Hi
I'm using 1.8.7.0. This morning I got an alert telling me
Asterisk on my-host exited on signal 11. Might want to take a peek.
When I had a look at the logs I can see a lot of errors like
ERROR[14924] astobj2.c: refcount -1 on object 0x2aaad4069068
ERROR[14924] astobj2.c: refcount -2 on
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 14, 2012 4:05:21 AM
Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object
Hi
I'm using 1.8.7.0. This morning I got an alert telling me
Asterisk
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Steve Edwards
Sent: Tuesday, November 13, 2012 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] On SIP REGISTER
On Wed, 2012-11-14 at 05:27 -0800, Michael L. Young wrote:
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: asterisk-users@lists.digium.com
Sent: Wednesday, November 14, 2012 4:05:21 AM
Subject: [asterisk-users] Error: astobj2.c: refcount -1 on object
Hi
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
On Wed, 14 Nov 2012, Danny Nicholas wrote:
What you will need to do is to monitor for the SIP REGISTER in AMI, then
launch a local channel call to run your AGI process. If the process is
not
We have also used enswitch from Integrics and I totally recommend it. It
does all the things you ask for and some more.
Regards,
Ali Pey
On Tue, Nov 13, 2012 at 9:48 PM, Alistair Cunningham
acunning...@integrics.com wrote:
Hello Marshall,
Please see Enswitch:
You can also consider using a proxy server such as opensips or Kamailio.
They would enable you to do much more at the signalling level and many
other advantages such as better security or nat traversal.
Regards,
Ali Pey
On Wed, Nov 14, 2012 at 11:42 AM, Steve Edwards
- Original Message -
From: Ishfaq Malik i...@pack-net.co.uk
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, November 14, 2012 9:25:37 AM
Subject: Re: [asterisk-users] Error: astobj2.c: refcount -1 on object
Thanks for
On Wed, Nov 14, 2012 at 7:42 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Face wrote:
Is there a way I can trigger a AGI script On SIP REGISTER event.
On Wed, 14 Nov 2012, Danny Nicholas wrote:
What you will need to do is to monitor for the SIP REGISTER in AMI,
On Wed, 14 Nov 2012, Face wrote:
Well, I have a SMS dialplan and the problem is I cannot send an offline
SMS to SIP users. Therefore, I am trying to edit the SMS dialplan to
save the failed SMS to a file and deliver them when the SIP user
REGISTER.
I see 2 parts:
1) When the 'send SMS'
On Wed, 14 Nov 2012, Ali Pey wrote:
You can also consider using a proxy server such as opensips or Kamailio.
They would enable you to do much more at the signalling level and many
other advantages such as better security or nat traversal.
Valid points, but as we get clarity on what the OP
Has anyone been able to configure Asterisk to work over 3G?
I bought Nokia Cell Phones just for that purpose and they register fine
over WiFi and 3G but the quality is just not good enough and sometimes
the call just disconnects.
I have Allow as:
ilbc
gsm
ulaw
alaw
--
Co-op Vacation Rentals
On Wed, Nov 14, 2012 at 3:15 PM, Roy Abshire r...@coopvr.com wrote:
Has anyone been able to configure Asterisk to work over 3G?
I bought Nokia Cell Phones just for that purpose and they register fine
over WiFi and 3G but the quality is just not good enough and sometimes the
call just
On Wed, Nov 14, 2012 at 3:30 PM, Roy Abshire r...@coopvr.com wrote:
Ok, what about 4G, I tried it with my 4G ATT Hotspot too. I get 4 full
signal and the call goes through but keeps cutting out or dropping.
I'm using the ATT Elevate with a Nokia N97. SIP connects fine, I can
make and
G729 requires a paid license right? If I wanted to test it using
G729...do I still have to buy a license for it?
My goal is to use my cell phone for Text Only and pay $15/month and use
my ATT WiFi Hotspot for VOIP calls wherever I go because I rely on my
VOIP for business.
Co-op Vacation
On Wed, Nov 14, 2012 at 3:34 PM, Roy Abshire r...@coopvr.com wrote:
G729 requires a paid license right? If I wanted to test it using
G729...do I still have to buy a license for it?
My goal is to use my cell phone for Text Only and pay $15/month and use my
ATT WiFi Hotspot for VOIP calls
On Wed, Nov 14, 2012 at 4:20 PM, Roy Abshire r...@coopvr.com wrote:
My goal is to always be connected to my VOIP system using 3G or 4G. I'm
still going to have to pay the ATT Data only $50/month plan for 4G and I
get 5gb a month which is plenty.
Why would you always want to be connected to
Believe me, there is a method to my madness that I didn't want to get
into but here it goes.
I want to get this to work reliably even with low quality and bandwidth
so I can install VOIP Phones at my vacation rental properties that can
only get 4G or low speed Satellite with only 128k upload.
Did you see this URL?
http://downloads.asterisk.org/pub/telephony/asterisk/
On Wed, Nov 14, 2012 at 4:24 AM, Justin Killen
jkil...@allamericanasphalt.com wrote:
In http://packages.digium.com/centos/ there is not yet a centos 6 branch
(Nor is there a RHEL 6 branch). Centos 6.0 was
On Wed, Nov 14, 2012 at 5:22 PM, Roy Abshire r...@coopvr.com wrote:
Believe me, there is a method to my madness that I didn't want to get
into but here it goes.
Quite often, telling people the core problem can lead to better solutions.
I assumed you wanted mobile, but fixed cellular could
I know this isnt an exact solution the way you want. But if cutting down on
cell phone costs is the goal what I've done in the past is if you just
setup 1 DID with DISA you can put that DID on your cell phones friends and
family and then you have unlimited minutes.
I wouldnt plan on too much
On Wed, Nov 14, 2012 at 10:14 PM, Steve Edwards
asterisk@sedwards.com wrote:
On Wed, 14 Nov 2012, Ali Pey wrote:
You can also consider using a proxy server such as opensips or Kamailio.
They would enable you to do much more at the signalling level and many other
advantages such as better
Dear,
i using this scenario.
jitsi--- asteriskEPABX-- Local Telephone
when i am calling from jitsi to no 88 its giving this message and getting
busy tone.
== Using SIP RTP CoS mark 5
-- Executing [88@myphones:1] Dial(SIP/sandeep-0004,
DAHDI/g0/88,20,rt) in new stack
--
Hello,
As I don't have any Polcyom SoundPoint at hand, at the moment, I would like
the following to this list readers.
I have a steup where two DHCP servers are installed : one dedicated for IP
telephony, the other for IT..
When a Polcyom SoundPoint gets a VLAN ID from a DHCP server, does it
Hi all,
Is there a simple way of disabling regular expressions in the dialplan?
Reason for asking, is that people hate to remember numbers.
So i want to use there full smtp address as as their extension.
In stead of 12345678 i would like to use b.c.o.gr...@minoss.nl
But afaicr the dots will
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