Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
exten =
Problem solved - It was missing username setting, I had assumed
fromuser would be used for authentication.
On Sat, Apr 14, 2012 at 9:30 PM, Ben WIlliams
bwilliams+aster...@jadeworld.com wrote:
This is a really simple problem that I just can't get to work. There
are two Asterisk servers with the
Le 15/04/2012 10:44, Olivier CALVANO a écrit :
Hi
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten =
Is it a good idea to use asterisk transcoding from G711 to iLBC or should I
find out any other solution not involving transcoding (f.e. using G.729 that is
supported in both sides). I'm worried about voice quality and trying to avoid
paying for G.729 licensing.
Anybody with experience or
On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
Is it a good idea to use asterisk transcoding from G711 to iLBC or
should I find out any other solution not involving transcoding (f.e.
using G.729 that is supported in both sides). I'm worried about voice
quality and trying to avoid paying
On 04/15/2012 07:26 PM, Patrick Lists wrote:
On 04/15/2012 01:15 PM, Gustavo Garcia Bernardo wrote:
Is it a good idea to use asterisk transcoding from G711 to iLBC or
should I find out any other solution not involving transcoding (f.e.
using G.729 that is supported in both sides). I'm worried
On Sat, Apr 14, 2012 at 7:55 PM, Joseph syscon...@gmail.com wrote:
I forgot to add:
clinic-amd*CLI iax2 show peers
Name/Username Host Mask Port Status
home_server (null) (D) 255.255.255.255 0
Unmonitored
iaxy-322/iaxy-3 (null)
Hi all,
I'm sorry for asking this newbie question, but I can't find any clue googling :(
I need to disable fax service and some others on startup on my elastix
machine. Which file should be edited? I was thinking
/etc/asterisk/modules.conf but can't find a entry that load the fax
service on
Add the line noload=app_fax.so to modules.conf (app_fax.so might not be
correct, do ls /usr/lib/asterisk/modules/*fax* to get the actual module on
your machine)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Ok, got the idea.
Thanks a lot, Danny :)
On 4/15/12, Danny Nicholas da...@debsinc.com wrote:
Add the line noload=app_fax.so to modules.conf (app_fax.so might not be
correct, do ls /usr/lib/asterisk/modules/*fax* to get the actual module on
your machine)
-Original Message-
From:
Hi,
I'm running asterisk 1.8.7.0
FreePBX 2.8.1
IP Phone Yealink T20
Trustrpid and sendrpid is on the sip.conf
Let say I pickup a call on ext A using *8, the caller's caller ID
successfully passed to my phone. I decide to pass the call to ext B.
On phone B, it display ext A not the original's
Il 20/01/2012 20:32, Alec Davis ha scritto:
This maybe not what you want.
Our solution was monitor a queue with a BLF, instead of a queue member
This reviewhttps://reviewboard.asterisk.org/r/1619/ allows a BLF lamp to
flash when a queue is ringing, then the queue can be picked up by the BLF
I think that part is working correctly I see boths asterisk are connected
on home_server:
syscon7*CLI iax2 show registry
Host dnsmgr UsernamePerceived Refresh State
192.168.141.1:4569N home_serve 192.168.141.8:4569 60
Registered
They are connected but not registered. The iax2 show registry on
home_server should show both peers.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, April 15, 2012 11:40 AM
To: Asterisk
As long as Host does not contain the peer's IP address in iax2 show peers then
it is not going to work and is not registered.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, April 15,
Do a blind transfer instead of attended transfer - the under the
covers changes in 10.X handle this for attended transfers, but to the best
of my knowledge, the blind transfer is the only solution in the 1.X tree.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten = _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
?
best regards
olivier
Le 15 avril 2012 12:55, Administrator TOOTAI ad...@tootai.net a écrit :
Le 15/04/2012 10:44, Olivier CALVANO a écrit :
Hi
I believe they were trying to say
exten = _x,2,AGI(MyScript.agi,${$CALLERID(num){0:4}})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Olivier
CALVANO
Sent: Sunday, April 15, 2012 1:52 PM
To: Asterisk Users
Very thanks for your help, but no, it's not good
Le 15 avril 2012 20:54, Danny Nicholas da...@debsinc.com a écrit :
I believe they were trying to say
exten = _x,2,AGI(MyScript.agi,${$CALLERID(num){0:4}})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
i am search on google ;=) but no result for this moment hihi
Le 15 avril 2012 21:14, Olivier CALVANO o.calv...@gmail.com a écrit :
Very thanks for your help, but no, it's not good
Le 15 avril 2012 20:54, Danny Nicholas da...@debsinc.com a écrit :
I believe they were trying to say
exten =
Maybe you could try sipaddheader but you will need to use sip trunk instead of
iax trunk
Att.
