Asterisk 1.8.3.2
I have been getting this warning constantly on CLI in a call scenario where
I use local channels to connect SIP with PSTN.
I use callfile and local channel to first call a PSTN number and if
answered, use local channel to call SIP phone with music on hold enabled in
Dial string.
Hello!
In your sip.conf use alaw as your first codec option and see what happens.Best
regards,
Fellipe Paes
Date: Tue, 28 Jun 2011 15:29:11 +0530
From: theasterisk...@gmail.com
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Asked to transmit frame type slin,while native
On Sat, Jun 18, 2011 at 6:40 AM, Larry Moore lmo...@starwon.com.au wrote:
On 18/06/2011 5:36 AM, Matteo Campana wrote:
Inviato da iPhone
Il giorno 16/giu/2011, alle ore 16:37, Eric Wielingewiel...@nyigc.com
ha scritto:
We experience the same thing. The solution we use is to not change
Hello Cédric
# when the virtual ip come up
ip r a SIP_PROVIDER_IP via GATEWAY_IP dev eth0:0
# when the virtual ip come down, maybe facultative because the route
is deleted when the interface fall down
ip r d SIP_PROVIDER_IP via GATEWAY_IP dev eth0:0
thank's for your hint.
i tried it
Thanks for the response.
I have disallow=all and allow=alaw in sip.conf for my SIP user.
Any other idea?
--AM
On Tue, Jun 28, 2011 at 4:23 PM, Fellipe Paes fellipe...@hotmail.comwrote:
Hello!
In your sip.conf use alaw as your first codec option and see what happens.
Best regards,
Fellipe
Hi,
I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of
those are corrupted (can`t be opened) or garbled. That is on only one
server, which is using the same Asterisk version (1.6.2.18) as the other
servers which are mostly fine.
What can be the cause? The
On Tue, Jun 28, 2011 at 10:30 AM, Mike l...@net-wall.com wrote:
Hi,
** **
I’ve had problems with MixMonitor recordings. A lot (I’d say almost 50%)
of those are corrupted (can`t be opened) or garbled. That is on only one
server, which is using the same Asterisk version (1.6.2.18) as
On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote:
I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of
those are corrupted (can`t be opened) or garbled. That is on only one
server, which is using the same Asterisk version (1.6.2.18) as the other
servers which are
We're a VoIP provider essentially competing with our local incumbent
Telco, and a sizeable number of our customers use satellite internet.
As a result, these customers never have ping times less than 500ms,
and are often exceeding 2500ms.
I manually apply a patch to the Asterisk source
Hi,
I've got a Polycom SPIP 650 3.3.1F-enabled phone with an 14-keys side-car.
What is the best way to assign a specific key to a given BLF (without having
to assign every previous key) ?
At the moment, I'm using settings like this :
...
attendant attendant.reg=1
Have you tried SIP session timer values in sip.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ernie Dunbar
Sent: Tuesday, June 28, 2011 9:33 PM
To: asterisk-users@lists.digium.com
Subject:
I think there is a bug in the Dial() application in Asterisk 1.6.2.17
that wasn't present in 1.4.23.1, and I'd like to see if anyone else
has this problem.
I've been able to reproduce this error: When you use the Dial()
command to send a call to both a SIP connection and a DAHDI
2011/6/20 Gord Urquhart gord...@gmail.com
I missed one important parameter in my setup of BLF for polycom phones (at
least if you want to do one touch directed pickup)
In sip.conf add
notifycid=yes
the notifycid=yes causes asterisk to add a target uri = callID to the XML
of the
Yes, these are our session-timer settings in sip.conf:
session-timers=originate
session-expires=600
session-minse=90
session-refresher=uas
Quoting Faisal Hanif fai...@vopium.com:
Have you tried SIP session timer values in sip.conf
-Original Message-
From:
On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote:
I've had problems with MixMonitor recordings. A lot (I'd say almost
50%) of those are corrupted (can`t be opened) or garbled. That is on
only one server, which is using the same Asterisk version (1.6.2.18)
as the other servers which
I've had problems with MixMonitor recordings. A lot (I'd say almost 50%) of
those are corrupted (can`t be opened) or garbled. That is on only one
server, which is using the same Asterisk version (1.6.2.18) as the other
servers which are mostly fine.
What can be the cause? The conversation
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
Pretty sure with Polycom's you can only specify in sequential order, you can't
pick and choose the buttons you assign.
Thanks,
--Warren Selby, dCAP
On Jun 28, 2011, at 11:58 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I've got a Polycom SPIP 650 3.3.1F-enabled phone with an 14-keys side-car.
