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
On Thu, Feb 14, 2013 at 8:35 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Thu, 14 Feb 2013, Deepesh D wrote:
The problem I am facing is that sometimes the variables are wrongly
received as 0 (zero) in the dialplan even if the AGI has set it to a
non-zero value.
Are the variables
Hello,
What is the maximum number of meetme's allowed by asterisk.
On my server with an 8 GB memory, I start getting the following error
after 150-160 meetme's are created
WARNING[3485]: app_meetme.c:1820 conf_run: Unable to open DAHDI pseudo
channel: Cannot allocate memory
At this time the
...@jttech.se wrote:
2012-12-24 16:13, Deepesh D skrev:
Hello,
What is the maximum number of meetme's allowed by asterisk.
On my server with an 8 GB memory, I start getting the following error
after 150-160 meetme's are created
WARNING[3485]: app_meetme.c:1820 conf_run: Unable to open DAHDI
way I can get those also working.
On Fri, Oct 12, 2012 at 1:12 AM, Joshua Colp jc...@digium.com wrote:
Deepesh D wrote:
If C1 dials S1 and then S1 dials S2 to transfer the call then S1 still
remains in the loop till the call is finished. What I wanted to do is
to reduce the number of calls
X, digest has
[Oct 12 18:21:06] NOTICE[30483]: chan_sip.c:22046
handle_request_invite: Failed to authenticate device
sip:X@192.168.1.1:14500;tag=3047
On Fri, Oct 12, 2012 at 5:00 PM, Joshua Colp jc...@digium.com wrote:
Deepesh D wrote:
I made these changes in dialplan and it worked
'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have
is 'default'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I
'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have
'
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have
context.
On Wed, Oct 10, 2012 at 9:22 PM, Sean Darcy seandar...@gmail.com wrote:
On Wed, Oct 10, 2012 at 8:06 AM, Deepesh D deep.d2...@gmail.com wrote:
Hello,
How do I use the asterisk application 'Transfer' to transfer a SIP
call from one asterisk to another?
I have the following scenario. I have
] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 5:44 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] How to use 'Transfer' to send calls to another
asterisk?
In all my peer definitions on S1 and S2 I define the context as
'test_context
the loop, the call should now be handled by S2
On Thu, Oct 11, 2012 at 7:22 PM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Thursday, October 11, 2012 8:48
Hello,
Is there a way by which a SIP peer can only transfer an incoming call?
When I set 'allowtransfer=yes' the peer is able to transfer both outgoing
calls and incoming calls. Is there any SIP setting or dialplan setting by
which I can restrict transfer of outgoing calls.
The asterisk version
Hello all,
What are the different type of bridging used by asterisk in a SIP
call? What is the difference between Packet2Packet bridging, Remote
bridging and Native bridging?
Can someone please explain me the differences or point me to a good
documentation of the same.
Thanks
--
, at 08:43 , Deepesh D wrote:
What are the different type of bridging used by asterisk in a SIP
call? What is the difference between Packet2Packet bridging, Remote
bridging and Native bridging?
Packet2Packet bridging is when RTP datagrams are forwarded by Asterisk
without modification
nogoodnameswereavaila...@gmail.com wrote:
On Mon, May 2, 2011 at 1:09 PM, Deepesh D deep.d2...@gmail.com wrote:
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
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
Hello Michael,
You could try to achieve this functionality in dialplan by using the
applications AddQueueMember/RemoveQueueMember which are used to
dynamically add/remove queue members.
An example dialplan flow for agent login will be
1. get the SIP interface from which the agent is logging
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
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
Hello,
Is it possible to use the asterisk manager interface to originate
multiple channels?
like
Action: Originate
Channel: SIP/101SIP/102
So that both extensions 101 and 102 rings simultaneously.
I am using asterisk manager interface over http.
Thanks
--
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
Hello,
I wanted to add the functionality of 3-way conference to my asterisk pbx
using meetme or confbridge. During a call the user should be able to put the
other party on hold and dial another number, then on dialing some key
sequence all three of them enters into a conference. Is it possible to
Hello,
I have the following dialplan
exten = _X.,1,Set(CDR(userfield)=test)
exten = _X.,n,Do some checks and hangup if checks fail
exten = _X.,n,Dial(SIP/${EXTEN})
exten = _X.,n,Hangup
1. If the Dial fails with a busy, noanswer or congestion then a cdr is
generated.
2. If the call fails before
will
populate CALLERID(name)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Wednesday, May 26, 2010 12:18 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Getting 'username' of sip
, Deepesh D wrote:
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 'TestSIPUser' then I want to be
able to get the value 'testuser'
I can think of two ways of doing this. The first is to use the
SIPCHANINFO() dialplan function
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
via socket.
WARNING[1819]: res_config_pgsql.c:1383 parse_config: PostgreSQL
RealTime: No database port found, using 5432 as default.
But there is no connection being made to the database.
On Sat, May 22, 2010 at 3:25 AM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/21/2010 02:48 AM, Deepesh
I am using Asterisk 1.6.2.7
On Sat, May 22, 2010 at 7:20 PM, Kevin P. Fleming kpflem...@digium.com wrote:
On 05/22/2010 02:07 AM, Deepesh D wrote:
I tried removing the dbhost and dbport entries and restarting asterisk.
During startup the following warnings are shown and it gets stuck up
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
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
Hello,
Is it possible to use asterisk in T.38 pass through mode with reinvite?
My fax calls are getting disconnected if canreinvite=yes. It works
only if I make canreinvite=no. Normal calls work in both cases.
Thanks
--
_
--
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
Thanks. It's working now.
In my sip.conf I had 't8pt_udptl=yes'. I changed it to
't8pt_udptl=yes,redundancy,maxdatagram=400' and it started working.
On Tue, Feb 9, 2010 at 6:22 PM, Tommy Botten Jensen
tommy.jen...@freecode.no wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA512
I have
Hello,
I have been trying to setup asterisk 1.6.1.1 to receive fax. Whenever
a SIP peer (zoiper soft phones) tries to send a fax message asterisk
responds by sending a 488 Not acceptable here and the sending fails.
I tried changing a few sip settings like canreinvite and codec
preferences, but it
Hello,
Can someone please explain me how the adminpin for conference rooms is used?
In the following dialplan
exten = 1122,1,MeetMe(${conf-room-no})
If users join this conference by dialing adminpin or pin will it make
any difference to the user?
Does dialling the adminpin give the user any
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