much load for 
the Asterisk-proces ?


Would I have better luck by using AMI to poll SIP peers ?



Kind regards,

Jonas.
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[asterisk-users] question on DTMF

[Jerry Geis]

 I am running asterisk 1.4.41.2 and dahdi 2.4.1 (64 bit centos)
I only have one small issue.

I initiate a call over AMI, call is answered and I run my AGI.
"sometimes" when I make calls out to cell phones I ask to press 1 to confirm
the user hears the message and press 1 but I never get the 1 back on my 
side.

Most times the digit is sent back - just sometimes its not.

I have a SIP trunk connection to cisco call manager version 6 something 
I think on the customers side.


Is there any reason why sometimes I would not get the DTMF digit back?

Jerry

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Re: [asterisk-users] /usr/sbin/asterisk -rx and AMI

[Danny Nicholas]

From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jonas Kellens
Sent: Monday, September 19, 2011 9:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] /usr/sbin/asterisk -rx and AMI

 

Hello,

currently I run a php script in cron which polls for information on SIP
peers using the /usr/sbin/asteirsk -rx method.

I notice that after some time the Asterisk interface freezes, SIP Peer
registrations become unreachable and sip reload or any other command on the
CLI does not respond !
Solution : /sbin/service asterisk restart

Is it possible that the 2 minute polling interval is to much load for the
Asterisk-proces ?

Would I have better luck by using AMI to poll SIP peers ?



Kind regards,

Jonas.

 

You don't say what release of Asterisk you are running, but here are a few
generalities:

1.   Asterisk -rx uses more overhead but shares many of the handles that
the AMI interface uses

2.   If you are using flat files, you could poll /var/log/asterisk/full
for this information

3.   If you are using MYSQL files, you could poll independently as well

4.   The 2-minute interval should not be overloading asterisk, but may
be overkill/underkill depending on your register timeouts.

 

Hope this is useful to you.

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Re: [asterisk-users] question on DTMF

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, September 19, 2011 9:55 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] question on DTMF

  I am running asterisk 1.4.41.2 and dahdi 2.4.1 (64 bit centos) I only have
one small issue.

I initiate a call over AMI, call is answered and I run my AGI.
"sometimes" when I make calls out to cell phones I ask to press 1 to confirm
the user hears the message and press 1 but I never get the 1 back on my
side.
Most times the digit is sent back - just sometimes its not.

I have a SIP trunk connection to cisco call manager version 6 something I
think on the customers side.

Is there any reason why sometimes I would not get the DTMF digit back?

Jerry

Depending on the cell phone you are calling, the DTMF length may need to be
set to LONG (I know this applies to Verizon phones).


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Re: [asterisk-users] single registration per user

[Eric Wieling]

No, I have no suggestions.  What you are describing has nothing whatsoever to 
do with registration.  Registration only applies to calls from Asterisk to the 
phone.  It has nothing to do with calls from the phone to Asterisk.

-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
Sent: Sunday, September 18, 2011 4:23 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] single registration per user

Hello Eric,
 
Is about outgoing calls from multiple devices with the same username at aprox 
same time. The overwritten is for incomming calls. I want to prevent using the 
same account in multiple devices at same time. The solution with IP will not 
apply because users may be behind nat or will change everytime multiple access 
points. Do you have any other clues?

Thank you for answers,
Best regards.


On Sun, Sep 18, 2011 at 8:37 PM, Eric Wieling  wrote:


Asterisk only allows one device per peer to register.  If a 2nd device 
registers, the first registration is overwritten.

You can use permit/deny to limit which IPs a device can register from.


-Original Message-
From: asterisk-users-boun...@lists.digium.com 
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Catalin S.
Sent: Sunday, September 18, 2011 4:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users] single registration per user

Hello,

I use asterisk 1.8.6.0 and I have aprox 100 extensions. I want to lock 
every extension to a single registration per device. Many of users tried to log 
on my asterisk from 2, 3 devices and I want allow only one.
Is there any solution for fix this?

Thank you.