Eduardo
On 15/04/2012, at 16:28, Olivier CALVANO o.calv...@gmail.com wrote:
i am search on google ;=) but no result for this moment hihi
Le 15 avril 2012 21:14, Olivier CALVANO
Change this
exten = _x,2,AGI(MyScript.agi,${$CALLERID(num){0:4}})
to this
exten = _x,2,Verbose(passed ID ${$CALLERID(num)})
exten = _x,3,AGI(MyScript.agi,${$CALLERID(num){0:4}})
and post your CLI output. We need to see if the OP's suggestion is getting
to Asterisk #2.
-Original Message-
Ok,
the CLI of the server one :
-- Executing [06@Unlimited-outgoing:3]
Dial(SIP/USRSIP05-0a7a52e8,
IAX2/Srv2/06/USRSIP05,180,rt) in new stack
-- Called Trader/06/USRSIP05
The CLI of the server two:
srv2*CLI
-- Accepting AUTHENTICATED call from 172.20.8.1:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi all!
Some time ago I'm using Asterisk (currently 1.8.10.0) at home to manage
the calls. Nothing yet very complex, just something compiled by me using
the source code from the official site of the project and configuring
the files manually to both
Le 15/04/2012 20:51, Olivier CALVANO a écrit :
Hi
Thanks for your help but i don't know this variable: $CALLID[1-4]
it's correct:
exten = _x,2,AGI(MyScript.agi,${$CALLID[1-4]})
$CALLID1 $CALLID2 $CALLID3 $CALLID4
[...]
Le 15 avril 2012 12:55, Administrator
hi,
there are 3 new cdr fields in asterisk 1.8
(https://wiki.asterisk.org/wiki/display/AST/New+in+1.8#Newin1.8-CDR)
linkedid - is based on uniqueID, but spreads to other channels as
transfers, dials, etc are performed. Thus the pieces of CDR can be
grouped into multilegged sets.
sequence -
Hi All;
Is it normal if I used asterisk 1.4 and dahdi, then I will not find chan_dahdi
under /usr/lib/asterisk/modules? And I will not be able to type dahdi commands
(dahdi restart for example) in the asterisk CLI?
Actually what I found only the following:
app_dahdibarge.so app_dahdiras.so
One-way is ok as long as you don't have 2-way calling. To call A-B and
B-A both must be registered.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Joseph
Sent: Sunday, April 15, 2012 7:20 PM
To: Asterisk
Sent from my iPhone
--
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-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
http://www.asterisk.org/hello
asterisk-users
On Sun, 15 Apr 2012, Olivier CALVANO wrote:
actually, i have a asterisk server with all SIP Account.
this Asterisk server sent all outgoing call to a second Asterisk
server (and this asterisk sent to the
telco)
On the first Asterisk, i use:
exten = _x,1,Set(CDR(CodeTier)=BUS-FRAMOBI)
First of all, I want apologize for the first two blank emails that I sent out
by mistake.
I have Xorcom USB fxo channel bank, asterisk 1.6, freepbx 2.8. Up to now, the
lines connected from Telekom did not have caller id feature enabled, now that
we enabled we cannot see incoming caller id
On Sun, Apr 15, 2012 at 3:48 PM, bilal ghayyad bilmar...@yahoo.com wrote:
Hi All;
Is it normal if I used asterisk 1.4 and dahdi, then I will not find
chan_dahdi under /usr/lib/asterisk/modules? And I will not be able to type
dahdi commands (dahdi restart for example) in the asterisk CLI?
On Sun, Apr 15, 2012 at 2:49 PM, Olivier CALVANO o.calv...@gmail.comwrote:
The CLI of the server two:
srv2*CLI
-- Accepting AUTHENTICATED call from 172.20.8.1:
requested format = alaw,
requested prefs = (alaw|g729),
actual format = alaw,
host prefs =
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