2013/3/7 Steve Edwards asterisk@sedwards.com
Please don't top-post.
On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
Are you wiring a building where multiple-line SIP gateways make sense?
How about a description of what you are trying
2013/3/7 Steve Edwards asterisk@sedwards.com
Please don't top-post.
On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
Are you wiring a building where multiple-line SIP gateways make sense?
How about a description of what you are
On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
Are you wiring a building where multiple-line SIP gateways make sense?
How about a description of what you are trying
That should be ok.
Try the following: open 2 shells. In the first one type watch df -h.
In the second one you start the compilation. While compilation is
running watch the first shell. The given command refreshes all 2 seconds
the display and shows the used/free disk space. _Perhaps_ it will
2013/3/7 Duncan Turnbull dun...@e-simple.co.nz
On 7/03/2013, at 9:29 PM, Kamlesh Kumar kamlesh_...@hotmail.com wrote:
On Thu, 7 Mar 2013, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
Are you wiring a building where multiple-line SIP gateways make sense?
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 Bad Extension back from 10.4.0.10
-- Stopped music on hold on
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk?
On 7/3/13 6:50 am, Bharat Lalcheta wrote:
You can use ATA box with pstn phone to reduce cost.
I would caution against that approach. Analogue to Digital conversions
often seem to have 'problems' - mostly related to hangup detection
and/or echo. If you really do want to use analogue phones,
Le 7/03/13 11:21, Steven Howes a écrit :
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
Hello Steve,
After checking, I confirm that the call is cut precisely to 900 seconds
(15 min).
10.4.0.1 =
From: Chris Bagnall aster...@lists.minotaur.cc
To: asterisk-users@lists.digium.com,
Date: 03/07/2013 06:43 AM
Subject:Re: [asterisk-users] asterisk with 1000 extensions
Sent by:asterisk-users-boun...@lists.digium.com
On 7/3/13 6:50 am, Bharat Lalcheta wrote:
You can
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
same = n(email),System(/usr/local/bin/emailme)
same =
sean darcy wrote:
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
same =
Can u debug on AS ?
On Thu, Mar 7, 2013 at 9:20 AM, Mickael Monsieur
mickael.monsi...@gmail.com wrote:
Le 7/03/13 11:21, Steven Howes a écrit :
On 7 Mar 2013, at 10:12, Mickael Monsieur wrote:
Do you have an explanation?
Put a SIP debug on and we may be able to find one..
Steve
Hello
Le 7/03/13 11:12, Mickael Monsieur a écrit :
Hello,
I have a Cisco AS5300 connected to Asterisk (1.6.2.9)
Between 15-16 minutes, the call is disconnected without reason.
Here is what is displayed in the debug:
Received an SDES from 10.4.0.10:17399
-- Got SIP response 420 Bad Extension back
There might be something wrong with the evaluation. Can you post more
console regarding the GotoIf?
And are you sure its in the right context?
Sent from my iPhone
On 7 mrt. 2013, at 15:48, sean darcy seandar...@gmail.com wrote:
Some calls I get from google voice, I just send myself an email
Hi,
I am developing a call recording application on Asterisk 11.2 and have this
configuration in my dialplan:
[macro-ccdev2-rec]
exten = s,1,MixMonitor(${ARG1},b)
[outgoing-originate]
exten = _X.,1,NoOp(Will send call to ${EXTEN})
exten = _X.,n,Dial(SIP/${EXTEN}@x.y.z)
On 03/07/2013 09:48 AM, Joshua Colp wrote:
sean darcy wrote:
Some calls I get from google voice, I just send myself an email about
the call and want to hangup. But I can't seem to make gv know I've
hung up.
extensions.conf:
same = n,GoToIf($[${CALLERID(num)}=office]?email)
.
On Thu, Mar 7, 2013 at 1:44 AM, Duncan Turnbull dun...@e-simple.co.nzwrote:
This is not school assignment or home work :) We need to setup in society
buildings. Each flat will have SIP extension (hard phone) registered on
asterisk server. Calling between SIP extensions is required. No PSTN /
Greetings.
I got an extension on my Elastix who cannot pick calls on the other
extensions, but It can transfer his calls to the other extensions. When
this extension tries to pickup a call pressing *8 it simply does not pick
it up. Transfering calls works just fine so dtmf may be not the
If I was in your shoes, I'll check in the elastix mailing list... Asterisk
itself can't be blamed.
Leandro
I am typing from my mobile phone...
Il giorno 07/mar/2013 19:06, Luis H. Forchesatto
luisforchesa...@gmail.com ha scritto:
Greetings.
