Hello all,
I use asterisk 1.6.0.13 with realtime DB.
when a phone (SIP) receive a new voicemail, asterisk update with a NOTIFY
the number of waiting message to the phone, so all is ok .
if I use the voicemail application to consult and to delete voicemail,
asterisk again update correctly the
Dear Mauro,
Your requirement seems Clone line feature for asterisk. The same question I've
asked here in this group, a months later but could't get well. But actually
implemented it now!
It is done using AMI. Here is its basic psudo code.
# ami-event.pl
Connect to AMI
Read the AMI Events
Hi Dhaval,
Echo depends of the far end not directly Asterisk or Digium cards.
If the remote telephone or PBX return your your voice, you will ear your echo.
If the remote don't return you your audio signal, no echo.
The passedthrough circuit along the complete path can also return you echo but
Hi!
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup with Asterisk - is it possible to get
RTP statistics out of Asterisk (e.g. jitter, packet loss, reordering ...)?
Thanks
I do a test with asterisk 1.4.21.2 and it work perfectly . any idea.
Laurent
2009/8/27 laurent schweizer laurent.schwei...@gmail.com
Hello all,
I use asterisk 1.6.0.13 with realtime DB.
when a phone (SIP) receive a new voicemail, asterisk update with a NOTIFY
the number of waiting
On 27/08/09 9:24 PM, Klaus Darilion wrote:
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup with Asterisk - is it possible to get
RTP statistics out of Asterisk (e.g.
Hi Olle and co
I'm really struggling to convert this into a feature request.
Can anyone help?
Regards,
Richard
--
Richard Brady
T: +44 (0)7771 623 348
E: rnbr...@gmail.com
2009/4/3 Richard Brady rnbr...@gmail.com:
Agreed Olle, it would definitely have to be option driven, not least
for
Is there any documentation on IAX RSA authentication because I followed
http://www.voip-info.org/wiki/index.php?page=Asterisk+iax+rsa+auth and
it's not working...
Asterisk 1 :
-r--r--r-- 1 root root 272 Aug 25 10:34 server2.pub
-r 1 root root 963 Aug 24 19:38 server1.key
Asterisk 2 :
Hi
I want to create a feature to on/off a kind of followme option.
Is it possible ?
I success to use default feature blindxfer and atxfer defined in
features.conf in my dialplan by writting something like this :
exten = 111,1,Set(DYNAMIC_FEATURES=atxferblindxfer)
exten = 111,n,Dial(SIP/111,,Tt)
So Registration succeeds but the server is still UNAUTHENTICATED :
[Aug 27 13:16:18] -- Registered IAX2 'BOX-YOCAN' (UNAUTHENTICATED)
at 78.xx.xx.xx:4569
I have initialised the keys on both the Asterisk-servers at startup
(asterisk -ic)
On Thu, 2009-08-27 at 11:40 +0200, jonas kellens
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laurent schweizer wrote:
how can I indicate to asterisk that number of message has changed and
that he need to do an update.
I'm not sure what effect realtime DB has on it but did you notice the
voicemail.conf parameters: pollmailboxes pollfreq ?
hallo Dhaval, to save money, you can use oslec as software EC in dahdi, then
you dont need hardware echo cancellation. AFAIK oslec now in dahdi trunk and
in the linux kernel. if you have problems with oslec, you can try the oslec
mailing list:
thanks Kristijan
thanks for your reply
i will try oslec
regards
Dhaval
On Thu, Aug 27, 2009 at 5:01 PM, Kristijan Vrban
vrban.l...@googlemail.comwrote:
hallo Dhaval, to save money, you can use oslec as software EC in dahdi,
then you dont need hardware echo cancellation. AFAIK oslec now in
I am running asterisk 1.2.9.1. I looked into devstate(). I still can't
figure how to use it.
Below is my script. I am dialing 1215,then I get a prompt on the state of
extension 1215. I have set extension 1215 to busy(by making several calls to
other extensions). I still get the prompt 1215 has a
...the above post lacked some details
I am running asterisk 1.2.9.1. I looked into devstate(). I still can't
figure how to use it. My dial plan gives 1215 has state 0 on all
scenarios.
Below is my script. I am dialing 1215,then I get a prompt on the state of
extension 1215. I have set extension
Hi,
Having seen the messages on the dev list a couple of weeks ago about chan_sebi
I thought I would try to get it going on my system.
I am using 1.6.1.4 so I first upgraded the driver to work with 1.6. I think it
is mostly correct and the bit I have wrong shouldn't be causing me the problems
Hi All
For recording inbound call we are using following line in dial
plan.But we wish to set file name which describe who attend the call or lets
say extension of the call attendant.
Current line in dial plan to set file name is like this-
yes thanks, it's working.
Laurent
2009/8/27 Barry L. Kline blkl...@attglobal.net
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laurent schweizer wrote:
how can I indicate to asterisk that number of message has changed and
that he need to do an update.