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Re: [asterisk-users] question on DTMF

[Jerry Geis]

Depending on the cell phone you are calling, the DTMF length may need to be
set to LONG (I know this applies to Verizon phones).

Danny

I am not familiar with this setting - where is it exactly.
I looked in my sip.conf and did not see anything.

Thanks-

Jerry


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Re: [asterisk-users] question on DTMF

[Danny Nicholas]

-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Monday, September 19, 2011 10:07 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] question on DTMF


> Depending on the cell phone you are calling, the DTMF length may need 
> to be set to LONG (I know this applies to Verizon phones).
Danny

I am not familiar with this setting - where is it exactly.
I looked in my sip.conf and did not see anything.

Thanks-

Jerry

It's not an Asterisk setting - it just affects Asterisk (and some other
PBXs) - in your phone settings, you can set DTMF tones to normal or long.


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Re: [asterisk-users] single registration per user

[Steve Edwards]

On Sun, 18 Sep 2011, Catalin S. wrote:

Is about outgoing calls from multiple devices with the same username at 
aprox same time. The overwritten is for incomming calls. I want to 
prevent using the same account in multiple devices at same time. The 
solution with IP will not apply because users may be behind nat or will 
change everytime multiple access points. Do you have any other clues?


If your intent is to limit the number of outgoing calls from a username, 
how about the GROUP and GROUP_COUNT functions?


If the GROUP_COUNT(username) is 0, set GROUP(username) and allow the call.

--
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-
Steve Edwards   sedwa...@sedwards.com  Voice: +1-760-468-3867 PST
Newline  Fax: +1-760-731-3000

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[asterisk-users] iLBC support in Asterisk after Google's acquisition of GIPS

[Asterisk Development Team]

Recently, we were notified that the mechanism included in our Asterisk
source code releases to download and build support for the iLBC codec
had stopped working correctly; a little investigation revealed that this
occurred because of some changes on the ilbcfreeware.org website. These
changes occurred as result of Google's acquisition of GIPS, who produced
(and provided licenses for) the iLBC codec.

We've determined that the change necessary to fix Asterisk's iLBC build
process is rather trivial, and so we're planning to make that change in
Asterisk 1.8.7.0-rc2, and subsequently in 1.8.7.0. We are not planning
on making new releases of Asterisk 1.4 and Asterisk 1.6.2, since they
are in security-maintenance mode and this is not a security issue. Users
who wish to make the same change on their own to their copies of those
versions are of course welcome to do so.

As part of the process of determining what had broken here, we also
became aware that the ilbcfreeware.org website no longer offers the iLBC
license agreement it used to offer; this agreement was required by the
iLBC licensors (GIPS) in order for users to safely distribute and use
iLBC (and this is why the Asterisk project does not include the iLBC
source code directly with Asterisk). The removal of this license
agreement also occurred as a result of the Google acquisition, but as of
this moment no alternative has been made available for those who wish to
use the iLBC source code published in RFC 3951 (which Asterisk uses).

Google does have an alternative implementation of iLBC available as part
of the WebRTC project, with a license that is compatible with Asterisk
(and does not require written agreements from end users), but the
codec_ilbc module in Asterisk cannot be built against the WebRTC
implementation of iLBC. Until such time as we have an improved version
of codec_ilbc, Asterisk users will have to continue using the RFC 3951
iLBC source code.

Unfortunately, that leaves Asterisk users in a bit of a bind; if they
had already signed and sent in the GIPS iLBC license agreement, we
believe they can continue to safely use the existing iLBC
implementation. New users, though, do not have the option of agreeing to
a license agreement that would allow them to use the RFC 3951 iLBC
source code, as there is no mechanism to do that currently available.
We've contacted Google and they are aware of the dilemma, and have said
that they will address it, but we don't have a timeframe for when an
alternative license mechanism will be available.

In summary, if you are a user of Asterisk and iLBC together, and you've
already executed a license agreement with GIPS, we believe you can
continue using iLBC with Asterisk. If you are a user of Asterisk and
iLBC together, but you had not executed a license agreement with GIPS,
we encourage you to research the situation and consult with your own
legal representatives to determine what actions you may want to take (or
avoid taking).

