On Mon, May 16, 2011 at 10:20 PM, Shaun Ruffell sruff...@digium.com wrote:
On Mon, May 16, 2011 at 09:26:48PM -0400, A E [Gmail] wrote:
following this advice, is there a quick and minimal way to install/use
res_timing_dahdi without having to build/compile/install the whole dahdi
package
Hello All;
If I need the Asterisk to do automatic dialing for a list of numbers and when
the destination answer, then to play the proper sound message, is it possible?
How?
About sending SMS, can asterisk do this?
Regards
Bilal
--
Thanks Alex for ur help and advise.
In case we decided to do a script for this reporting, we will depend on the
logs or we need to use the AGI?
In case we will do a dashboard to display how many agents are login and how
many calls in the queue and how many calls in specific skill group?
I would think that that is down to either your indications.conf (could
be wrong) or the handset itself.
I know most Yealink and GrandStream handsets let you change tones in
their individual config. Not too sure about others.
-Original Message-
From:
And why would you post a reply 5 days after my last post - and 4 days
after the threads last one?
Do you want to keep this thread going?
I suggest letting it die on it's own.
_
-Original Message-
From: asterisk-users-boun...@lists.digium.com
is there a simple way to receive a response from an action such as a Ping
?
On Mon, May 16, 2011 at 4:21 PM, Alex Balashov abalas...@evaristesys.comwrote:
On 05/16/2011 04:17 PM, vip killa wrote:
forgive me for i am very new to asterisk and perl. but how could you
detect if you were
Hi,
Which XMPP server would you recommend to host Jingle-enabled XMPP service ?
This Jingle-service would target a small audience (20 XMPP users and 5
simultaneous calls).
Regards
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On 04/17/2011 02:13 AM, Stefan Gofferje wrote:
has anyone managed to establish an XMPP connection to the facebook
Jabber servers?
I'd like to send messages on missed calls vie FB.
I finally figured it out.
For facebook chat to work you have to use
I use Openfire.
regards
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On Sat, May 14, 2011 at 05:40:52PM -0700, Steve Edwards wrote:
On Sun, 15 May 2011, Hans Witvliet wrote:
It's a bit more complicated
after the last rules, it is handy to put:
$iptables -A INPUT -i $EXTERNAL_DEV -j LOG --log-prefix EXT; INC
iptables -A OUTPUT -o $EXTERNAL_DEV -j LOG
Hi List,
How to put multiple call on hold by dialplan in asterisk?
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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New
Hello, all.
Could you tell me how to set the type of number in the outgoing SETUP
message sent over PRI trunk?
I need to have:
Called Number (len=18) [ Ext: 1 TON: Unknown Number Type (0) NPI:
ISDN/Telephony Numbering Plan (E.164/E.163) (1) '318989263037666' ]
But always have:
Called
hi list,
please help me how to know how many calls are on hold.
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Thanks and regards
Virendra Bhati
+91-9172341457
Asterisk Engineer
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New
Hi Guys,
I am getting an error when executing another mysql query in dialplan after
calling stored procedure.
If calling the procedure from mysql cli it gives a result like:
mysql call call_control(78236721,1000,1233);
+--+
| pass |
+--+
|1 |
+--+
So I need asterisk to recognize
Hi, my log is full of errors from this mobile user:
-- Registered SIP '0010106' at 212.93.97.135:7759
[2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804
handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms /
1ms)
[2011-05-17 17:44:06] ERROR[21456]: netsock2.c:245
I was debugging a turnup with Global Crossing the other day and they
presented me with a web page that displayed a 'ladder diagram' of a call
including a ton of detail all neatly organized in tabs and links so you
could drill down to any level of detail needed.
The copyright notice says
Hi,
Is there any softphone or TAPI plug-in that allows one to dial from a web
page? As you may know, Skype has a mechanism that converts phone numbers on
a web page to a click-to-dial application. I'd like to use this but on a
normal softphone (Bria, Xlite, other).
Regards,
Mike
--
On Tue, May 17, 2011 at 01:30:33PM -0400, Mike wrote:
Is there any softphone or TAPI plug-in that allows one to dial from a web
page? As you may know, Skype has a mechanism that converts phone numbers on
a web page to a click-to-dial application. I'd like to use this but on a
normal softphone
On Tue, 17 May 2011, Mike wrote:
Is there any softphone or TAPI plug-in that allows one to dial from a
web page? As you may know, Skype has a mechanism that converts phone
numbers on a web page to a click-to-dial application. I’d like to use
this but on a normal softphone (Bria, Xlite,
Exactly what I was looking for. Thank you.
