Hi,
Over the weekend tried to setup a test using the new app_calendar code but
receiving the following error:
[Jan 17 09:23:35] WARNING[27663]: res_calendar_icalendar.c:146 fetch_icalendar:
Unable to retrieve iCalendar 'testcal' from
Hello!
I have compiled Asterisk 1.8.1.1 on my SheevaPlug. It works all right for
me, except for one problem that I have encountered: I can only register a
SIP client (X-Lite in my case) if the secret field of the extension is left
blank. Otherwise it throws a 'bad auth' error.
Does anybody have
Hi,
I am facing an audio-problem with the dial application and I (!) think,
that it is connected to the dahdi parameter overlapdial=yes. Sangoma
support does not see any connection between this. But when enabling this
option I face with some(!) dial-partners a audio one-way issue (the
called
Try to disable certificate verification on the app. I had never tried
it personally but check for that option.
Sent from my iPhone
On Jan 17, 2011, at 5:51 PM, --[ UxBoD ]-- ux...@splatnix.net wrote:
Hi,
Over the weekend tried to setup a test using the new app_calendar code but
receiving
- Original Message -
Try to disable certificate verification on the app. I had never tried
it personally but check for that option.
Sent from my iPhone
On Jan 17, 2011, at 5:51 PM, --[ UxBoD ]-- ux...@splatnix.net
wrote:
Hi,
Over the weekend tried to setup a test using the
Can you post your confs.
2011/1/17 Arie Goldfeld arik.goldf...@gmail.com
Hello!
I have compiled Asterisk 1.8.1.1 on my SheevaPlug. It works all right for
me, except for one problem that I have encountered: I can only register a
SIP client (X-Lite in my case) if the secret field of the
Hi,
Anybody seen this before?
(using a pre-compiled asterisk from the OBS on a sles11sp1)
(I mean, i did the same with a 1.6 without any problem, but i need 1.8)
after starting:
kc3004:~ # /usr/sbin/safe_asterisk: line 145: 16133 Segmentation fault
(core dumped) nice -n $PRIORITY
On 17 Jan 2011, at 11:29, Hans Witvliet wrote:
Missing something obviously,
core dump / backtrace? ;)
Might be worth knocking a few of the modules out that were listing errors to
see if any of them are causing it. It's possible something not loading isn't
being handled gracefully.
S
--
I am posting the contents of three files: sip.conf, sip_additional.conf, and
extensions_additional.conf. Tell if anything else is important.
Thanks a lot in advance.
Arik
On Mon, Jan 17, 2011 at 1:28 PM, Andre Magalhaes alcmagalh...@gmail.comwrote:
Can you post your confs.
2011/1/17 Arie
Vladimir,
Sorry, I was a little “unclear”, Asterisk SVN-trunk-r280589M was not a 1.8
release. It was compiled before 1.8 was released.
May did a lot of patches that he comitted to the trunk version and then when I
got everything to work, I didnt recompile until I tried 1.8.2.
But I will dig into
Thanks. I fixed that.
Still does not work.
On Mon, Jan 17, 2011 at 12:53 AM, Jeroen Eeuwes jeroeneeu...@gmail.com wrote:
Hi Thomas,
register = 999:999...@sip.callwithus.comi
Perhaps this should be .com instead of .comi ?
Best regards,
Jeroen Eeuwes
--
*Hello* all,
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/aheevaccs/recordings/nosales
i have created i php code in order to read this files from server
when i put the file in www folder i can
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of salaheddine
elharit
Sent: Monday, January 17, 2011 10:57 AM
To: asterisk-users@lists.digium.com
Subject: [asterisk-users] how to read mp3
Hello all,
i have asterisk installed in our
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/aheevaccs/recordings/nosales
i have created i php code in order to read this files from server
when i put
On Sunday 16 January 2011 21:18:54 William Kenworthy wrote:
Peoples email clients, work habits and environment mean that people to
work the way thats comfortable to them. You want your mails read, you
work with them, not get on a soap box and say YOU MUST BOTTOM POST.
That was exactly my
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday, January 17, 2011 11:53 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Top Posting
On
On Mon, Jan 17, 2011 at 1:12 PM, Danny Nicholas da...@debsinc.com wrote:
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tilghman
Lesher
Sent: Monday, January 17, 2011 11:53 AM
To: Asterisk Users Mailing
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).
Anyone have ideas on cause? These calls are on an
I've searched through the wiki but I can't find what I need...I'm trying to
figure out what the max call duation is. I found references to show
application AbsoluteTimeout but that isn't in 1.6 (not even prepending core
to the front). A core help show didn't help...
--
On Mon, Jan 17, 2011 at 8:16 PM, Michelle Dupuis mdup...@ocg.ca wrote:
I've searched through the wiki but I can't find what I need...I'm trying to
figure out what the max call duation is. I found references to show
application AbsoluteTimeout but that isn't in 1.6 (not even prepending
core to
With SIP 3.2.X firmware (available on the Polycom download site) and
Asterisk 1.6.1, Polycom phones now support a full featured BLF showing
statuses of Ringing, Inuse and Online and one touch directed call pickup.
On the asterisk side all that needs to be done is to add a hint to the
extension. On
On 17/01/11 4:29 PM, jon pounder wrote:
Surely there is some mail client smart enough to be able to flip around
the levels of indenting so most recent is top or bottom.
If not quit bitching and make one - I will continue top posting since I
don't seem to be alone in preferring it.
I'm
On 01/17/2011 08:26 PM, Matt Riddell wrote:
On 17/01/11 4:29 PM, jon pounder wrote:
Surely there is some mail client smart enough to be able to flip around
the levels of indenting so most recent is top or bottom.
If not quit bitching and make one - I will continue top posting since I
don't seem
hi all,
a customer does want to have mobile integration within his asterisk
based pbx - i have already an idea how to provide it - but wanted to
ask here if someone already has an better approach - or other ideas...
What the customer basically wants is to see the status of employees
mobile
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. I can see that the call
comes in and tries to
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