Let me turn the question around slightly:
Are there any circumstances under which the h extension _won't_ get run ?
Julian
Julian Lyndon-Smith wrote:
Steve Edwards wrote:
On Sun, 29 Mar 2009, Julian Lyndon-Smith wrote:
Steve Edwards wrote:
Please show us the output from
Dear all. I have next question.
I am using SIP protocol to connect to VoIPGW. Now I need in my
extensions.conf in script to operate with phone number that is passed
to asterisk and insert it into database.
Which parameter is holding A number and can be used in extension
script? Thak you in advance
Hi,
i think the primary question here is - should it be java, activex or
flash based.
There are some implementations for each of there types out there.
Basicaly it is a simple softphone which can get configured and accessed
using javascript. So you can do the layout in html - and invoke
Is there a limit to how many manager commands you can send pipelined in
a single Asterisk management session?
Right now it seems that if we dump a bunch of requests like this (two
isn't enough though):
action: ExtensionState
exten: 906
actionid: foo-141489356-52
context: Hints
action:
My configuration is simple as below.
SIP phone - asterisk - CISCO - T1
Do you mean the hum noise is created by electric-magnetic field?
Asterisk can do nothing to eliminate it?
On Sun, Mar 29, 2009 at 2:47 AM, Steve Edwards
asterisk@sedwards.com wrote:
On Sat, 28 Mar 2009, Rilawich Ango
Christian Victor wrote:
Here in germany D-Link sells a device called the Horst-Box
Professional wich is a ADSL modem/router with WiFi and an integrated
embedded asterisk platform with 1xBRI in, 1xBRI out and 3xFXS if my mind
serves me right. Size is about 180x250x50mm. Its been around for
The TTS interface is one that I designed myself using Java. I just
call the program with the command line parameters I need. I basically
designed it to work similar to Festival's text2wave utility.
As for returning the file name, I don't know you can do it that way in
AGI. Rather, I pass
You could store the who is who information in Asterisk, so you know
thatSIP/123 is Agent/301 before logging the agent - see e.g.
http://queuemetrics.com/faq.jsp#faq-038-agent_tracking
Thanks
l.
2009/3/27 Miguel Molina mmol...@millenium.com.co
Hi all,
For those of you people that use
Hi,
From various readings, I thought that the main hurdle that kept iphones away
from asterisk were :
1. a clause in iphone Developpers agreement that forbid applications running
in background,
2. lack of sip clients.
Now it seems skype is available on iphones.
Has someone tried it ?
Along new
Rilawich Ango wrote:
My configuration is simple as below.
SIP phone - asterisk - CISCO - T1
Do you mean the hum noise is created by electric-magnetic field?
Asterisk can do nothing to eliminate it?
That would be my bet. No, Asterisk can't do anything to remove EM
noise. That's up to
Hi,
Now and then, I've got report from users asking if I could deal with :
When two persons are trying to pick the same incoming call, the second one
currently doesn't hear anything explaining him or her, the call has been
pick by someone else.
Is there anything that could be done to improve
Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
However, I am needing an FXS port integrated into a small footprint computer.
Hi,
I'm bringing this discussion here from
http://www.moythreads.com/wordpress/2007/12/24/asterisk-asynchronous-agi/
about how to manage stopping a playback on a extension previously launched
with AsyncAGI and redirecting the call to another exension.
If I make the Redirect without a playback,
Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
However, I am needing an FXS port integrated into a small footprint computer.
There are a couple SIP clients in the App store; the first one I tried
(WeePhone) supports STUN but breaks horribly when you try to connect to an
Asterisk server across a VPN connection, which is critical in our case since we
aren't exposing our Asterisk deployments to the public Internet (same
On Mon, 30 Mar 2009 14:34:31 +0100, Alan Lord (News) wrote:
Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
http://www.marvell.com/products/embedded_processors/developer/kirkwood/sheevaplug.jsp
However, I am needing
Hi Community!
If this issue was already topic, please excuse or delete my request...
Topic 1 no ringtone:
I configured a SIP registration with my SIP provider (SIPCall).
Everything works fine except the ring tone for the caller. The caller
hears silence until the called party takes up the
Hello,
Which Asterisk version are you using? I was unable to reproduce your
problem with Asterisk 1.6.0.3, also please post details about your
dial plan extensions.
