Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
--
_
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New
Please, show your dial plan and name your Asterisk version. You might be call the Dial
application with incomplete arguments.
jg
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
--
On Thu, 3 Jul 2014, Andrew Colin wrote:
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c: Don't know what to do with 'SIP/
1) Please choose a more meaningful subject. Lots of errors can be
considered strange. (Note that actually, this is a
Sound like chan_sip was not build.
Just a guess: check that openssl-dev is available
---
Dennis Guse
On Thu, Jul 3, 2014 at 12:01 PM, Andrew Colin and...@vsave.co.za wrote:
Hi Guys,
Does anyone know what this error means and how to fix it?
[Jul 3 11:57:27] WARNING[17040] pbx.c:
Hi Guys,
Anyone ever seen this before.
on asterisk 1.8 if i set one of my pabx extensions to show private
number and send a call over VoIP with g729 the call fails but with alaw
it works.
If i enable the callerid on g729 it also works
see error below
From:
On 25/09/13 15:42, Andrew Colin wrote:
Hi Guys,
Anyone ever seen this before.
on asterisk 1.8 if i set one of my pabx extensions to show private
number and send a call over VoIP with g729 the call fails but with
alaw it works.
If i enable the callerid on g729 it also works
see error below
...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark
Sent: Wednesday, April 28, 2010 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Strange Error -- ASterisk 1.6
All,
I just noticed this in my logs, and am rather lost as to what
All,
I just noticed this in my logs, and am rather lost as to what module
it pertains to. I would assume pseudo-realtime priority for the process,
but I am looking for a little confirmation from the group:
[Apr 28 12:28:36] WARNING[20773] asterisk.c: The canary is no more. He
has
: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Seann Clark
Sent: Wednesday, April 28, 2010 2:30 PM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] Strange Error -- ASterisk 1.6
All,
I just noticed this in my logs, and am
I found no info about this strange error:
logger.c: No more room in scheduler
logger.c: Asked to delete sched id -1???
Only in verbose mode. Someone know how to solve this?
Asterisk 1.2.13 with sangoma A104EC
Hints?
Thnks.
begin:vcard
fn:Massimo Nuvoli
n:Nuvoli;Massimo
org:Progetto Archivio
Does this occur in the latest 1.2.17 release?
On 4/10/07, Massimo Nuvoli [EMAIL PROTECTED] wrote:
I found no info about this strange error:
logger.c: No more room in scheduler
logger.c: Asked to delete sched id -1???
Only in verbose mode. Someone know how to solve this?
Asterisk 1.2.13 with
Sean Bright ha scritto:
Does this occur in the latest 1.2.17 release?
I dont know, this is a production system with 2 pri linked to telco
and 2 pri linked to a pbx, i planned a large update but the release
in use is the 1.2.13.
And, also, i checked the changelog of the 1.2.17, and i found no
Someone know why my asterisk gives me the following msgs?
Thank you
- Got SIP response 603 Declined (no dialog) back from
X.X.X.Xhttp://82.51.224.34/
-- Got SIP response 603 Declined (no dialog) back from
X.X.X.Yhttp://82.51.224.34/http://82.104.4.192/http://82.104.4.192/
-- Got SIP
Dear All,
After doing the test everything went fine, Thanks Anthony for putting
me on the right direction.
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Anthony,
I believe you where right the dial plan seems to have been missing the
TRUNK= statement and I found one in the file extensions.conf but
: [asterisk-users] Strange
error
Hi Friends,We are using "Asterisk" in our office and
using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It
is working fine. But, sometime, when i am trying to make a call to USA, my
softpho
Dear Anthony,
The dial plan is currently very simple it should pick up any call
and send it to a sip phone registered, you can see the context below named
zap-in is what I am using, it is only that and nothing more, is there something
extra I have to add to dial plan or to that context ?
Thx
MAG
Dear Anthony,
I believe you where right the dial plan seems to have been missing the
TRUNK= statement and I found one in the file extensions.conf but not the
correct group I configured so I changed it and will test again.
Thx
MAG
"Mohamed A. Gombolaty" wrote:
Dear Anthony,
The dial plan is
Hi Friends, We are using "Asterisk" in our office and using "XLite" as softphone and "Teliax" service for USA dialing. Sometimes It is working fine. But, sometime, when i am trying to make a call to USA, my softphone is telling that "I am sorry. That is not a valid extension. Please try again.
Dear All,
I have a strange problem in recieving calls on the pri the zaptel
is green and everything seems very well, but when a call comes I can see
the call along with the caller ID but then I get this strange message which
make the call hungup:
error msg: 'zap-in' from '0109687348' does not
- Original Message -
From: Mohamed A. Gombolaty
[mailto:[EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent:
Wed, 26 Jul 2006 18:40:07 -0300
Subject: [asterisk-users] Strange Error when
calling
Dear All,
Greetings.
I have a strange problem in recieving calls on the pri
This looks like a dialplan problem - do you have a match for
0109687348 in the zap-in context in your dialplan?
A.
On 26-Jul-06, at 2:40 PM, Mohamed A. Gombolaty wrote:
Dear All,
I have a strange problem in recieving calls on the pri the zaptel
is green and everything seems very well,
Has anyone seen this before ?
Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we
dont know about.
I've got a whole load of them (328 in the last 5 minutes ...)
Julian
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Asterisk wrote:
Has anyone seen this before ?
Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call we
dont know about.
I'm guessing that happens when asterisk has hung up on some device but
that device hasn't figured it out yet(therefore it's still trying to
talk back to
Replies inline:
Andrew Thompson wrote:
Asterisk wrote:
Has anyone seen this before ?
Feb 15 18:37:34 DEBUG[866]: That's odd... Got a response on a call
we dont know about.
I'm guessing that happens when asterisk has hung up on some device but
that device hasn't figured it out yet(therefore
Hmm,
Part of the show channels gives me this:
SIP/6028-b1ff (AgentQ s1 ) Up (None)(None)
SIP/6011-da2f (AgentQ s1 ) Up (None)(None)
SIP/6019-8fe7 (AgentQ s1 ) Up (None)(None)
SIP/6019-f866 (AgentQ
Hello all,
I have a Linux Box
running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b channel driver for H323. All
installation and packages compilation was successful. I have a SIP account to a
SIP provider and I use it for outgoing calls. Im using Cisco ATA boxes
both SIP and H323, and all
you're using out of date and buggy versions of * and oh323.
try to update them and check if the error is occurring again.
On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa
[EMAIL PROTECTED] wrote:
Hello all,
I have a Linux Box running Asterisk-1.0-RC2 and asterisk-oh323-0.6.3b
PROTECTED] On Behalf Of Daniel Eboa
Sent: 12 November 2004 12:16
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Strange
error
Hello all,
I have a Linux Box running Asterisk-1.0-RC2 and
asterisk-oh323-0.6.3b channel driver for H323. All installation and packages
compilation was successful. I have
Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Strange error
you're using out of date and buggy versions of * and oh323.
try to update them and check if the error is occurring again.
On Fri, 12 Nov 2004 18:16:23 +0100, Daniel Eboa
[EMAIL PROTECTED] wrote:
Hello
hi,
i write this looking for free conference room, i
checl code and don´t see any error but die at priority 7 if room 1001 have users
in
exten =
_1NXXNXX,1,RouteCall(${EXTEN})exten =
_1NXXNXX,2,GotoIf($[${DESTINATION1:0:3} = CONF]?3:13)exten =
_1NXXNXX,3,Setvar,var=0exten =
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