Although there's no requisite mention of ${Horrible_Dictator}, can't
we pretend there was, call a Godwin and kill this subject?
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us fo
hi guys,
i have a problem with 1.8 branch no matter which release of 1.8 i'm
using. i can't make any sip calls, this is the error message i get on
each call:
[Jan 18 19:02:15] ERROR[1698] rtp_engine.c: No RTP engine was found.
Do you have one loaded?
[Jan 18 19:02:15] ERROR[1698] chan_sip.c: Got S
On Tue, 18 Jan 2011, abhinav anand wrote:
The exact error thrown on Asterisk CLI is "chan_sip.c:20039
handle_request_invite: Call from [IMSI310410270465840] to extension
"2103" rejected because extension not found"
What context does 'sip show user IMSI310410270465840' show?
What does 'dialpl
Is there a way to make ConfBridge hang up on the final participant in a
conference (obviously after some sort of initial grace period)?
Background - I have just moved all of the phones in my house to separate
extensions. As a replacement for the POTS-style shared line, I have
implemented a "barge
Paul Belanger wrote:
> Moderation would be another option (personally opinion). Regardless,
> we should all now be aware of the rules [1] of the mailing lists.
> All we can do now is hope people respect them.
>
> [1] http://www.asterisk.org/community/rules
>
> --
> Paul Belanger
> Digium, Inc.
On 11-01-18 08:52 PM, abhinav anand wrote:
> I have tried all the methods suggested by others in the Asterisk User
> community but still the problem remains same. If anybody knows the solution
> to this
> one, please let me know.
>
Which context is your incoming calls using? When you know that, yo
Hi All,
I am using Asterisk for one of my projects in OpenBTS. I am having the age
old problem of "extension not found" when try to make
a call from one registered SIP phone to other registered SIP phone (two
mobile phones connected to Asterisk via OpenBTS).
The exact error thrown on Asterisk CLI
Hi everyone,
We have an Asterisk 1.4.17 user who has problems with sometimes not getting
a ring tone on the calling phone.
We're considering setting progressinband = yes, but would like to know how
much extra CPU load this will require? If anyone can give something even
roughly specific (eg "30%
On 11-01-18 07:42 PM, Chad Wallace wrote:
> We need to ban all versions of outlook until microsoft decides to fix
> it.
>
Moderation would be another option (personally opinion). Regardless, we
should all now be aware of the rules [1] of the mailing lists. All we
can do now is hope people respect
On 1/18/11 6:55 PM, sean darcy wrote:
On 01/18/2011 05:27 PM, Shaun Ruffell wrote:
On 01/18/2011 04:06 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Here's a call comin
On Jan 18, 2011, at 6:42 PM, Chad Wallace wrote:
> We need to ban all versions of outlook until microsoft decides to fix
> it.
Amen.
Chris
--
-
Chris Owen - Garden City (620) 275-1900 - Lottery (noun):
President
On 1/18/11 12:01 AM, Michelle Dupuis wrote:
> We have an application that plays a variety of sound files on one leg of
> a call (generated by a call file). We've been told that the party
> listening to the audio files intermittantly hears "robotic" sounding
> audio (on/off during the same call).
>
On 01/18/2011 05:27 PM, Shaun Ruffell wrote:
On 01/18/2011 04:06 PM, Danny Nicholas wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Tuesday, January 18, 2011 3:58 PM
To: asterisk-users@
On Tue, 18 Jan 2011 18:17:31 +0200
Tzafrir Cohen wrote:
> It is interesting to note that your mailer (MS-Outlook) has very bad
> support for threading. In fact, it (combined with the MS-Exchange
> server) does not really bother reproducing the mail headers that are
> required to keep the proper t
Tom Rymes wrote:
On 01/18/2011 10:18 AM, Andrew Thomas wrote:
Why do I top post? Simple. I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to read them all over again on the last one?
OK, this is a stupid thread, nobody
Hello!