Moy
On Mon, Mar 30, 2009 at 7:13 AM, Jose Arias cyr2...@gmail.com wrote:
Hi,
I'm bringing this discussion here from
2009/3/30 Lincoln King-Cliby linc...@controlworks.com
There are a couple SIP clients in the App store; the first one I tried
(WeePhone) supports STUN but breaks horribly when you try to connect to an
Asterisk server across a VPN connection, which is critical in our case since
we aren’t
I had a similar situation a while ago and the fix was setting up
indications.conf:
http://www.voip-info.org/wiki-Asterisk+config+indications.conf
-Dave
snip
I configured a SIP registration with my SIP provider (SIPCall).
Everything works fine except the ring tone for the caller. The caller
Hello
For the ringtone try progressinband=yes in sip.conf.
I don't think you can bridge do a ringback at the same time, why not
proxy the RTP and send the ringback yourself using the 'm' modifier?
Cheers
Jean-Michel.
2009/3/30, alex.mosbur...@orange-ftgroup.com
Hi David,
Thanks for the answer!
By using the h extension now I'm able to check that the Faxes are sent
successfully.
Best regards,
Santi
On Fri, Mar 27, 2009 at 4:42 PM, David Backeberg dbackeb...@gmail.com wrote:
On Tue, Mar 24, 2009 at 1:57 PM, Santiago Gimeno
santiago.gim...@gmail.com
Andreas Anderson wrote:
Hi Guys,
since about two weeks pbx_dundi.so from svn segfaults when i load it, 1.4.24
release works fine on the same box. Can someone tell me if that's something
weird with my Fedora8 system or a possible bug in svn?
Program terminated with signal 11,
D Tucny wrote:
%changelog
[snip]
awesomeness here
[/snip]
I'm speechless. This is far beyond what I could have possibly hoped for. It is
also extremely accurate.
Thank you very much for this. I'll be sure to keep this (and others) up to date
in the future.
2009/3/30 Peer Oliver Schmidt po...@theinternet.de
The Horst-Box Professional has a lot of problems in the ADSL area
(like stopping transfers after a dozen or so megabytes for example),
and I have had lots of needs to hard-reboot the box, after enabling
VoIP functionality.
Well - I never
Hello,
I've just installed the current trixbox-Version; calling from and to
outside via sipgate (germany) works well; internal calls too.
But I have set up an ring group with an external member (mobile phone)
When calling to this ring group, the CallerID of my sipgate-Account is
shown and not
Hi,
I have a task to load test a few VoIP servers, and also test our trunks on
regular intervals to see how reliable they are, i.e. how often they go down,
if at all.
I did some search and found a lot of VoIP testing tools. I selected some of
them and testing them one by one. But so far haven't
If the app is running it rings like any other softphone... if the app isn't
running calls get diverted as if the client wasn't available. For my
organization's application (primarily returning calls when out of the office/on
the road or calling a coworker to confirm some detail/information in
On Mar 30, 2009, at 9:18 AM, Zeeshan Zakaria wrote:
Hi,
I have a task to load test a few VoIP servers, and also test our
trunks on regular intervals to see how reliable they are, i.e. how
often they go down, if at all.
I did some search and found a lot of VoIP testing tools. I
On Mon, Mar 30, 2009 at 6:23 PM, Lincoln King-Cliby
linc...@controlworks.com wrote:
If the app is running it rings like any other softphone… if the app isn’t
running calls get diverted as if the client wasn’t available. For my
organization’s application (primarily returning calls when out of
There are a large number of potential sources of hum and each situation will
narrow them. The first thing would be to quantify the observation. I am
assuming it is power line frequency, although that may not be the case. It is
also useful to notice whether it is fairly pure or rich in
I have seen that include statement but s,h,i,t and T are special extensions
which are belong to the context that they are defined in. A hang up call
will not get to [questionnaire-hangup] h extension in your context. That is
my best guess based on what I know about extensions behaveyour.