I'm not sure what effect
Could be a CASE or character issue. Have you tried the simple foo/bar
combination to eliminate other considerations?
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of jonas kellens
Sent: Wednesday, August 26, 2009 4:30 PM
Hi Matt!
Matt Riddell schrieb:
On 27/08/09 9:24 PM, Klaus Darilion wrote:
I want to use Asterisk as load generator to test quality degradation
with increased load (e.g. testing other SIP equipment or IP-links).
Is anybody aware of such a setup with Asterisk - is it possible to get
RTP
On Aug 25, 2009, at 5:59 AM, Olle E. Johansson wrote:
Hello Asterisk users around the world!
Recently, I have been working with pretty large Asterisk
installations. 300 servers running Asterisk and Kamailio (OpenSER).
Replacing large Nortel systems with just a few tiny boxes and other
Hi Muhammad, and thanks a lot for the answer.
On this moment I'm making some tests in order to collect enough
information to participate of a meeting at the end of this day regarding
the use of Asterisk.
I won't have time to validate your contribution before this meeting and
this info would be
Danny Nicholas wrote:
This may be a dumb question, but here goes. When I was
on 1.4.21.2 using Zaptel, I had (at least as far as I could tell) access
to line supervision on my POTS lines using a TDM400P/TDM410P. Since
upgrading to the DAHDI branches of 1.4 (SVN and
Greetings,
This may be a dumb question, but here goes. When I was on
1.4.21.2 using Zaptel, I had (at least as far as I could tell) access to
line supervision on my POTS lines using a TDM400P/TDM410P. Since upgrading
to the DAHDI branches of 1.4 (SVN and 1.4.26.1), I've only
I guess this is sort of a moot point since POTS is a dinosaur technology. I
just wish I had good technical notes to prove that this really did work
under Zaptel and at least 1 flavor of DAHDI/libpri.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
What's the big deal I can get 11.000 (11) calls on an Acer netbook ;-).
11,000 Wow!
My testing has shown that multiple core really didn't do much for
asterisk scaling.. My tests show a very busy first core and the rest
idle. However I am a big fan of HP servers I run 8 core DL360's VMware
Danny Nicholas wrote:
I guess this is sort of a moot point since POTS is a dinosaur technology. I
just wish I had good technical notes to prove that this really did work
under Zaptel and at least 1 flavor of DAHDI/libpri.
libpri is not related, since it is only used for PRI and BRI links.
--
On Thu, Aug 27, 2009 at 10:14:25AM -0500, Danny Nicholas wrote:
Greetings,
This may be a dumb question, but here goes. When I was on
1.4.21.2 using Zaptel, I had (at least as far as I could tell) access to
line supervision on my POTS lines using a TDM400P/TDM410P. Since
Thanks for the link!
Original Message
Subject: Re: [asterisk-users] Fw: app_swift issue
From: ABBAS SHAKEEL shakeel.abbas@gmail.com
Date: Wed, August 26, 2009 7:37 pm
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Thanks Todd
To answer your questions - 1 - yes it did or I wouldn't be chasing this dog.
2 - no, but I'm going to try and regress dahdi.conf to match Zapata.conf
3 - yes.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
For the benefit of our readers, I will convert this back to a more
readable in-line replies format:
On Thu, Aug 27, 2009 at 11:45:10AM -0500, Danny Nicholas wrote:
On Thursday, August 27, 2009 11:36 AM, Tzafrir Cohen wrote:
Has this actually ever worked?
yes it did or I wouldn't be chasing
I'm trying to figure out the maximum length of a cisco 7960 password
in the SIPmac.cfg file. An Aastra9133i can take at least a
36-character password, but the cisco craps out (can't authenticate)
In order to stop me from doing a brute-force test, does anyone know
the password lengths of
Aastra
My company for various reasons has asked that I come up with a way to
have previously parked calls be re-parked in the same parking slot. I
have looked at setting up asterisk so that the receptionist chooses
which slot to place a call, but I think there is an easier way. That is
when I came
Hello, all. In our multi-tenant environment, we would like to be able
to use the reinvite media redirection within Asterisk for calls within a
tenant but not between tenants. We would like inter-tenant calls to be
fully proxied by the Asterisk server. I think the answer is, we
can't, but I
Hey guys,
I've been having a very odd problem that happens intermittently. I've
had this happen with only a couple of providers and somewhat rarely but
its to the point now that we need to fix it to be able to do business.
The scenario is as follows: We have a DID provider that routes calls to
You could put something into the Asterisk Database with DBput/DBget.
I don't have an example off hand, but create a stickypark family and
store which channels go back into which parking slot. Or something to
that effect, and it would exist until you remove it from the database.
-Jonathan
On
Hi list,
I'd like to have the callers to listen to the advertisement (music on
hold) before the agents answer them. So, I have wrapuptime=10 in
queue.conf, but the call still goes straight to the agents without
delay.