I have just upgraded to asterisk 1.8.2.1 and see some weird messages
in log when client tries to register:
[2011-01-19 00:52:47] WARNING[25624] chan_sip.c: Failed to parse contact info
[2011-01-19 00:52:50] NOTICE[25624] chan_sip.c: Peer '0010101' is now
UNREACHABLE! Last qualify: 105
[20
On 01/18/2011 04:06 PM, Danny Nicholas wrote:
> -Original Message-
> From: asterisk-users-boun...@lists.digium.com
> [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
> Sent: Tuesday, January 18, 2011 3:58 PM
> To: asterisk-users@lists.digium.com
> Subject: [asterisk-
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of sean darcy
Sent: Tuesday, January 18, 2011 3:58 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] 1.8.2: dahdi-2.4: calls dropping
Here's a call com
On Tue, Jan 18, 2011 at 1:21 PM, Steve Totaro
wrote:
> I got it fixed with an all nighter, but I took a beating for the
> problems for not fully testing and monitoring. After that, nobody had
> faith in the fax solution.
So is FFA working for you now? What did you have to do to fix it (I
like
Here's a call coming in over PSTN to dahdi/4, connected to a local
extension dahdi/1:
-- Executing [s@incoming-pstn-line:1] Answer("DAHDI/4-1", "") in
new stack
..
-- Executing [s@incoming-pstn-line:6] Dial("DAHDI/4-1",
"DAHDI/g0,36") in new stack
-- Called g0
-- DAHD
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
Log when dialing "0924343424"
[snip]
A normal internal call to "2000" is:
[snip]
These two calls do not demonstrate your issue:
1-If user dial "012345" there is an error and the call isn't made and
the error is "handle_request_invite: Ca
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos
Flausino
Sent: Tuesday, January 18, 2011 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Calling rule
On 01/18/2011 3:20 PM, Vitor Carlos Flausino wrote:
== Spawn extension (DLPN_DialPlan1, 0924343424, 1) exited non-zero on
'SIP/6005-0002'
Vitor,
Can you please clarify whether the "0" should be received by Asterisk
and processed internally, or whether it should be passed to the DAHDI
On 01/18/2011 3:21 PM, Steve Totaro wrote:
If you are swapping out systems in really busy offices that rely on
faxing to keep the doors open, do a whole bunch of testing.
I have no experience with Digium's FFA, beyond installing it and
receiving a fax or two. So I can't really agree or disagr
On Tue, Jan 18, 2011 at 9:02 AM, Kevin P. Fleming wrote:
> On 01/16/2011 09:18 PM, Jeremy Kister wrote:
>>
>> On 1/16/2011 4:13 PM, Paul Belanger wrote:
>>>
>>> I don't believe Digium is blind to its users: Users of Free Fax For
>>> Asterisk are not entitled to any Digium technical support [1].
>>
- Original Message -
> From: "Steve Edwards"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, January 18, 2011 8:54:11 PM
> Subject: Re: [asterisk-users] Calling rules
> >> On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
>
> >>> 1-If user dial "0123
On Mon, 17 Jan 2011 18:01:14 -0500
Michelle Dupuis wrote:
> We have an application that plays a variety of sound files on one leg
> of a call (generated by a call file). We've been told that the party
> listening to the audio files intermittantly hears "robotic" sounding
> audio (on/off during t
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
1-If user dial "012345" there is an error and the call isn't made and
the error is "handle_request_invite: Call from 'XXX' to extension
'012345' rejected because extension not found in context
'DLPN_DialPlanX'. 2-If user dials "0" waits for th
- Original Message -
> From: "Steve Edwards"
> To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>
> Sent: Tuesday, January 18, 2011 8:06:47 PM
> Subject: Re: [asterisk-users] Calling rules
> Un-top-posting and discarding cruft...
>
> On Tue, 18 Jan 2011, Vitor Carlos Flau
>> I've been working with computers for over 40 years and don't have the
>> foggiest notion how the "Green Day--Wake Me Up When September Ends" video
>> applies to Top Posting.