If
I eventually found the problem - the h extension was getting run on the
Zap channel as soon as the bridge between the SIP client and Zap client
was broken. This is because of changes made to the cdr code in 1.4
trunk. However, the problem would not manifest itself to anyone except
those using
I'd suggest calling an echo test or playback(silence) extension on your
asterisk from the SIP phone. If there's hum, then it's almost certainly
coming from the phone (my bet would be a dodgy power supply).
Probably also worth checking there are no cellphones near either endpoint. They
seem
One of the more common embedded platforms for Asterisk is the Soekris
net5501 (or 4501 if you don't need as much processing power)
Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for almost the
same money (Soekris stuff isn't cheap in the UK) and is about the same
footprint, it
On Mon, Mar 30, 2009 at 12:18:09PM -0400, Zeeshan Zakaria wrote:
Hi,
I have a task to load test a few VoIP servers, and also test our trunks on
regular intervals to see how reliable they are, i.e. how often they go down,
if at all.
I did some search and found a lot of VoIP testing tools.
On March 30, 2009 12:48:59 pm randulo wrote:
Except for roaming and in particular international roaming, isn't the
best plan to forward calls the iPhone. It is a phone, too isn't it? Or
just a game platform, browser and GPS?
That's pretty much what I do; I use siax (I have a jailbroken iPhone)
I've been quite satisfied with one of these:
http://www.pikatechnologies.com/english/View.asp?x=652
On 03/26/2009 5:28 PM, Anthony Plack wrote:
Hey all,
I have a potential project which calls for a very small form-factor computer
like this:
Yes, you make a good point. Electromagnetic fields are another source of
ingress, whether from a nearby cell phone or by being located a mile away from
a 50 KW AM radio transmitter (etc.).
one does wonder why there's such inadequate shielding
As a ham radio operator, I can say that has been
There are no sip packets created to the hard phone (I'm using a softphone
and those sip packets are there)
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 da...@safedatausa.com
-Original Message-
From:
On Mon, 30 Mar 2009, Julian Lyndon-Smith wrote:
Thanks for all the help and pointers - Steve, I'm getting to like
templates ;)
Congratulations on finding your solution.
I discovered templates after a couple of years of coding dialplans and
fell in love.
In case they're of any use, here are
If it makes a difference the phone is a gxp 2000
Thanks,
David Ruggles
CCNA MCSE (NT) CNA A+
Network EngineerSafe Data, Inc.
(910) 285-7200 da...@safedatausa.com
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com]
On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote:
One of the more common embedded platforms for Asterisk is the Soekris
net5501 (or 4501 if you don't need as much processing power)
Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for
almost the same money
Tzafrir Cohen wrote:
On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote:
One of the more common embedded platforms for Asterisk is the Soekris
net5501 (or 4501 if you don't need as much processing power)
Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for
Tzafrir Cohen wrote:
On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote:
One of the more common embedded platforms for Asterisk is the Soekris
net5501 (or 4501 if you don't need as much processing power)
Agreed. Though, given the Asus eeeBox (1.6Ghz Atom) can be had for
Sorry, forgot a link
http://www.eeextra.com/eee/eeebox-specs.html
Singer XJ Wang wrote:
Tzafrir Cohen wrote:
On Mon, Mar 30, 2009 at 06:20:20PM +0100, Chris Bagnall wrote:
One of the more common embedded platforms for Asterisk is the Soekris
net5501 (or 4501 if you don't need as much
Hi all
If Asterisk is bridging a call between two SIP peers and one peer puts
the other on hold by means of a re-INVITE with SDP containing
a=sendonly, Asterisk will play locally generated MOH instead of
relaying the media streamed by the SIP peer which took the hold
action.
Any ideas how to
On Mon, Mar 30, 2009 at 09:04:04PM +0300, Tzafrir Cohen wrote:
The SheevaPlug also seems an interesting option. An SDK sells for 100$
(well, 99$, but who counts?). I hope to get one soon and see how well it
performs.
http://www.globalscaletechnologies.com/t-sheevaplugdetails.aspx
Hi,
Some ATAs (SPA3102, M-ATA, ...) have a long local FXS settings list such as
:
FXS port gain,
Ring Waveform
Frequency
...
1. My understanding of these is that those settings define how calls coming
from SIP side, trigger a signal which will in turn, ring analog device.
is this correct ?
2.