Mike
Is there any softphone or TAPI plug-in that allows one to dial from a
web page? As you may know, Skype has a mechanism that converts phone
numbers on a web page to a click-to-dial application. I?d like to use
this but on a normal
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
I was debugging a turnup with Global Crossing the other day and they
presented me with a web page that displayed a 'ladder diagram' of a call
including a ton of detail all neatly organized in tabs and links so you
On 11-05-17 11:46 AM, Nick Ustinov wrote:
Hi, my log is full of errors from this mobile user:
-- Registered SIP '0010106' at 212.93.97.135:7759
[2011-05-17 17:44:05] NOTICE[21456]: chan_sip.c:19804
handle_response_peerpoke: Peer '0010106' is now Reachable. (381ms /
1ms)
[2011-05-17
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
I was debugging a turnup with Global Crossing the other day and they
presented me with a web page that displayed a 'ladder diagram' of a
call including a ton of detail all neatly organized in tabs and links
so
On Tue, May 17, 2011 at 2:32 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
I was debugging a turnup with Global Crossing the other day and they
presented me with a web page that displayed a 'ladder diagram'
- Original Message -
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
I was debugging a turnup with Global Crossing the other day and
they
presented me with a web page that displayed a 'ladder diagram' of a
call including a ton of detail all
I am using asterisk 1.4.41 and the AMI
I am trying to execute a command over AMI, specifically core show
channels concise
sometimes I get this back:
asterisk_command_show_channels() execute failed. 'Response: Follows[CR
][LF ]Privilege: Command[CR ][LF
Jerry Geis wrote:
I am using asterisk 1.4.41 and the AMI
I am trying to execute a command over AMI, specifically core show
channels concise
sometimes I get this back:
asterisk_command_show_channels() execute failed. 'Response: Follows[CR
][LF ]Privilege: Command[CR ][LF
Hey Guys!
Sorry i am posting scripting question in asterisk forum but i had no choice.
also i am not script expert so i though anyone here might help me.
following is my example sip.conf now i want to add
accountcode=callerid_name for example accountcode=Katie Wilson in
entire file. we
On Tue, May 17, 2011 at 4:21 PM, satish patel satish...@hotmail.com wrote:
Hey Guys!
Sorry i am posting scripting question in asterisk forum but i had no
choice. also i am not script expert so i though anyone here might help me.
following is my example sip.conf now i want to add
Holy cow! you made my day
Thank you so much... It works great!!!
S.
From: mden...@gmail.com
Date: Tue, 17 May 2011 17:02:55 -0400
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] script to trim sip.conf
On Tue, May 17, 2011 at 4:21 PM, satish patel satish...@hotmail.com
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
I was debugging a turnup with Global Crossing the other day and they
presented me with a web page that displayed a 'ladder diagram' of a call
including a ton of detail all neatly organized in tabs and links so you
In article alpine.DEB.2.00.1105171432580.30550@localhost.localdomain,
Steve Edwards asterisk@sedwards.com wrote:
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
I was debugging a turnup with Global Crossing the other day and they
presented me with a
On Tue, May 17, 2011 at 1:12 PM, Steve Edwards
asterisk@sedwards.com wrote:
I was debugging a turnup with Global Crossing the other day and they
presented me with a web page that displayed a 'ladder diagram' of a call
including a ton of detail all neatly organized in tabs and links so you
Hello folks,
I have a litle problem here, i did not run into this problem yet, but it
will be in the future!
Let think!
I have 8 agents, 2 of them speak english and Portuguese.
Now i have many clients most of them speaking english... And we send all
users to the queue app. Now lets thinks all
Hi,
You would have to setup two queues, one English and one portuguese. All
agents are in the English queue, but only 2 are also in the Portuguese
queue. Give the Portuguese queue a higher Weight value and voilà (I think)
Mike
From: asterisk-users-boun...@lists.digium.com
We are having a problem where agents are not logging off at the end of
the day and they complain about receiving calls early the next day. Is
there a simple way to automatically log off all agents (dynamic) from
all queues at a certain hour? Or do I have to parse all queues for
agents
I thougth at same thing, but as you said i think i had the same doubt!
But thanks!
as i said, we ar enot there in the project yet, so it will be an while still
get there.. but i asked to see if it was possible cause it is an pre
requisit!
[]'sf.rique
On Tue, May 17, 2011 at 8:58 PM, Mike
On Tue, May 17, 2011 at 7:30 PM, Mike l...@net-wall.com wrote:
Is there any softphone or TAPI plug-in that allows one to dial from a web
page? As you may know, Skype has a mechanism that converts phone numbers on
a web page to a click-to-dial application. I’d like to use this but on a
normal
Hi all, does the asterisk 1.4.x support TLS and SRTP?
Thanks
--
havesoftware, Inc.
http://www.havesoftware.com
Jakson Kalsson
Senior Programmer
jakkals...@havesoftware.com
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2011/5/18 Jakson Kalsson sipmaill...@gmail.com
Hi all, does the asterisk 1.4.x support TLS and SRTP?
no
Thanks
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havesoftware, Inc.
http://www.havesoftware.com
Jakson Kalsson
Senior Programmer
jakkals...@havesoftware.com
--
Thank you, I saw the 1.8 support asterisk ?
On Wed, May 18, 2011 at 1:31 PM, Olivier oza_4...@yahoo.fr wrote:
2011/5/18 Jakson Kalsson sipmaill...@gmail.com
Hi all, does the asterisk 1.4.x support TLS and SRTP?
no
Thanks
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http://www.havesoftware.com
Jakson
2011/5/18 Jakson Kalsson sipmaill...@gmail.com
Thank you, I saw the 1.8 support asterisk ?
yes, you need 1.8 for SRTP and TLS
regards
On Wed, May 18, 2011 at 1:31 PM, Olivier oza_4...@yahoo.fr wrote:
2011/5/18 Jakson Kalsson sipmaill...@gmail.com
Hi all, does the asterisk 1.4.x
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