On Tuesday 23 of October 2012 22:32:12 Liutauras Adomaitis wrote:
Hi all,
I am having problems configuring followme with realtime ldap database. The
error I get is:
[2012-10-23 21:01:40] WARNING[16004]: res_config_ldap.c:967
realtime_multi_ldap: realtime retrieval requires at least 1
Just upgraded to 1.8, we use the multi lot parking feature by dialling *4.
We are not getting the parking slot announcement being played to the person
who parks the call, so it's impossible to tell which slot they've gone
into. Could someone check our config?
On Debian Squeeze using packages from
Hi,
I've got a ISDN Interface: *Tiger Jet* Network Inc. Tiger3XX Modem/ISDN
interface, I'm trying to use it with DAHDI 2.6 but it doesn't work, I'm
thinking that dahdi doesn't support this device, I've loaded all of
available dahdi drivers and none of them worked. Does anybody know what I
can do
Hello everyone!
We use Asterisk for various services like voicemail. Our SIP clients
usually use rtp events (rfc2833) for DTMF, which works just fine and
independent from the codec (g711 vs. g726 etc.).
Now we noticed there are some SIP clients that announce telephone-event
in their SDP, but
Asterisk 1.8.10.1~dfsg-1ubuntu1
See dial plan code below. When I dial 123 from a phone in this context,
I simply get a busy signal. Why doesn't the i extension get
triggered? Console at verbosity of 10 only shows == Using SIP RTP
CoS mark 5.
[DockPhone]
exten =288,1,NoOp(Dock Phone)
exten =i,1,NoOp(invalid extension from dock phone i)
Was this a typo?
I believe it should be:
exten = i,1,NoOP()
What does your console output look like?
Doug
--
Ben Franklin quote:
Those who would give up Essential Liberty to purchase a little Temporary
Safety, deserve neither Liberty
It would be good to see OP's output, but noop() is essentially the same as
Verbose(), whatever goes in the () is just a comment/message.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent:
Asterisk 1.8.10.1~dfsg-1ubuntu1
See dial plan code below. When I dial 123 from a phone in this
context,
I simply get a busy signal. Why doesn't the i extension get
triggered? Console at verbosity of 10 only shows == Using SIP RTP
CoS mark 5.
[DockPhone]
exten =288,1,NoOp(Dock Phone)
On Thursday 25 October 2012, Mitch Claborn wrote:
Asterisk 1.8.10.1~dfsg-1ubuntu1
See dial plan code below. When I dial 123 from a phone in this context,
I simply get a busy signal. Why doesn't the i extension get
triggered? Console at verbosity of 10 only shows == Using SIP RTP
CoS mark
I set logger.conf to
console =debug,notice,warning,error,verbose
and get the following output:
== Using SIP RTP CoS mark 5
[Oct 25 10:32:53] NOTICE[3501]: chan_sip.c:22622 handle_request_invite:
Call from 'Mitch295' (192.168.5.104:5060) to extension '123' rejected
because extension not
Based on the output below, DockPhone is expecting to be reached with a
dialstring of 444. If you change 444 to ZXX, the problem should go away.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mitch Claborn
That does sound quite suspicious.
Mitch
It looks like you are seeing this issue that was fixed earlier
this month:
https://issues.asterisk.org/jira/browse/ASTERISK-20455
Richard
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On Thu, Oct 25, 2012 at 10:42 AM, Danny Nicholas da...@debsinc.com wrote:
Based on the output below, DockPhone is expecting to be reached with a
dialstring of 444. If you change 444 to ZXX, the problem should go away.
The point is that he's trying to trigger the invalid extension dialplan.
A little more background will help. This is a phone that will be
outside on our receiving dock. When a driver lifts the handset, the
ObiTalk 110 dials 444 automatically. That all works fine and it rings
the phones that it should.
What I'm trying to do with the i extension is give a
Our phone operators work off of an Asterisk queue. They take calls from
customers and take orders with our back end systems. What I need to be
able to do is tie the orders taken to the specific CDR record that
reflects the call from which the order originated.
The typical/sample CDR table
You have the uniqueID, which is a pseudo timestamp. More useful to your
described effort, though would be the answer and end of call fields. Your
backend system is going to have the timestamp of when the order was placed,
so you just need to address the calls that sandwich that timestamp.
In article 50895cab.9080...@claborn.net,
Mitch Claborn mitch...@claborn.net wrote:
I set logger.conf to
console =debug,notice,warning,error,verbose
and get the following output:
== Using SIP RTP CoS mark 5
[Oct 25 10:32:53] NOTICE[3501]: chan_sip.c:22622 handle_request_invite:
Call
On Thu, Oct 25, 2012 at 11:24 AM, Tony Mountifield t...@softins.co.ukwrote:
The 'i' extension is not used when entering a context. You can only enter
a context (with Dial(), Goto(), etc), at an extension that exists. If it
doesn't exist, the context cannot be entered.