On Tue, Jun 28, 2011 at 01:24:43PM -0400, Mike wrote:
On Tue, Jun 28, 2011 at 11:30:54AM -0400, Mike wrote:
I've had problems with MixMonitor recordings. A lot (I'd say almost
50%) of those are corrupted (can`t be opened) or garbled. That is on
only one server, which is using the
Hello, I have Asterisk 1.6 running on Centos, Also I have one analog
telephone line coming on
Wildcard TDM400P REV E/F Board 5
I can't get asterisk to dectect call coming from analog line.
Here is my /etc/dahdi/system.conf
fxsks=1
# global data
loadzone = us
defaultzone = us
A couple things -
First, in extensions.con your context is [my-phone], but you're using my-phones
in your dahdi and sip.conf files.
Second, you need an 's' extension somewhere in your receiving context in order
for asterisk to answer the incoming analog call.
Third, I think you've got some
My mistake I had fix that typo but no luck
Thanks,
motty
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Richard
Mudgett
Sent: Tuesday, June 28, 2011 10:37 AM
To: Asterisk Users Mailing List - Non-Commercial
Thanks Warren,
I have gone ahead and correct my typo. Also, I created 's' extension as you
suggested.
exten = s,1,Wait(1)
exten = s,2,Answer()
exten = s,3,NoOp(${CALLERID})
exten = s,n,NoOp(${CALLERIDNUM})
exten = s,n,NoOp(${CALLERIDNAME})
exten = s,n,Wait(4)
exten = s,n,Playback(tt-easels)
What you say...David Backeberg (dbackeb...@gmail.com):
On Fri, Jun 24, 2011 at 4:55 PM, Hose hose+aster...@bluemaggottowel.com
wrote:
Can anyone recommend some kind of virtual t.38 fax software? I'd like
to test/debug some of the t.38 stuff, but it'd be much easier if I had a
software
Hi all,
Anybody know if is it possible to add # at the end of dialled number ?
kinda : exten = _00[1-5]XXX,1,Dial(DAHDI/g0/021${EXTEN:4},25,T)
In this line I am switching the C.O code but , how could I put # automatic at
the end ?
Thanks in advanced!
Att,
Flavio Roberto
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Flavio Miranda
Sent: Tuesday, June 28, 2011 1:22 PM
To: Asterisk Asterisk
Subject: [asterisk-users] Add # at the end of dialled number
Hi all,
Anybody know if is it possible
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Shaun Ruffell
Sent: Tuesday, June 28, 2011 1:34 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] MixMonitor -
2011/6/28 Warren Selby wcse...@selbytech.com
Pretty sure with Polycom's you can only specify in sequential order, you
can't pick and choose the buttons you assign.
Thanks,
--Warren Selby, dCAP
On Jun 28, 2011, at 11:58 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
I've got a Polycom SPIP
I need to make an IVR as follows:
1 an incoming call and run an
AGI script to alert the database, everything perfect here.
2 Play a
music on hold and executes a loop while searching the database for a
change in a field when the field change, cut the music on hold and keep
doing things. I
Hi List,
I'm trying to get working SIPp with media but something is wrong (it's
working well without media), please help:
This is the command I send at SIPp server:
./sipp -sn uac_pcap -d 5000 -s 2006 192.168.1.18 -l 20 -trace_err
This is the result I see:
Last Error: Aborting call
Waitexten is preferable to using the AGI method
Exten = 4321,n,waitexten(5) - just wait 5 seconds
Exten = 4321,n,waitexten(5,m) – wait with music
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Ezequiel Lovelle
Sent:
Asterisk Project Security Advisory - AST-2011-011
++
| Product | Asterisk |
Hi everyone,
When doing a sip show settings on Asterisk 1.6.2.18, I see the following:
Match Auth Username:No
Allow unknown access: Yes
Allow subscriptions:Yes
Allow overlap dialing: Yes
Allow promsic. redir: No
Enable call counters: No
What do each of above signify?
The Asterisk Development Team has announced the release of Asterisk versions
1.4.41.2, 1.6.2.18.2, and 1.8.4.4, which are security releases.
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/releases
The release of Asterisk 1.4.41.2,
On Tue, Jun 28, 2011 at 4:53 PM, Bruce B bruceb...@gmail.com wrote:
Hi everyone,
When doing a sip show settings on Asterisk 1.6.2.18, I see the following:
Match Auth Username: No
Allow unknown access: Yes
Allow subscriptions: Yes
Allow overlap dialing: Yes
Allow promsic.
works !
Thanks!
Att,
Flavio Roberto Miranda
MSN:flaviormira...@hotmail.com
Skype: flaviormiranda
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Tue, 28 Jun 2011 13:31:41 -0500
Subject: Re: [asterisk-users] Add # at the end of dialled number
From:
Dear sir,
i would like to request you please support your web . do some work your web ,
please accept my request.
Thanks
akram
Dhaka, Bangladesh
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