Hello,
I need to setup system of aroud 60 DECT phones with asterisk.
So far I found
http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710
However is there some cheap base station(similar to GSM cell) so I can
handle all DECT phones centralized and plug
Hi,
I have to create scenario like following,
I have 2 sip soft phone.I configured Asterisk server on local network, on
Linux.With two soft-phone , local asterisk sever, i able to communicate.Now i
have communicate with other network SIP client.For that i have opened account
at @sip2sip.info,
2013/1/21 Mitch Claborn mitch...@claborn.net
Asterisk 11
Occasionally we will have a partial power outage, or a piece of network
equipment will fail, and our queue agents who are on active calls with
callers will be disconnected from the caller. What I'd like to do is
capture those calls
And how would you have this working together with Asterisk queueing? I have
seen solutions like this using agent pauses and then making everyithing
happen outside the normal ACD flow, but it's a bit of a hack
l.
2013/1/22 Danny Nicholas da...@debsinc.com
For just the messaging part, you
Hi,
I am using:
Asterisk 11.2.0
libpri 1.4.12
Dahdi: 2.6.1
Sangoma E1-Card with Wanpipe-Drivers 3.5.28
I call my asterisk box via SIP and connect the call to an AGI-Script.
Within the script I do
EXEC SetCallerPres prohib
or
EXEC SetCallerPres prohib_not_screened
But I get the following
Simplest question first. Does it show up in core show applications or
core show application SetCallerPres? If not, do a make menuselect and see
if something broke in the ability to make the application.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Thanks! It is not activated. Also I found a comment there:
Support Level: deprecated, Replaced by: func_callerid
So I use this instead.
Am 24.01.2013 15:33, schrieb Danny Nicholas:
Simplest question first. Does it show up in core show applications or
core show application SetCallerPres? If
Am 23.01.2013 um 18:33 schrieb Carlos Alvarez:
On Wed, Jan 23, 2013 at 10:20 AM, Sebastian Arcus s...@open-t.co.uk wrote:
I have an Asterisk server with one SIP trunk to a SIP provider. As my server
registers with the SIP provider, I don't have any SIP ports open at my end to
the Internet.
Dear;
Using Cisco IP Phones: How I can assign a button for a function. For example,
if we pressed on this button, then we need to pickup the call from the group.
Another thing:
If the button pressed, then the call forward function to be enabled (and it
should appear on the phone that it is
Yes it might be “hacky”, but anything that isn’t somewhat is going to come at a
premium price. Today’s motto is “get her done as quick and cheap as possible”.
It is a luxury to have a well-trained, professional staff providing solid
solutions when folks want Top Quality at slave wage labor
On Thu, Jan 24, 2013 at 8:03 AM, bilal ghayyad bilmar...@yahoo.com wrote:
Dear;
Using Cisco IP Phones: How I can assign a button for a function. For
example, if we pressed on this button, then we need to pickup the call from
the group.
Which model line? The SPA series, or the 7900 and
They advised me to check jabber.org.
Yes, jabber.org has a client that can send/receive and integrate with other
social media (facebook, msn, twitter, ... etc).
But, as an Agent who can login/logout and take a calls, how can I make it to be
single login for voice and messages. So, if the agent
This is how I would see the process working
1. use curl/wget to query Facebook (etc.)
2. determine whether we are to drop a call into the queue or just process a
message
3. determine agent availability through AMI process or asterisk -rx
process.
4. drop the call into the queue or place the
It appears that there are no transcoders from g723 to anything else in
Asterisk 10.7.1. Does anybody know how to fix that?
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Hi
Look at it this link
http://asterisk.hosting.lv/
Kind Regards
On Thu, Jan 24, 2013 at 10:34 AM, Richard Kenner ken...@gnat.com wrote:
It appears that there are no transcoders from g723 to anything else in
Asterisk 10.7.1. Does anybody know how to fix that?