I got an extension on my Elastix who cannot pick
On 8/03/2013, at 7:46 AM, Leandro Dardini ldard...@gmail.com wrote:
If I was in your shoes, I'll check in the elastix mailing list... Asterisk
itself can't be blamed.
Leandro
I am typing from my mobile phone...
Il giorno 07/mar/2013 19:06, Luis H. Forchesatto
On 03/07/2013 10:32 AM, Zohair Raza wrote:
Its Centos 6
with kernel 2.6.32-279.19.1.el6.x86_64
Please follow the instructions on the wiki [1] for generating a
backtrace. When you have a backtrace from the crash, please create an
issue in the issue tracker [2] and attach the backtrace to
Did u test it without abrt?
On Mar 7, 2013 10:03 PM, Zohair Raza engineerzuhairr...@gmail.com wrote:
Its Centos 6
with kernel 2.6.32-279.19.1.el6.x86_64
Regards,
Zohair Raza
On Thu, Mar 7, 2013 at 8:28 AM, Bharat Lalcheta
bharatlalch...@gmail.comwrote:
Can you provide OS details ?
hi,
hard to understand, what your objective is... at least for me ;-)
so you want to establish a call (triggered by ami) between two partys,
record the conversation
and save the file to a(nother) server (afterwards), right?
and another task is to establish (also ami triggered) a call to a
do you have only ONE phone, that can´t pickup, or is this a general problem?
is pickup configured (feature.conf) AND enabled ?
regards,
yves
Am 07.03.2013 19:05, schrieb Luis H. Forchesatto:
Greetings.
I got an extension on my Elastix who cannot pick calls on the other
extensions, but It
Its only ONE phone who doesnt pickup calls.
2013/3/7 Yves A. yves...@gmx.de
do you have only ONE phone, that can´t pickup, or is this a general
problem?
is pickup configured (feature.conf) AND enabled ?
regards,
yves
Am 07.03.2013 19:05, schrieb Luis H. Forchesatto:
Greetings.
I
is it the same type and make of phone than one of the working ones?
- compare (dtmf) settings, firmware release etc.
- check call-group and pickup group... is the non working extension
configured there?
regards,
yves
Am 07.03.2013 20:28, schrieb Luis H. Forchesatto:
Its only ONE phone who
Yes, both are configured in the same ata (linksys pap2) and the
configuration options are the same. Call group and pick group are the same
for both too.
2013/3/7 Yves A. yves...@gmx.de
is it the same type and make of phone than one of the working ones?
- compare (dtmf) settings, firmware
mmh... should work... (i think you checked double and applied any
changes, right..?
sometimes deleting the extension and configuring a new one can fulfil
wonders...)
I have no further tip... maybe elastix support or forum can help... if
you are familiar with
cli output and sip debugging...
Hi,
Here is the configuration of the server that I currently have
extension 100 (SIP) =(SIP)asterisk server 1.8.18(IAX trunk) ===(IAX
trunk)asterisk server 1.4.32(SIP) === SIP Providers
The issue is while dialing out from extension 100(sip) if the providers
sends back 180 Rining the SIP
Hi, thanks.
.Did that include 'agi set debug on?'
Yes, that include this command.
Can you 'cut-n-paste' the relevant 'sanitized' console output?
Ok. This is:
trixbox146002*CLI
-- Launched AGI Script /var/lib/asterisk/agi-bin/agi-test.agi
trixbox146002*CLI
Has any one ever worked with placing idle display images onto the Polycom
SPIP331 phones? I have got it working but when the image is displayed the
clock is moved to the top of the screen. That is great but it scrolls
between the clock and the registered extension(s) . Has anyone figured out
a
I was hoping someone might have some knowledge to impart regarding
VOIP PRI Gateways or the psudo ISDN services being offered these days.
The official line in Australia is that true ISDN services are on
their way out.
I am testing a service provided by one of the telcos I am told that it
cannot
On Thu, 7 Mar 2013 17:12:47 -0800
Bryan Anderson shadow...@gmail.com wrote:
Has any one ever worked with placing idle display images onto the
Polycom SPIP331 phones? I have got it working but when the image is
displayed the clock is moved to the top of the screen. That is
great but it
As far as i understand your requirements, there is no need to use
macro for recording, You can directly call mixmonitor before Dial
application in your dialplan with required options. For transfer of
file, you are using AGI in h priority. However, you have to use
DeadAgi in h extension. As your
If I understand you correctly, you test a service which converter SIP to ISDN
PRI
On Mar 8, 2013, at 2:16 AM, Daniel Harper dan...@harper.net.nz wrote:
I was hoping someone might have some knowledge to impart regarding
VOIP PRI Gateways or the psudo ISDN services being offered these days.
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