So it sounds like what
No, it isn't.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Christopher
Harrington
Sent: Thursday, October 25, 2012 12:29 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
BOP! You don't need no stinkin I in this case! Just put this in front of
the Dial()
Exten = 444,2,Gotoif(${DOCK_RECIPIENTS} != 444]?i,1)
This catches anything they dial that isn't the magic 444.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Thanks Tony, this helps.
Mitch
On 10/25/2012 11:24 AM, Tony Mountifield wrote:
The 'i' extension is not used when entering a context. You can only enter
a context (with Dial(), Goto(), etc), at an extension that exists. If it
doesn't exist, the context cannot be entered.
The 'i' extension
DOCK_RECIPIENTS is a long list of 5+ SIP phones, so this won't work.
Mitch
On 10/25/2012 11:31 AM, Danny Nicholas wrote:
BOP! You don't need no stinkin I in this case! Just put this in front of
the Dial()
Exten = 444,2,Gotoif(${DOCK_RECIPIENTS} != 444]?i,1)
This catches anything they dial
My wife and I have a home telephone/answering machine. We've decided that
having voicemail stuck in the machine and waiting for us to return home is not
working for us any longer. We'd like for calls coming into our home to be
routed to our cell phones. It seems to me that Asterisk might be
On Thu, Oct 25, 2012 at 11:09:01AM -0700, Matthew Hixson wrote:
- Is the Linksys SPA3102 a good piece of hardware for this type of setup or
is there something cheaper? Perhaps a card that can go right into the Linux
box?
I'm using an OpenVox A400 (with an FXO module), which Asterisk can
On Oct 25, 2012, at 11:22 AM, Roger Burton West ro...@firedrake.org wrote:
On Thu, Oct 25, 2012 at 11:09:01AM -0700, Matthew Hixson wrote:
- Is the Linksys SPA3102 a good piece of hardware for this type of setup or
is there something cheaper? Perhaps a card that can go right into the Linux
On Thu, Oct 25, 2012 at 11:33:06AM -0700, Matthew Hixson wrote:
Is there any reason a regular old voicemodem wouldn't work?
IME the voice quality and reliability are pretty grotty. If you find
one that works, great!
R
--
_
--
Danny - good idea. That works for the first report that I'm creating.
Another idea I had that I may explore:
Create another table keytable with an auto increment PK. When I place
the call in the queue, insert a row into keytable and retrieve the
generated PK. Put that value into the CDR as
I'm looking for an fxs - sip gateway/router/switch for about 100 existing
analog phones. I'd like to get this done cheaply, but I want to make sure that
whatever we buy works well with asterisk as well. As far as I can tell, digium
make no such device. The only ones I've been able to find
On 10/25/2012 11:18 AM, Mitch Claborn wrote:
Our phone operators work off of an Asterisk queue. They take calls from
customers and take orders with our back end systems. What I need to be
able to do is tie the orders taken to the specific CDR record that
reflects the call from which the order
On 10/25/2012 04:21 PM, Justin Killen wrote:
just talking in general terms here I have found this sort of hardware is
not the most reliable, and the more physical devices you spread it
across the more fault tolerant you are of a single fault taking down a
big chunk of your users.
I wouldn't
Or you could use the followme feature to have asterisk just call your
cell phones.
Roger Burton West ro...@firedrake.org wrote:
On Thu, Oct 25, 2012 at 11:09:01AM -0700, Matthew Hixson wrote:
- Is the Linksys SPA3102 a good piece of hardware for this type of setup or
is there something
Agree with 24 port being the max for a single device. In that vein I
just deployed a handful of Grandstream 24 port FXS devices that seem to
be working well at a decent price point. I don't normally recommend
Grandstream for anything, and in the past we have only deployed
Audiocodes for
What would be the advantage of using 100 single units vs. just buying VoIP
phones? That doesn't seem very cost effective to me in the long run.
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to
On Thu, Oct 25, 2012 at 1:21 PM, Justin Killen
jkil...@allamericanasphalt.com wrote:
I’m looking for an fxs - sip gateway/router/switch for about 100
existing analog phones. I’d like to get this done cheaply, but I want to
make sure that whatever we buy works well with asterisk as well. As
Cost and ease of deployment, yes. At this specifc location we are currently
using Centrex lines (ATT hosted) and are looking for a way to move into
something cheaper without throwing away the existing phones. I like the idea
of using a channel bank - I'll look into that as an option as well.
That is just silly. You mean to say that the Adtran and the Adit
units are not as reliable as these new devices. No way.