-- Asterisk Development Team

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[asterisk-users] SIP OPTIONS... Error?

[Bruce Ferrell]

I know over time SIP OPTIONS message handling has changed and I've seen 
some write ups that seem to indicate that an s extension in the default 
context is needed now to get them to work.


It's probably my error in any case.

So, what am I doing wrong or what do I need to do to get the sip ping to 
work?


Bruce Ferrell

Just for fun, I created a sip peer called ping at a fixed address with a 
default context of ping:


sip.conf
===
[ping]
type=peer
host=XX.XX.XX.XX
defaultip=XX.XX.XX.XX
context=ping

extensions.conf

[ping]
exten => s,1,NoOp()


I am sending the following:


OPTIONS sip:YY.YY.YY.YY SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5061;branch=z9hG4bK1316447867
From: 
To: 
Contact: 
Call-ID: 20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx
CSeq: 102 OPTIONS
User-Agent: sip_ping.pl
Date: Mon, 19 September 2011 08:57:47 PDT
Allow: ACK, CANCEL
Content-Length: 0


And the asterisk instance is logging the following with core set debug 9:

<--- SIP read from UDP:XX.XX.XX.XX:6051 --->
OPTIONS sip:YY.YY.YY.YY SIP/2.0
Via: SIP/2.0/UDP XX.XX.XX.XX:5061;branch=z9hG4bK1316447867
From: 
To: 
Contact: 
Call-ID: 20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx
CSeq: 102 OPTIONS
User-Agent: sip_ping.pl
Date: Mon, 19 September 2011 08:57:47 PDT
Allow: ACK, CANCEL
Content-Length: 0
<->
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  0 [ 32]: OPTIONS 
sip:YY.YY.YY.YY SIP/2.0
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  1 [ 59]: Via: 
SIP/2.0/UDP XX.XX.XX.XX:5061;branch=z9hG4bK1316447867
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  2 [ 29]: From: 

[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  3 [ 22]: To: 

[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  4 [ 32]: Contact: 

[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  5 [ 54]: Call-ID: 
20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  6 [ 17]: CSeq: 
102 OPTIONS
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  7 [ 23]: 
User-Agent: sip_ping.pl
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  8 [ 41]: Date: 
Mon, 19 September 2011 08:57:47 PDT
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header  9 [ 18]: Allow: 
ACK, CANCEL
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c:  Header 10 [ 18]: 
Content-Length: 0

[2011-09-19 08:57:47] VERBOSE[2979] chan_sip.c: --- (11 headers 0 lines) ---
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c: = Looking for  Call ID: 
20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx (Checking From) --From tag  
--To-tag
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c: OPTIONS request has no 
from tag, dropping callid: 20d664d681edcbd48dd76f57915d4...@xx.xx.xx.xx 
from: 
[2011-09-19 08:57:47] DEBUG[2979] chan_sip.c: Invalid SIP message - 
rejected , no callid, len 355


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Re: [asterisk-users] SIP OPTIONS... Error?

[Alex Balashov]

Every request needs a From tag.

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Re: [asterisk-users] SIP OPTIONS... Error?

[Bruce Ferrell]

On 09/19/2011 09:33 AM, Alex Balashov wrote:

Every request needs a From tag.



Uh... OK.  Isn't this a From tag:

From: 

Line three of what I send?


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Re: [asterisk-users] SIP OPTIONS... Error?

[Alex Vishnev]

no, you need a tag i.e From: ;tag=xxx, where xx 
is a unique identifier

see the definition of SIP Dialog

Dialog: A dialog is a peer-to-peer SIP relationship between two
 UAs that persists for some time.  A dialog is established by
 SIP messages, such as a 2xx response to an INVITE request.  A
 dialog is identified by a call identifier, local tag, and a
 remote tag.  A dialog was formerly known as a call leg in RFC
 2543.

On Sep 19, 2011, at 1:11 PM, Bruce Ferrell wrote:

> On 09/19/2011 09:33 AM, Alex Balashov wrote:
>> Every request needs a From tag.
>> 
> 
> Uh... OK.  Isn't this a From tag:
> 
> From: 
> 
> Line three of what I send?
> 
> 
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Re: [asterisk-users] SIP OPTIONS... Error?

