TIMEOUT function.
Thanks again,
Nenad
On Wed, Nov 30, 2022, 19:03 wrote:
>
> On Wed, Nov 30, 2022 at 5:57 AM Nenad Radosavljevic wrote:
>
> > Hello everyone,
> >
> > Does anyone know is it possible to cancel the call duration limit set in
> > app Di
Hello everyone,
Does anyone know is it possible to cancel the call duration limit set in
app Dial with options S(x) or L(x[:y[:z]]), by for instance entering custom
feature code (application map) during the call ?
I have read somewhere that bridge features can be set using the
${BRIDGE_FEATURES}
Thanks Joshua !
I was not aware that extension's hint in dialplan can have additional data
in the parentheses - it might be worth adding that info to wiki.
On Wed, Jun 3, 2020 at 10:08 AM Nenad Radosavljevic
wrote:
>
> > Hello !
> >
> > Is there something wrong with
SIP/10
- second noop prints: DEVNAME is
Way I understand the documentation for function HINT, I would expect the
DEVNAME variable to be set to "Test extension".
Any "light" on this topic would be much appreciated. Thanks.
(or something) CDR record for the call.
Does anyone knows the way to make this work, or knows the alternate
way to find out file name of auto-monitored call and add it to a
CDR(userfiled) on Asterisk 1.2.16(or 17) ?
Thanks for any info on this issue !
--
Kind regards,
Nenad
-swich.org site.
Regards,
Nenad Radosavljevic
___
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
, peerstate Null
NEW_HANGUP DEBUG: Destroying the call, ourstate Null, peerstate Null
-
Does anyone have any idea why this is happening, and how to solve this
problem ?
Regards,
Nenad Radosavljevic
would be nice info also) ?
Thank you very much.
Nenad Radosavljevic
___
--Bandwidth and Colocation sponsored by Easynews.com --
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
Hi !
First of all thank you all for fast response on matter of T.38 capable ATAs.
I have asked a UK VoIP suplier to check with manufacterers of various ATAs
they sell, do they support T.38 and here is what they/I have got as a
result:
1. Sipura SPA-2100 only and with firmware 3.2.1 is T.38
in my case.
Regards,
Nenad Radosavljevic
Dear All,
I'm having problem with spandsp/txfax,
I'm not able to send a multi paged tiff file,
the fax machine receives the first page of the document and
complains about communication problem.
The file what I'm trying to send has 2
Just 2 questions:
1. Is there a plan for supporting mISDN, CAPI and SCCP exts. and trunks ?
2. Is it compatible with asterisk STABLE 1.0.X ?
regards,
Nenad
Message: 13
Date: Fri, 19 Aug 2005 12:40:22 +0200
From: Thorben Jensen [EMAIL PROTECTED]
Subject: [Asterisk-Users] IPManager
Hi !
Did anyone had issues/managed to solve issues with DISA over Zap channels on
* 1.0.X (STABLE) ?
I have a situatuion where DTMFs that should be recognized in DISA work over
SIP channels and do not work over ZAP channels (Zap channels are on TE110P)
I have in default context:
exten=
Hi
I have a connection with Panasonic TDA200.
1. For echo,,, echocancel=yes, must be before the channel definition of
the
PRI.
I'm very glad if that worked for you, but in case of D500 (connected as
described in previus messages in this thread), it seems that there is no
solution for
much.
Regards,
Nenad Radosavljevic
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo
We have Panasonic D500 and Asterisk with TE110P hybrid setup successfully
and it is possible to route DID to a PRI card in D500 if it is set up as EXT
card, and extension number of PRI card is defined for example as 2XX. This
gives you opportunity to have extensions 200 - 299 routed to PRI
Yes its true, but that version is for * HEAD, and the second one posted here
(one that have #include features.h istead of #includeparking.h in it)
works great for me, on * 1.0.7/1.0.8
Nenad
Actually, anthm is now hosting this at http://www.pbxfeeware.org.
On 6/20/05, Paul Zimm [EMAIL
Hi !
Managed to fix app_changrab.c to compile and start working under 1.0.X.
it is working on my installation, but is tested well enough.
Regards,
Nenad
Here is diff -u :
--
--- app_changrab.c.orig 2005-06-20 22:10:50.0 +0200
+++ app_changrab.c 2005-06-22
Unfortunatly it won't compile under 1.0.7 :(
I have uncommented #define AST_10_COMPAT but I don't see any usage of it in
app_changrab.c.
