Displaying 13 results from an estimated 13 matches for "d2010".
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2010
2010 Jun 26
2
Detecting hook flash in asterisk
Hello,
Is it possible to detect a hook flash in asterisk. I want to be able to
perform some functions an hook flash.
I have the following entry in features.conf which executes a Macro on
detecting key press '**'.
[applicationmap]
test => **,caller,Macro,testflash
Is it possible to do this action on hook flash?
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2010 Jul 01
3
Originate multiple channels
Hello,
Is it possible to use the asterisk manager interface to originate
multiple channels?
like
Action: Originate
Channel: SIP/101&SIP/102
So that both extensions 101 and 102 rings simultaneously.
I am using asterisk manager interface over http.
Thanks
2011 May 02
1
default context overrides context of peer
Hello,
I upgraded from asterisk 1.6.2.7 to asterisk 1.6.2.17.
I have context=defcontext set in sip.conf. For each peer I have
context=outcontext in the peer definition since I want outgoing calls
from registered SIP peers to go through context 'outcontext'. This
used to work in the older version (1.6.2.7), but after upgrading this
has stopped working. Now outgoing calls are going to
2010 Jan 16
0
Realtime cached values
Hello,
Does the cached values for realtime peers expire automatically?
I have rtcachefriends=yes in sip.conf. When the peer registers for the first
time it is cached. After the first registration if I modify the peer in the
database the new values are not used until I do a 'sip prune realtime peer'
or 'sip reload'. Is it possible to specify an expiry time for the cached
values so
2010 Jan 16
1
Hint for realtime peers
Hello,
When I create a sip peer? in users.conf then a hint is automatically
created for that peer. But when I create a peer in sip.conf or a
realtime peer with the same values then this hint is not created.
Every time I add such peers I have to add a hint in extensions.conf.
Is it possible to have something like?? exten =>
_XXX,hint,SIP/${EXTEN}? in extensions.conf so that I don't have
2010 Feb 09
1
Not able to receive fax
Hello,
I have been trying to setup asterisk (1.6.2.0) to receive fax. I am
able to receive faxes sent from a zoiper softphone connected to
asterisk.
I have some DID numbers (with T.38 support) forwarded to my asterisk
pbx. I am not able to receive faxes from these numbers. The error I
get on the asterisk console is
"phase_e_handler: Error transmitting fax. result=49: The call dropped
2010 Feb 27
0
"Unexpected message received" when receiving Fax
Hello,
I have been trying to setup asterisk 1.6.2.0 to receive fax. I have
two SIP trunks connected to asterisk. One of them is a VoIP service
provider and the other is an audiocodes gateway connected with pstn
and fax lines. I am able to receive faxes on the DID numbers provided
by the VoIP service provider, but I am not able to receive fax through
the fax lines connected through audiocodes.
2010 Jul 15
0
Get channel name of originated channel
Hello,
I am using asterisk manager interface (http) for originating calls.
How can I get the name of the channel which is created by originate? I
want to use this channel for other manager commands like Atxfer,
Monitor, Hangup etc.
If I do action=originate, channel=SIP/200 then it creates a channel
like 'SIP/200-0865ff80' which I can see in the asterisk console using
"core show
2010 Jul 27
0
sip peer becomes unreachable in Asterisk 1.6
Hello,
I recently upgraded from asterisk 1.4 to 1.6. I am using the same SIP
settings in sip.conf in this version also. I am facing a problem when
a SIP client makes a call.
When a SIP client registers to asterisk its status shows 'OK' and it
is able to receive incoming calls. But as soon as this client make a
call, its status becomes 'UNREACHABLE' and it cannot receive any
2013 Feb 23
0
Connecting to multiple databases using res_config_pgsql
Hello,
How do I use multiple postgresql databases using res_config_pgsql?
I tried creating multiple contexts in res_pgsql.conf, but asterisk is
only using the 'general' context.
My res_pgsq.conf is
[general] ;; Connect to mydb on localhost
dbport=5432
dbname=mydb
dbuser=pgdbuser
requirements=warn
[pgwritedb] ;; Connect to mydb2 on another host
dbhost=<IP
2010 May 26
2
Getting 'username' of sip peer
Hello,
I have a few entries for sip peers in sip.conf with different name and
username, like
[TestSIPUser]
type=peer
host=dynamic
username=testuser
secret=1234
context=test_context
[TestNewUser]
type=peer
host=dynamic
username=newsipuser
secret=3456
context=test_context
When a call is made from any of these peers I want to get the username
of the peer.
for eg:- If a call is being made from
2010 May 21
2
Using unix socket to connect with database
Hello,
I am using asterisk realtime with a postgresql database on the same server.
In res_pgsql.conf I have specified
[general]
dbhost=localhost
dbport=5432
dbname=asteriskdb
dbuser=psql
dbsock=/tmp/.s.PGSQL.5432
Since both asterisk and db are on same server, I would like asterisk
to connect to db using the local unix socket. However asterisk is not
using the local unix socket to connect to
2010 Jan 22
5
Set CDR userfield for Queues
Hello,
I am using Queue application with multiple agents in each queue. I
want to set the CDR(userfield) for each cdr based on the agent
answering the call. Is it possible to do this?
Thanks