[Alex Balashov]

On 09/19/2011 01:11 PM, Bruce Ferrell wrote:

On 09/19/2011 09:33 AM, Alex Balashov wrote:

Every request needs a From tag.



Uh... OK. Isn't this a From tag:

From: 

Line three of what I send?


No, that's a From URI.

A From tag is a header parameter that is appended to the URI, 
delimited by a semicolon:


   From: ;tag=abc123xyz

Although, RFC 3261 Section 12.1.1 ("UAS behavior") does seem to 
contradict me:


   A UAS MUST be prepared to receive a request without a tag in
   the From field, in which case the tag is considered to have
   a value of null.

However, it goes on to say:

   This is to maintain backwards compatibility with RFC 2543,
   which did not mandate From tags.

In other words, a non-backward-compatible 3261 implementation will 
always generate From tags for all requests.


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Re: [asterisk-users] SIP OPTIONS... Error?

[Alex Balashov]

On 09/19/2011 01:16 PM, Alex Vishnev wrote:


no, you need a tag i.e From: ;tag=xxx, where
xx is a unique identifier

see the definition of SIP Dialog

Dialog: A dialog is a peer-to-peer SIP relationship between two
  UAs that persists for some time.  A dialog is established by
  SIP messages, such as a 2xx response to an INVITE request.  A
  dialog is identified by a call identifier, local tag, and a
  remote tag.  A dialog was formerly known as a call leg in RFC
  2543.


OPTIONS requests don't create a dialog, just a transaction.

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Fax: +1-404-961-1892
Web: http://www.evaristesys.com/

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Re: [asterisk-users] SIP OPTIONS... Error?

[Bruce Ferrell]

Thank you Alex.  That was useful information.   The sip_ping.pl program 
I was using used to work without the tag.  It seems asterisk now demands it.


Bruce Ferrell

On 09/19/2011 10:18 AM, Alex Balashov wrote:

On 09/19/2011 01:16 PM, Alex Vishnev wrote:


no, you need a tag i.e From: ;tag=xxx, where
xx is a unique identifier

see the definition of SIP Dialog

Dialog: A dialog is a peer-to-peer SIP relationship between two
  UAs that persists for some time.  A dialog is established by
  SIP messages, such as a 2xx response to an INVITE request.  A
  dialog is identified by a call identifier, local tag, and a
  remote tag.  A dialog was formerly known as a call leg in RFC
  2543.


OPTIONS requests don't create a dialog, just a transaction.




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[asterisk-users] Looking for Asterisk-FreePBX in a Flash Support in Tampa

[Keith Ware]

Hello,

Looking for Asterisk-FreePBX in a Flash Technical Support in Tampa

Please contact me by phone: 813-842-6941



Thank You,

[cid:image001.gif@01CC76D3.D2584900]
Secure2ware Inc.
813-425-5900
Keith A. Ware, ext. 211
ke...@secure2ware.com
www.secure2ware.com


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[asterisk-users] oddity with CISCO CCM and Asterisk

[Danny Nicholas]

Hi List,

 I have a system that connects into Asterisk 1.4.41 using CISCO
CCM 7.  Everything works great except when a call is transferred to the
operator.  The call goes to the operator via a native bridge and is
completed, then a "phantom process" starts and tries to launch a new call
every 15 minutes.  I modified the dialplan to hangup these phantom calls,
but no still joy.  I get this message:

[Sep 19 10:32:47] NOTICE[14249] chan_sip.c: Call from 'XXX' to extension
'X' rejected because extension not found.  

 

14249 is not showing as running, but it is accessible vi pmap 14249

 

Could this be a problem with pbx_loopback?

 

Thanks

Danny Nicholas

 

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[asterisk-users] Asterisk-FreeBPX in Flash Support in Tampa_Not needed any longer

[Keith Ware]

Hello All,

Thank you all for responding so quickly.

I just hired someone for Technical Support.

Just wanted to inform everyone.


Thank you all and good luck to everyone!


