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 thomas.per...@gmail.comwrote:
Thanks. I fixed that.
Still does not work.
On Mon, Jan 17, 2011 at 12:53 AM, Jeroen Eeuwes jeroeneeu...@gmail.com
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
one of my Digium TDM04 back into port 2.
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:
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
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
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
yes i want to know how can i do in order to read this files using apche
2011/1/17 Steve Edwards asterisk@sedwards.com
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
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
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
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
-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 an
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
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
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
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
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!
Stop
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
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
On Tue, Jan 18, 2011 at 8:18 AM, Andrew Thomas a...@datavox.co.uk 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
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
--
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
-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
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 maybe If
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
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
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
Asterisk Project Security Advisory - AST-2011-001
ProductAsterisk
SummaryStack buffer overflow in SIP channel driver
Nature of Advisory Exploitable Stack Buffer Overflow
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
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
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
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
one
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 asterisk_l...@earthshod.co.uk
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
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:
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
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
snip
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
-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
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 installing
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 remember a
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: Danny
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
Now this thread is really starting to annoy me
Dieses Video enthlt Content von WMG. Es ist in
deinem Land nicht verfgbar.
..for non
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?v=R1JXYgwwDeY
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
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 dirty,
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 dirty,
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 18, 2011 8:06:47 PM
Subject: Re: [asterisk-users] Calling rules
Un-top-posting and discarding
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 the
On Mon, 17 Jan 2011 18:01:14 -0500
Michelle Dupuis mdup...@ocg.ca 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
- Original Message -
From: Steve Edwards asterisk@sedwards.com
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Tuesday, January 18, 2011 8:54:11 PM
Subject: Re: [asterisk-users] Calling rules
On Tue, 18 Jan 2011, Vitor Carlos
On Tue, Jan 18, 2011 at 9:02 AM, Kevin P. Fleming kpflem...@digium.com 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
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
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
-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
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: Call
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
-- DAHDI/1-1
On Tue, Jan 18, 2011 at 1:21 PM, Steve Totaro
stot...@asteriskhelpdesk.com 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
-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
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:
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
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
On Tue, 18 Jan 2011 18:17:31 +0200
Tzafrir Cohen tzafrir.co...@xorcom.com 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
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:
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 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 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
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
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%
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
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, you
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. |
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
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 'dialplan
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
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 --
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