>>
>It's a reference to the Everlasting September in 1993. AOL added
>usenet access to its service, unleashing a horde o
>
> I've been working with computers for over 40 years and don't have the
> foggiest notion how the "Green Day--Wake Me Up When September Ends" video
> applies to Top Posting.
>
It's a reference to the Everlasting September in 1993. AOL added
usenet access to its service, unleashing a horde of di
On 01/18/2011 10:18 AM, Andrew Thomas wrote:
Why do I top post? Simple. I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to read them all over again on the last one?
OK, this is a stupid thread, nobody is going to be convi
> On Tuesday 18 Jan 2011, Don Kelly wrote:
> > "PLONK" is "retro"--like bottom-posting :)
> >
> > --Don
> boun...@lists.digium.com] On Behalf Of A J Stiles
> Retro? For those of us who actually know what PLONK means, it's
hilarious.
> Now, here is a link
> http://www.youtube.com/watch?
Now this thread is really starting to annoy me
Dieses Video enthält Content von WMG. Es ist in
deinem Land nicht verfügbar.
..for non
Un-top-posting and discarding cruft...
On Tue, 18 Jan 2011, Vitor Carlos Flausino wrote:
Users, have to dial "0" to get an external line, and afterwords the
number they want to dial (exe 12345). The thing is:
1-If user dial "012345" there is an error and the call isn't made and
the error is
My dial plan was generated by asterisk GUI, and the line is:
exten = _0,1,Macro(trunkdial-failover-0.3,${trunk_1}/${,${EXTEN:1})},,trunk_1,)
where trunk_1 "is DAHDI/1"
Notice the difference between your "0." and my "_0".
Is "mine" correct?
Best regards,
-vcf
- Original Message -
From:
On Tuesday 18 Jan 2011, Don Kelly wrote:
> "PLONK" is "retro"--like bottom-posting :)
>
> --Don
Retro? For those of us who actually know what PLONK means, it's hilarious.
The fact that some people *don't* know what it means only makes it doubly so.
Now, here is a link that those of us who reme
On Tuesday 18 January 2011 11:31:07 Ira wrote:
> At 01:00 AM 1/18/2011, you wrote:
> >On Tuesday 18 January 2011 01:05:20 Ira wrote:
> > > I have tried installing many of the beta versions and most of the
> > > release versions of 1.8. I have 3 SIP phones which we use for all
> > > our calls. After
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Vitor Carlos
Flausino
Sent: Tuesday, January 18, 2011 12:10 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Calling rules
Hello.
I don't know if
Although I put my e-mail in /etc/hylifax/Dispatch I can't receive.
Flavio Roberto Miranda
It may be different for your Hylafax version, etc., but you may want your
email in
/var/spool/hylafax/etc/FaxDispatch
And you probably want to post your questions to the Hylafax list
http://lists.
Hello.
I don't know if this is a problem, but I was expecting a different behavior.
Users, have to dial "0" to get an external line, and afterwords the number they
want to dial (exe 12345). The thing is:
1-If user dial "012345" there is an error and the call isn't made and the error
is "handle
Hi all,
I know Hylafax is an application and not Asterisk but I'd like to post a
problem found in configuring such application and Asterisk.
I am able to reveive fax,but , I can't receive it in e-mail. Although I put my
e-mail in /etc/hylifax/Dispatch I can't receive.