Keith



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[asterisk-users] Ghost DID in System

[Aaron Krohn]

This is going to sound ridiculous, but there appears to be a ghost DID 
in our system. We are going to get the number ported to us, but it has 
not happened yet. From a phone outside of our voip system, the call 
still goes through. When calling the did from a phone within our system, 
there is just dead air.


In the asterisk CLI, I can see our primary server, voip1 trying to do 
pass the call to voip2 after it complains about not knowing what to do 
with the call.


I have removed all references to this number from all dialplans and 
sip-did lists and restarted many times. I simply don't understand why 
our voip1 server believes it should try to route the call instead of 
passing it to the outside world. Does anyone have an explanation or know 
where I could look? (dialplans, obviously =)



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Re: [asterisk-users] single registration per user

[Dave Platt]

> Is about outgoing calls from multiple devices with the same username at
> aprox same time. The overwritten is for incomming calls. I want to prevent
> using the same account in multiple devices at same time. The solution with
> IP will not apply because users may be behind nat or will change everytime
> multiple access points. Do you have any other clues?

As others have noted, this doesn't really have anything to do with
"registration" per se.

Registration by a user, tells where calls *to* that user should be
sent (IP address and port).

Authorization to initiate an outbound call through Asterisk doesn't
depend on the device having registered.  It depends on the device
sending an INVITE with the appropriate user-ID, and the device's
ability to respond to the corresponding security challenge from
Asterisk.  Any device having the appropriate ID and secret can
thus authenticate on the outbound call... Asterisk won't (unless
you jump through a lot of hoops) "know" whether this is the same
device that has currently registered with that ID.

I can think of several approaches which might work:

(1) Set "call-limit=1" in the SIP user definition for this user.
This will (if I'm reading the documentation correctly) limit
Asterisk to only one call to this user/peer at a time.  There
used to be separate limits for "incoming" and "outgoing" calls,
but that was eliminated several versions ago.

(2) As others have suggested, do it in the dialplan using the GROUP
function.  Perhaps the simplest way to do this would be to
set up a dialplan context which each of these users is bound to.
In its "s" ruleset, set the GROUP() value to be the user-ID, and
then check the number of members in the group... if it's more than
1, jump to a rule which does a Congestion() or plays a "You are
a cheater and I make rude motions in your direction" recorded message
or hangs up or ...

If the group-count test succeeds, jump to another dialing context
which actually does the dialing based on the $EXTEN passed by the
caller.

This would be a bit like example 2 in the page at
http://www.voip-info.org/wiki/view/Asterisk+func+group but you
would use a specific group name per user e.g. $CHANNEL(peername)
rather than a group per outbound trunk.

(3) Do something like (2), but instead of using the GROUP feature to
limit calls, compare the caller's IP address with the IP address
currently registered by the calling user e.g. compare
$CHANNEL(peerip) with $SIPPEER($CHANNEL(peername),ip).

This wouldn't be as robust as approach (2) - there would probably be
moments when a second device could make a call - and so I don't really
encourage it.



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Re: [asterisk-users] NC DATA FINDOUT IN AUTO DIALER

[Nasir Iqbal]

Please check our voice sms and fax broadcasting / smart autodialler / smart
predictive dialler based on asterisk  named ictbroadcast , it provide real
time report of busy, answered, congestion , failed, no answer call
statistics of running campaign

HTTP://www.ictinnovations.com/ictbroadcast

Regards

On 19-Sep-2011 7:13 PM, "mahesh katta"  wrote:
> Hi List,
>
> I have one query, I am using Go autodial in this using auto dialing.
> autodial can do only whenever customer pick the call that call will go to
> agents.
> but problem autodial dialing the database in that I am not getting NC data
> means, not reachable,switch off ,outofservice data. how can I get this
data.
> is there any software get the telcovoice and give report ?
>
> Best Regards,
>
> Mahesh Katta
> *BUZZ**WORKS* Business Services Private Limited
> BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
> 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
(E)
> Mumbai 400069
> GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
> Web http://www.buzzworks.com
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Re: [asterisk-users] oddity with CISCO CCM and Asterisk

[Sam Govind]

Hi Danny,
If you explain some more about this phantom process !! I've never seen
asterisks doing this before. This initiation of a new call is always
dependent upon arrival of an INVITE. I doubt its CCM that is doing some
re-INVITES or sort of keepalive for this call and thus a phantom call is
created !