--
Hi,
We before, used facebook graph api (json) on a php script.
php would check new posts every minute, and write a new .call file into
asterisk, with a sort of TTS
call would go on queue, and once a member picks it up, he hears 'new
facebook call from, bla bla, stating bla bla bla'
He would then
When I am monitoring the AMI I see the following event
for a call I just made over a SIP trunk.
Event: Newchannel
Privilege: call,all
Channel: SIP/testmachine-000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
AccountCode:
Exten:
Context: testmachine
Uniqueid:
Have you tried and looked up all events generated when you place the call?
some of them are bound to have the variable callerid set
On 24 January 2013 16:46, Jerry Geis ge...@pagestation.com wrote:
When I am monitoring the AMI I see the following event
for a call I just made over a SIP
On Thu, Jan 24, 2013 at 10:40:28AM -0500, Carlos Rojas wrote:
Hi
Look at it this link
http://asterisk.hosting.lv/
Kind Regards
On Thu, Jan 24, 2013 at 10:34 AM, Richard Kenner ken...@gnat.com wrote:
It appears that there are no transcoders from g723 to anything else in
Not the greatest solution, but since you are most likely using a script for the
AMI process, you could do an
Asterisk –rx “core show channels verbose”|grep SIP/testmachine-000d
And get the dialed number from that.
Actually you could issue the AMI command core show channels verbose.
Have you tried and looked up all events generated when you place the call?
some of them are bound to have the variable callerid set
yes I have looked at all of them, CallerID is not set to the number I am
calling.
Jerry
Not the greatest solution, but since you are most likely using a script for the
AMI process, you could do an
Asterisk --rx core show channels verbose|grep SIP/testmachine-000d
And get the dialed number from that.
Actually you could issue the AMI command core show channels verbose.
there
Both: SPA and 7900. let us say 7942. How?
Regards
Bilal
Dear;
Using Cisco IP Phones: How I can assign a button for a
function. For
example, if we pressed on this button, then we need to
pickup the call from
the group.
Which model line? The SPA series, or the 7900 and
On Thu, Jan 24, 2013 at 12:11 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Both: SPA and 7900. let us say 7942. How?
Googled cisco 7942 soft keys, first result:
http://www.cisco.com/en/US/docs/voice_ip_comm/cucme/admin/configuration/guide/cmesoftk.html
This is pretty off-topic, by the way.
On 01/24/2013 10:46 AM, Jerry Geis wrote:
When I am monitoring the AMI I see the following event
for a call I just made over a SIP trunk.
Event: Newchannel
Privilege: call,all
Channel: SIP/testmachine-000d
ChannelState: 0
ChannelStateDesc: Down
CallerIDNum:
CallerIDName:
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
Note that the Channel: field will contain the name initiating the Dial,
the Destination: field will
This might have changed but IIRC /etc/asterisk/manager.conf controls what
events you have access to.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Jerry Geis
Sent: Thursday, January 24, 2013 1:14 PM
To: Asterisk Users Mailing List -
On 01/24/2013 10:37 AM, Zyumbilev, Peter wrote:
Hello,
I need to setup system of aroud 60 DECT phones with asterisk.
So far I found
http://www.grandstream.com/index.php/products/ip-voice-telephony/enterprise-ip-phones/dp715_710
However is there some cheap base station(similar to GSM cell) so
On Thu, Jan 24, 2013 at 12:52 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
Polycom also has DECT stuff. I doubt it will come cheap.
http://spectralink.polycom.**com/dect_communications/index.**htmlhttp://spectralink.polycom.com/dect_communications/index.html
Not cheap, but this
I second what Carlos said. The Spectralink is the quality and reliability
you want.