Here's my queue.conf:
[general]
persistentmembers = yes
[738]
musiconhold =
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Andy Kuo wrote:
Hi list,
I'd like to have the callers to listen to the advertisement (music on
hold) before the agents answer them. So, I have wrapuptime=10 in
queue.conf, but the call still goes straight to the agents without
delay.
Andy
At 11:52 AM on 27 Aug 2009, Mat Murdock wrote:
[parkedcallstimeout]
exten = _SIP011XX,1,Answer()
exten = _SIP011XX,n,NoOp(Call Parked on: ${PARKINGSLOT})
exten = _SIP011XX,n,NoOp(This is who parked us: ${EXTEN})
exten = _SIP011XX,n,Set(PARKINGEXTEN=${PARKINGSLOT})
;This sets the
Hey guys,
For future reference if it happens to anyone else. In this case the
problem was that for whatever reason sending a callerid with a + in it
caused the carrier to not connect our calls.
Igor Hernandez wrote:
Hey guys,
I've been having a very odd problem that happens intermittently.
Hi Barry,
Thank you for the hint, but I forgot to mention that we have a few
advertisements, and we want the callers to listen to only one at a
time, and in a round robin or random order. Using Playback() doesn't
seem to serve that purpose. Is there any better way to achieve that?
Thanks.
Andy
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Andy Kuo wrote:
Hi Barry,
Thank you for the hint, but I forgot to mention that we have a few
advertisements, and we want the callers to listen to only one at a
time, and in a round robin or random order. Using Playback() doesn't
seem to serve
Hello:
when using any fct or sip to gsm gateway, is possible to transfer an
incomming call to another number automatically from asterisk
say
incoming call (gsm gateway)
answer;
mobile user dial 0XXX
then retention 4 SEND KEY
then dial XXX
then x answer
then recover first call 4 send
John Todd wrote:
5) Any summary stats on RTP packet loss, etc? (from
CHANNEL(rtpqos,audio,all)) on channels?
I wonder how to retrieve those stats:
- after Dial()?
- during Dial()? (how?)
regards
klaus
___
-- Bandwidth and Colocation Provided by
C. Chad Wallace wrote:
At 11:52 AM on 27 Aug 2009, Mat Murdock wrote:
[parkedcallstimeout]
exten = _SIP011XX,1,Answer()
exten = _SIP011XX,n,NoOp(Call Parked on: ${PARKINGSLOT})
exten = _SIP011XX,n,NoOp(This is who parked us: ${EXTEN})
exten = _SIP011XX,n,Set(PARKINGEXTEN=${PARKINGSLOT})
From:
http://www.usac.org/_res/documents/hc/pdf/training-2009/USAC-USF-overview.pdf
-The FCC now requires all VoIP telecom providers terminating and originating
to the PSTN to charge Universal Services Fund tax to end user customers.
-USF is 8.4% of revenue for VoIP Telco's.
-Last year USAC
On Thu, 2009-08-27 at 14:23 -0400, John A. Sullivan III wrote:
Hello, all. In our multi-tenant environment, we would like to be able
to use the reinvite media redirection within Asterisk for calls within a
tenant but not between tenants. We would like inter-tenant calls to be
fully proxied
John A. Sullivan III wrote:
Hope this helps someone else. Improvements, suggestions, and
constructive criticism welcome. If anyone knows why we are not getting
the expected reinvite prevention from the L() option, please let me
know. Thanks - John
Umm... it's a feature? 'reinvite
On Thu, 2009-08-27 at 16:20 -0500, Kevin P. Fleming wrote:
John A. Sullivan III wrote:
Hope this helps someone else. Improvements, suggestions, and
constructive criticism welcome. If anyone knows why we are not getting
the expected reinvite prevention from the L() option, please let me
John A. Sullivan III wrote:
grin Glad to hear of the improvement - just sorry for us. We'll use
the safest of the remaining options we can think of. Is there a better
way to do what we are trying to do? Thanks - John
No, you've covered the bases pretty well. Unfortunately I think that any
After dial.
I have put this in my hangup context as:
exten = h,1,Noop(QOS=${RTPAUDIOQOS})
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Klaus
Darilion
Sent: Thursday, August 27, 2009 13:04
To: Asterisk Users
11:09:12 AM Fotos 18/08..:
Imagens Anexadas..: DSC_0401.jpg - DSC_0402.jpg - DSC_0403.jpg
Videos Hotmail.com..: www.hotmail.com/videos.avi
_
Brrr... its getting cold out there Find someone to snuggle up with
Sticky Park sounds like somewhere you go late at night wearing a plastic
raincoat.
PaulH
Mat Murdock wrote:
My company for various reasons has asked that I come up with a way to
have previously parked calls be re-parked in the same parking slot. I
have looked at setting up asterisk so
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