Get Adtrans, buy a four port T1 card or even better get the redfone
device and do HA Linux between to boxes, you have immediate failover.
On 25/10/12 9:49 pm, Justin Killen wrote:
What would be the advantage of using 100 single units vs. just buying VoIP
phones? That doesn't seem very cost effective to me in the long run.
In older buildings with existing single pair cabling, there might not be
a great deal of choice.
We
On Thu, Oct 25, 2012 at 5:01 PM, Chris Bagnall
aster...@lists.minotaur.cc wrote:
On 25/10/12 9:49 pm, Justin Killen wrote:
What would be the advantage of using 100 single units vs. just buying VoIP
phones? That doesn't seem very cost effective to me in the long run.
In older buildings with
On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen
jkil...@allamericanasphalt.com wrote:
**
Cost and ease of deployment, yes. At this specifc location we are
currently using Centrex lines (ATT hosted) and are looking for a way to
move into something cheaper without throwing away the existing
On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote:
I always advocate throwing out old analog phones as they will be a pain,
but understand if you absolutely cannot. Just keep in mind you can get a
decent VoIP phone for $60 that is very likely to be nicer than what they
On Thu, Oct 25, 2012 at 2:14 PM, Christopher Harrington ch...@acsdi.comwrote:
On Thu, Oct 25, 2012 at 4:09 PM, Carlos Alvarez car...@televolve.comwrote:
I always advocate throwing out old analog phones as they will be a pain,
but understand if you absolutely cannot. Just keep in mind you can
On Thu, Oct 25, 2012 at 12:18 PM, Mitch Claborn mitch...@claborn.net wrote:
Our phone operators work off of an Asterisk queue. They take calls from
customers and take orders with our back end systems. What I need to be able
to do is tie the orders taken to the specific CDR record that
yealink T18 and T20 are decent phones available for $60
Regards,
Mitul Limbani,
Chief Architech Founder,
Enterux Solutions Pvt. Ltd.
110 Reena Complex, Opp. Nathani Steel,
Vidyavihar (W), Mumbai - 400 086. India
http://www.enterux.com/
http://www.entvoice.com/
email: mi...@enterux.in
DID:
I think if we were to go to VoIP phones, one thing that we would have to
consider very highly in a phone would be that they have VLAN settings and a
built-in Ethernet hub/switch so that we can just inject it into the user's
computer LAN connection. The cost and time of rewiring some of these
On Thu, Oct 25, 2012 at 2:34 PM, Justin Killen
jkil...@allamericanasphalt.com wrote:
** ** **
I think if we were to go to VoIP phones, one thing that we would have to
consider very highly in a phone would be that they have VLAN settings and a
built-in Ethernet hub/switch so that we can just
On 10/25/2012 05:09 PM, Carlos Alvarez wrote:
On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen
jkil...@allamericanasphalt.com
mailto:jkil...@allamericanasphalt.com wrote:
Cost and ease of deployment, yes. At this specifc location we are
currently using Centrex lines (ATT hosted) and
On 10/25/2012 05:01 PM, Steve Totaro wrote:
That is just silly. You mean to say that the Adtran and the Adit
units are not as reliable as these new devices. No way.
I have had channel banks fail yes, and I stick by my assertion that
failing a small $50 box is a lot less painful on the
On Thu, Oct 25, 2012 at 5:35 PM, jon pounder j...@inline.net wrote:
On 10/25/2012 05:09 PM, Carlos Alvarez wrote:
On Thu, Oct 25, 2012 at 2:01 PM, Justin Killen
jkil...@allamericanasphalt.com wrote:
Cost and ease of deployment, yes. At this specifc location we are
currently using Centrex
On 26/10/2012, at 10:09 AM, jon pounder j...@inline.net wrote:
On 10/25/2012 05:01 PM, Steve Totaro wrote:
That is just silly. You mean to say that the Adtran and the Adit
units are not as reliable as these new devices. No way.
I have had channel banks fail yes, and I stick by my
On Thu, Oct 25, 2012 at 5:09 PM, jon pounder j...@inline.net wrote:
On 10/25/2012 05:01 PM, Steve Totaro wrote:
That is just silly. You mean to say that the Adtran and the Adit
units are not as reliable as these new devices. No way.
I have had channel banks fail yes, and I stick by my
On 10/25/2012 01:21 PM, Justin Killen wrote:
I'm looking for an fxs- sip gateway/router/switch for about 100 existing
analog phones. I'd like to get this done cheaply, but I want to make sure that
whatever we buy works well with asterisk as well. As far as I can tell, digium make
no such
Chris,
Thanks for answering my message.
I'm currently using version 10.5.1. I included the error message on the
dial plan to show what errors I was displaying. The call goes through
after that error message is displayed. As soon as I hear the phone ring,
it drops my call on the calling phone
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