On Jan 24, 2013 2:06 PM, Carlos Alvarez car...@televolve.com wrote:
On Thu, Jan 24, 2013 at 12:52 PM, Patrick Lists
asterisk-l...@puzzled.xs4all.nl wrote:
Polycom also has DECT stuff. I doubt it will come
I'm trying to interface Asterisk with an Alcatel PABX and trying to find
a code that works well. It says it doesn't support ulaw, though it
doesn't reject it. It supports G.729, and that works fine, but we'd prefer
not to use compression.
When I use alaw, the path from Asterisk to the Alcatel
Your sounds might be too loud. We use a lot of custom sounds here and when
the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and
clicks.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Your sounds might be too loud. We use a lot of custom sounds here and when
the volume approaches 0 db (asterisk standard is -3 db) we get fuzz and
clicks.
Sorry I wasn't clear. This is *always*. I hear it over the call when
there's talking and when there's dead silence (e.g., an empty
I am curious, is your version of asterisk correctly compiling the regserver
field? Each server needs to have a distinct server name.
Upgrading to the latest version didn't help. After about 30 minutes, Asterisk2
tries to send out OPTIONS keepalive packets to peers listed as Registered on
2013/1/24 Dan Journo d...@keshercommunications.com
I am curious, is your version of asterisk correctly compiling the
regserver field? Each server needs to have a distinct server name.
** **
Upgrading to the latest version didn't help. After about 30 minutes,
Asterisk2 tries to send
On 01/24/2013 09:44 PM, Richard Kenner wrote:
[snip]
When I use alaw, the path from Asterisk to the Alcatel is completely
clean, but the other way has a set of clicks that kind of sound like
old-fashioned audio noise.
[snip]
It's been ages since I experienced that but things to check that come
When I use alaw, the path from Asterisk to the Alcatel is completely
clean, but the other way has a set of clicks that kind of sound like
old-fashioned audio noise.
[snip]
It's been ages since I experienced that but things to check that come to
mind in no particular order are:
On 22/01/2013, at 5:27 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Dears;
Can someone advise me where to find a technology (open source) that let us
able to integrate with social media like whatsapp and facebook? And use this
in call center (queuing the messages and routing it for agent)?
On 24/01/2013, at 10:24 AM, bilal ghayyad bilmar...@yahoo.com wrote:
They advised me to check jabber.org.
Yes, jabber.org has a client that can send/receive and integrate with other
social media (facebook, msn, twitter, ... etc).
But, as an Agent who can login/logout and take a calls, how
Hi Richard,
the macro you linked to did the trick for me - thank you!
Greetings from Wuppertal
Max Grobecker
Am 24.01.2013 00:18, schrieb Richard Mudgett:
- Original Message -
Hello out there,
I'm running an Asterisk 1.8.15-cert1 with DAHDI.
Today I noticed that Asterisk is
- jitterbuffer settings (try on/off)
I added
jbenable=yes
and get lots of:
[Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-6c7 received frame with invalid timing info:
has_timing_info=1, len=0, ts=371371424, src=RTP
[Jan 24 17:53:41] WARNING[12317]:
On 01/24/2013 01:13 PM, Jerry Geis wrote:
You probably want the Dial event. It is raised both at the beginning of
the Dial, as well as when the Dial completes.
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+ManagerEvent_Dial
Note that the Channel: field will contain the name
On 01/24/2013 11:57 PM, Richard Kenner wrote:
- jitterbuffer settings (try on/off)
I added
jbenable=yes
and get lots of:
[Jan 24 17:53:41] WARNING[12317]: abstract_jb.c:284 ast_jb_put:
DAHDI/i1/2128518396-6c7 received frame with invalid timing info:
has_timing_info=1, len=0,
Check https://issues.asterisk.org/jira/browse/ASTERISK-12042
I did. But that was with an unofficial G.729. This is with the supplied
alaw codec.
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New
- Original Message -
From: Bryant Zimmerman brya...@zktech.com
Hey all
For some reason the mailing list is sending all messages from the
sending party.
This makes it less than ideal when responding; as selecting reply
goes to the person and not the list.
Can we have it set back
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