On Mon, Sep 19, 2011 at 10:58 PM, Danny Nicholas  wrote:

> Hi List,
>
>  I have a system that connects into Asterisk 1.4.41 using CISCO
> CCM 7.  Everything works great except when a call is transferred to the
> operator.  The call goes to the operator via a native bridge and is
> completed, then a “phantom process” starts and tries to launch a new call
> every 15 minutes.  I modified the dialplan to hangup these phantom calls,
> but no still joy.  I get this message:
>
> [Sep 19 10:32:47] NOTICE[14249] chan_sip.c: Call from 'XXX' to extension
> 'X' rejected because extension not found.  
>
> ** **
>
> 14249 is not showing as running, but it is accessible vi pmap 14249
>
> ** **
>
> Could this be a problem with pbx_loopback?
>
> ** **
>
> Thanks
>
> Danny Nicholas
>
> ** **
>
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Re: [asterisk-users] Ghost DID in System

[C F]

Without your dialplan there isnt much that can be done to help.
Can you please post your relevant dialplans?
Whats voip1 and voip2?
When you say outside the voip system call goes thru, to where?
Who has the number currently?
Any sip debug you care sharing?


On Mon, Sep 19, 2011 at 6:51 PM, Aaron Krohn  wrote:
> This is going to sound ridiculous, but there appears to be a ghost DID in
> our system. We are going to get the number ported to us, but it has not
> happened yet. From a phone outside of our voip system, the call still goes
> through. When calling the did from a phone within our system, there is just
> dead air.
>
> In the asterisk CLI, I can see our primary server, voip1 trying to do pass
> the call to voip2 after it complains about not knowing what to do with the
> call.
>
> I have removed all references to this number from all dialplans and sip-did
> lists and restarted many times. I simply don't understand why our voip1
> server believes it should try to route the call instead of passing it to the
> outside world. Does anyone have an explanation or know where I could look?
> (dialplans, obviously =)
>
>
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Re: [asterisk-users] NC DATA FINDOUT IN AUTO DIALER

[mahesh katta]

Thanks for reply,
I had check it. in auto dialer whenever dial the number there is no voice to
get agent. dialer will dial the number asterisk not getting voice like
swo,NA. how we can get the voice in there.
Best Regards,

Mahesh Katta
*BUZZ**WORKS* Business Services Private Limited
BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri (E)
Mumbai 400069
GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
Web http://www.buzzworks.com



On Mon, Sep 19, 2011 at 6:23 PM, Nasir Iqbal wrote:

> Please check our voice sms and fax broadcasting / smart autodialler / smart
> predictive dialler based on asterisk  named ictbroadcast , it provide real
> time report of busy, answered, congestion , failed, no answer call
> statistics of running campaign
>
> HTTP://www.ictinnovations.com/ictbroadcast
>
> Regards
>
> On 19-Sep-2011 7:13 PM, "mahesh katta"  wrote:
> > Hi List,
> >
> > I have one query, I am using Go autodial in this using auto dialing.
> > autodial can do only whenever customer pick the call that call will go to
> > agents.
> > but problem autodial dialing the database in that I am not getting NC
> data
> > means, not reachable,switch off ,outofservice data. how can I get this
> data.
> > is there any software get the telcovoice and give report ?
> >
> > Best Regards,
> >
> > Mahesh Katta
> > *BUZZ**WORKS* Business Services Private Limited
>
> > BANGALORE | CHENNAI | HYDERABAD | MUMBAI| DELHI
> > 201, Crystal Tower, 75 Gundavali Cross Lane, Andheri Kurla Road, Andheri
> (E)
> > Mumbai 400069
> > GSM +91.97029.70779 | Phone +91.22.4229.2634 | Fax +91.22.4229.2634
> > Web http://www.buzzworks.com
>
>
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