Complains about missing asterisk.h ( I think it should be #include
../asterisk.h )
It also complains about ASTERISK_FILE_VERSION() function, and about
compile with 1.0.X,
and SuperValletParking (www.asterlink.com/svp/) seems to be for * HEAD
(1.1.X), so it wont do me any good.
Kind regards,
Nenad Radosavljevic
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http
Nope ! This is the one that tries to include PRE 1.0.X header file
parking.h.
It cannot compile on * 1.0.X (I have tried also to include features.h
instead of parking.h (as far as I know features.h is successor to
parking.h), but still without results).
Thanks anyway.
Nenad
Try this
This one has compiled cleanly ! Thank you very much.
Nenad
Oops, I sent the wrong one. Here's one I modified to work with 1.0.X
Try again
Nope ! This is the one that tries to include PRE 1.0.X header file
parking.h.
It cannot compile on * 1.0.X (I have tried also to include
features.h
In this case you just put it into a apps subdir of astereisk tree, and add
it to a APPS= in a makefile, do a make ; make install ... and you will have
it in installed asterisk modules since by default, all modules in
asterisk modules directory are loaded, asterisk will know that it have
OK I see the ponit (although I never said that second page is interrupted -
I said that in some combinations of resolutions and TIFF options receiving
fax spits another blank sheet of paper beside the clearly received first
page).
I have read someware (some faxing tutuorial) that there is some
Lee,
thanks for the explanations.
Since I finaly figured out where to find spandsp debug information (it is
printed only on the console where the asterisk is run, not in Asterisk
remote console nor in Asterisk log files), I have gathered some debug logs
and emailed them to Steve, asking him for
Hi everyone !
I have some aditional info on problem with TXFax and sending multi-page
TIFFs.
I have made 6 different multi-page TIFFs (Group3 1D with fillbits EOL
aligned - 3 pages one TIFF in lowres and one in hires, Group3 2D -3 pages
againg in both resolutions , and Group 4 - 3pages in both
like
that txfax has
some issue with Panasonic during the hanshaking phase in between pages ???
Regards,
Nenad
Nenad Radosavljevic wrote:
Hi everyone !
I have some aditional info on problem with TXFax and sending
multi-page TIFFs.
I have made 6 different multi-page TIFFs (Group3 1D
: single image plane
Page Number: 2-3
Software: EFax Version 2.2d74
Tag 32860: Tiff Ver 2.0
Tag 32861: 555-
Group 3 Options: EOL padding (4 = 0x4)
Fax Data: clean (0 = 0x0)
Bad Fax Lines: 0
Consecutive Bad Fax Lines: 0
Regards,
Nenad Radosavljevic
On Sun, 27 Mar 2005, Nenad Radosavljevic wrote:
Only way I have managed to get Zap channel to reject a call on TE110P
without answering it, is to dial number that is not handled in dialplan
(I
have a ISDN PRI with 100 number DID service, and about 30 of them are
handled by dialplan). So far I
Hi,
Im testing asterisk for callback functionality and want to reject a call
after a few seconds for freeing the line for callback. But if I use a
congestion, there is a connection for a (billing) short time. Is there
an ability to reject a call (like the red button on a mobile phone)
without
on the lists this type of echo should not occur at all, but it simply does !
Regards,
Nenad Radosavljevic
Hi,
I'm having the same problems in echo cancellations that are mentioning
in this mail of the list
http://lists.digium.com/pipermail/asterisk-users/2003-July/016073.html ,
but I
Hi, all !
I have a situation like this:
[SIP Terminals] - [*] -ISDN-PRI- [Panasonic D500] - Telecom (conn to
Telecom is with second PRI card in Panasonic and 16 POTS lines).
Panasonic has 2 ISDN PRI cards (one to Telco, and second to Asterisk), 16
POTS lines to telco and 32 (advanced hybrid
Try to check if you have /dev/capi20 ?
If not, you can create it with:
mknod /dev/capi20 c 68 0
chown root.dialout /dev/capi20
chmod 660 /dev/capi20
That worked for me on one instalation (Debian Sarge) that somehow finished
without making /dev/capi20.
Regards,
Nenad Radosavljevic
--
Kind regards,
Nenad Radosavljevic
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http
I'm wondering why are you using SCCP and not SIP as most of us that use
Cisco 7960 phones?
Martin
Mostly because 7914 addon module is not supported in SIP images for
7960. Alternative, SIP solution, for a device like 7960+7914 could be Snom
220 + Keypad 220, but I still didn't managed to
34 matches
Mail list logo