Anybody know where I
I am having a problem trying to use originate from the CLI on Asterisk
1.8.1.1. The SIP peer is defined correctly and it works if I dial using
my IP phone. When I try to dial from the CLI I get this message:
[Jan 18 12:00:09] WARNING[3336]: chan_sip.c:19048
handle_response_invite: Receiv
Thank you so much for your response I will try this operation and I will
update you as soon as I have any result
2011/1/18 A J Stiles
> On Tuesday 18 Jan 2011, salaheddine elharit wrote:
> > yes i want to know how can i do in order to read this files using apche
>
> Either make a symbolic link
At 01:00 AM 1/18/2011, you wrote:
On Tuesday 18 January 2011 01:05:20 Ira wrote:
> I have tried installing many of the beta versions and most of the
> release versions of 1.8. I have 3 SIP phones which we use for all our
> calls. After installing 1.8 the first thing I try is calling out port
> on
On 01/18/2011 10:53 AM, Jeff LaCoursiere wrote:
On Tue, 18 Jan 2011, Asterisk Security Team wrote:
Asterisk Project Security Advisory - AST-2011-001
Product Asterisk
Summary Stack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
Susceptibility Remot
On Tue, 18 Jan 2011, Asterisk Security Team wrote:
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
Susceptibility
The Asterisk Development Team has announced security releases for the following
versions of Asterisk:
* 1.4.38.1
* 1.4.39.1
* 1.6.1.21
* 1.6.2.15.1
* 1.6.2.16.1
* 1.8.1.2
* 1.8.2.1
These releases are available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk/release
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
SEE THE BOTTOM :P
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tzafrir
Cohen
Sent: 18 January 2011 16:18
To: asterisk-users@lists.digium.com
Subject: Re: [asterisk-users] Top Posting
On Tue, Jan 18, 2011 at
On Tue, Jan 18, 2011 at 03:18:49PM -, Andrew Thomas wrote:
> Why do I top post? Simple. I read every message in the thread - and if
> there are 10 messages (for example) in that thread - then why should I
> have to read them all over again on the last one?
You mean: why should I have to read
I'm top-posting this simply to be consistent with the previous couple posts.
I agree that top-posting is preferable for the reason that Andrew pointed
out and I prefer no trimming (other than signatures--especially legal
disclaimers, etc.) so I can delete every message except the most recent and
m
"I also agree this is a pointless discussion because, clearly, nobody is
willing to budge, and it has nothing to do with Asterisk."
Amen :)
It may yet have a point - another few hundred (thousand) of these and the
board will blacklist items with the words "top post" and "bottom post" :)
And mayb
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Andrew Thomas
Sent: Tuesday, January 18, 2011 10:01 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top Posting
"I als
"I also agree this is a pointless discussion because, clearly, nobody is
willing to budge, and it has nothing to do with Asterisk."
Amen :)
[oh no, a bottom post]
If you have received this communication in error we would appreciate
you advising us either by telephone or return of e-mail. The c
Hi
I am researching if there is a practical number of SIP accounts that
Asterisk can register against as a UA. I have an idea for a project
but it would need to register multiple accounts from multiple
providers to work.
Regards
Jon
--
Jon Farmer
Tel 07795 118140
--
_
On Tue, Jan 18, 2011 at 8:18 AM, Andrew Thomas wrote:
> Why do I top post? Simple. I read every message in the thread - and if
> there are 10 messages (for example) in that thread - then why should I
> have to read them all over again on the last one?
That's not the alternative (having ten mess
On Tuesday 18 Jan 2011, salaheddine elharit wrote:
> yes i want to know how can i do in order to read this files using apche
Either make a symbolic link to the location of the files from somewhere Apache
knows about, using something like
# ln -s /path/to/files /path/to/webroot/mp3files/
and set i
Hello List,
To whom it might concern:
I have been working in some SlackBuilds (script for making Slackware
Packages) for my personal use, but thought they might be useful for
someone else here.
Beside of the exceptional distributions used so far (CentOS, Debian,
Ubuntu, etc.), you might wa
On Tue, 2011-01-18 at 15:18 +, Andrew Thomas wrote:
> Why do I top post? Simple. I read every message in the thread - and if
> there are 10 messages (for example) in that thread - then why should I
> have to read them all over again on the last one?
>
> Top posting is here - to stay!
>
> St
I'm top posting this so you will see it and if you don't understand it,
look it up.