Richard Brady wrote:
If Asterisk is bridging a call between two SIP peers and one peer puts
the other on hold by means of a re-INVITE with SDP containing
a=sendonly, Asterisk will play locally generated MOH instead of
relaying the media streamed by the SIP peer which took the hold
action.
2009/3/30 Olivier oza-4...@myamail.com
Hi,
Some ATAs (SPA3102, M-ATA, ...) have a long local FXS settings list such as
:
FXS port gain,
Ring Waveform
Frequency
...
1. My understanding of these is that those settings define how calls coming
from SIP side, trigger a signal which will in
Regarding compression with g.729/gsm/etc. and Asterisk
If we convert all the voice files to the corresponding format g.729/gsm/etc.
and we send digits using RFC 3261 and we do not need silence detection, is
there still a need to decompress the media stream ?
If doable how to make sure this
i've been playing with 1.6 voicemail w/ IMAP storage. it
seems to work fine. however once IMAP storage is enabled.
everyone VM will use IMAP. is there a way to configure
some users use IMAP and other users use traditional
file base storage?
--
Edwin Lam edwin@officegeneral.com
Systems
On Monday 30 March 2009 03:45:19 pm Edwin Lam wrote:
i've been playing with 1.6 voicemail w/ IMAP storage. it
seems to work fine. however once IMAP storage is enabled.
everyone VM will use IMAP. is there a way to configure
some users use IMAP and other users use traditional
file base storage?
Hello,
This is my first asterisk installation, and having read up on the
documentation, and trying several tutorials i'm unable to get my
outbound route working. I'm certain it's an issue with my configuration
and my inexperience with asterisk. So i have my POTS phone connected to
my digium
Howdy,
Was there ever a fix for this?
I have Trix 2.6 running asterisk 1.4 and have to set an extension with
call-limit=1. However that user can no longer do attended transfers from Linkys
962 ip phone.
Is there anyway around this?
Cheers,
Taff..
On Mon, Mar 30, 2009 at 5:16 PM, Bruce Thayre br...@mipscomputation.comwrote:
Up to this point, all i have set up are two SIP phones, my POTS phone,
and 1 ring group. My POTS line is connected to channel 1, and my POTS
phone is connected on channel 3. Perhaps my understanding of how the
Show us your dialplan.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Bruce Thayre
Sent: Monday, March 30, 2009 4:16 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Newbie trying to make calls
carl Lougher wrote:
Howdy,
Was there ever a fix for this?
I have Trix 2.6 running asterisk 1.4 and have to set an extension with
call-limit=1. However that user can no longer do attended transfers from
Linkys 962 ip phone.
Is there anyway around this?
Cheers,
Taff..
Yes, set
Thank you for the prompt input! My extension.conf can be viewed here:
http://dpaste.com/21356/
I'm currently doing the configuration through the GUI bundled with the
trixbox distro, and i'm not entirely sure where it stores all of the
changes as i haven't seen the changes to extension.conf that i
Hi all.
I received a PAP2T-NA from a potential customer to see if I could get it
configured for testing. I plugged it into my network and plugged a phone
into it and attempted to do a factory reset from the handset.
I pressed and got NOTHING! Just silence. So, is this TA a brick? Or
We use call-limit set to 1 for hints. I guess i'll look into the dtmf method
and debug the linksys phone to see what it uses for attended transfers.
Cheers
--- On Mon, 30/3/09, Mark Michelson mmichel...@digium.com wrote:
From: Mark Michelson mmichel...@digium.com
Subject: Re:
I think the comment was more along the lines of use call-limit, but put a
number higher than 1.
Mike
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of carl Lougher
Sent: Monday, March 30, 2009 21:21
To:
Probably not. There is an option to turn off access to the IVR (fairly
important if you have it installed somewhere in production). Sniff the
packets coming out of it to see if you can determine its IP, but I am
guessing if the previous owner already disabled the IVR, they probably
locked
Yes, it is an NA. So, I can assume that the ivr has been disabled. Does
anyone know how to do a COMPLETE factory reset on it?
Mike.
On Monday 30 March 2009 20:08:43 Jeff LaCoursiere wrote:
Probably not. There is an option to turn off access to the IVR (fairly
important if you have it
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