PLONK!!
On Tue, 18 Jan 2011, Andrew Thomas wrote:
Why do I top post? Simple. I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to re
Why do I top post? Simple. I read every message in the thread - and if
there are 10 messages (for example) in that thread - then why should I
have to read them all over again on the last one?
Top posting is here - to stay!
Stop being so anal and 'retro'. Bottom posting belongs in forums - top
Thanks for the info. I did get it working without any SIP Proxy. There is a
bug in pfSense v1.2.3 where certain configs are not removed and
some inconsistencies exist in the xml config file. Once I cleaned that and
when I limited my Asterisk servers to use different port ranges for UDP
traffic now
On 11-01-18 04:22 AM, Andrew Thomas wrote:
> Top posting? Who cares? Get a life!
>
Clearly not you, so why both even replying? At worst case it is just
redundant information for people, best case somebody reads the email
thread at starts bottom posting. I suggest taking a moment and
re-reading
>-Original Message-
>From: asterisk-users-boun...@lists.digium.com
>[mailto:asterisk-users-boun...@lists.digium.com] On >Behalf Of Carlos Chavez
>Sent: Saturday, January 15, 2011 2:02 AM
>To: Asterisk
>Subject: [asterisk-users] Asterisk stops responding
>
> I am having a problem with
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Tuesday, January 18, 2011 6:25 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] how to read mp3
yes i want to know ho
On 01/16/2011 09:18 PM, Jeremy Kister wrote:
On 1/16/2011 4:13 PM, Paul Belanger wrote:
I don't believe Digium is blind to its users: Users of Free Fax For
Asterisk are not entitled to any Digium technical support [1].
I'm not looking for technical support; I'm just looking for a way to
report
Hello Bruce,
Sorry for the delay. I don't really have time to follow this list much.
In your original setup, you did use a sort of SIP Proxy (the central Asterisk
feeding the others) depending on your definition. A SIP Proxy would probably
solve your issue, but as I stated in my previous mail,
yes i want to know how can i do in order to read this files using apche
2011/1/17 Steve Edwards
> On Mon, 17 Jan 2011, salaheddine elharit wrote:
>
> i have asterisk installed in our call centre and I have all the clients
>> conversation saved in this file
>>
>> /usr/apache-tomcat-5.5.17/webapps
On 01/18/2011 04:41 AM, ishagh ouldbah wrote:
Good morning
My situation is as folowing
I have a numer that connect to my asterisk
I configured another phone to transfer to this number
So when somebody call me he will be transffered to the number which
asterisk connect to
i.e my asterisk connected
Good morning
My situation is as folowing
I have a numer that connect to my asterisk
I configured another phone to transfer to this number
So when somebody call me he will be transffered to the number which asterisk
connect to
i.e my asterisk connected phone is not the originated number
My questi
Something that often gets forgotten is the on-site LAN infrastructure as well.
It could be a bad/faulty switch, rubbish cabling, induced interference etc.
etc. all at the customers premises.
Maybe a handset plugged directly in to the back of the router, before it hits
the LAN would tell you whe
Top posting? Who cares? Get a life!
Now - can we get back to Asterisk et al?
Thanks!
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mark
Murawski
Sent: 18 January 2011 02:57
To: asterisk-users@lists.digium
On Tuesday 18 January 2011 01:05:20 Ira wrote:
> I have tried installing many of the beta versions and most of the
> release versions of 1.8. I have 3 SIP phones which we use for all our
> calls. After installing 1.8 the first thing I try is calling out port
> one of my Digium TDM04 back into port
enable sip debug and check which error or error code you are getting
also try nat=yes
On Mon, Jan 17, 2011 at 5:34 PM, Thomas Perron wrote:
> Thanks. I fixed that.
> Still does not work.
>
>
> On Mon, Jan 17, 2011 at 12:53 AM, Jeroen Eeuwes
> wrote:
> > Hi Thomas,
> >
> >> register => 999:999
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