Hi list,
I had a problem with the dahdi XPP driver.
After this error in syslog, the Xorcom disconnect from the server:
Jun 3 15:03:29 kernel: [361010.637858] *NOTICE-xpp_usb: xusb-0
(usb-:00:1d.7-3) [X1047686]: Sluggish USB. Dropping next PCM frame (p**
**ending_writes=5)*
Jun 3 15:03:52
On 6/6/13 4:53 am, Gopalakrishnan N wrote:
Any other HA applications available or the lsyncd with pacemaker is good?
I generally use Pacemaker with Heartbeat, which seems to work pretty well.
Kind regards,
Chris
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This email is made from 100% recycled electrons
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On Thu, Jun 06, 2013 at 09:31:39AM +0200, Matteo wrote:
Hi list,
I had a problem with the dahdi XPP driver.
After this error in syslog, the Xorcom disconnect from the server:
Has it happened once? More then once? Reproducable?
How long has the Astribank been working till then?
On Thu, Jun 6, 2013 at 9:56 AM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Thu, Jun 06, 2013 at 09:31:39AM +0200, Matteo wrote:
Hi list,
I had a problem with the dahdi XPP driver.
After this error in syslog, the Xorcom disconnect from the server:
Has it happened once? More then
Hi Justin
I have done some configuration and basic IVR is working . But I 'm still
finding , how to redirect IVR call to a certain VOICE XML server.
For instance http://192.168.2.10:8080/test has an IVR service.
could you please help me with an guide to do the configurations ?
Many Thanks
Kamailio has both a ha1 and ha1b column in it's user schema:
ha1 = H(A1) = MD5(user:realm:password)
ha1b = H(A1b) = MD5(user@realm:realm:password)
This is intended to support some devices that append @realm to the user
and/or to allow users to put either user-part only or user@domain
into the
Hello,
when picking up an incoming call from one ip phone on another ip phone,
the call terminates after about 5 to 10 seconds.
When reading out the hangup cause variable in the h-extention of the
dialplan, the hangup cause seems to be 111.
In the dialplan output, you can see that
Is the template capability in sip.conf compatible with realtime sip.conf
entries such as users in a database?
I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
don't mention a template column:
https://wiki.asterisk.org/wiki/display/AST/SIP+Realtime%2C+MySQL+table+structure
hello list ,
i want to use meetme with asterisk1.4 i check in this forum and i found
this code :
exten = 508,1,MeetMe(1000,ipdM)
when i use this code in my server i can say my name and i press 1 in order
to enter in the conference ; but i want to asks the customer to press an
number and
I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over
Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two
guides. The first is the one from Asterisk: The Definitive Guide 4th
edition
(http://ofps.oreilly.com/titles/9781449332426/asterisk-Install.html) and
the
what is host architecture ?
try to install ubuntu x86 not x86_64.
On Thu, Jun 6, 2013 at 5:12 PM, jorgeart...@protoboardmx.com wrote:
I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running over
Oracle VM VirtualBox (v 4.1.8). So far I have tried it following two
guides. The
On 06/06/13 15:51, Daniel Pocock wrote:
Is the template capability in sip.conf compatible with realtime sip.conf
entries such as users in a database?
I notice that contrib/realtime/mysql/sippeers.sql and the wiki page
don't mention a template column:
I'm unfamiliar with Voice XML. Just looking around a bit, it looks like Voxy
does exactly what you want - I'd look into that:
http://sourceforge.net/projects/voxy/
From: luke devon [mailto:luke_de...@yahoo.com]
Sent: Thursday, June 06, 2013 1:37 AM
To: Justin
Hi,
I need to rebuild a system which has 4 BRI ports and is connected to
Point-to-multiPoint lines, in a country where telco often drop lines for
energy savings.
I'm planning to use latest 11.4.0 asterisk version along with dahdi and
libpri (no misdn).
Which version are recommended for Dahdi
Hello,
enter in make menuselect - Compiler flags and disable
BUILD_NATIVE option; then recompile Asterisk
Regards
El 06/06/2013 10:12, jorgeart...@protoboardmx.com escribió:
I'm trying to install and run Asterisk 11 on Ubuntu 12.04.2 running
over Oracle VM VirtualBox (v 4.1.8). So far I
On 06/06/2013 05:55 PM, Olivier wrote:
Hi,
I need to rebuild a system which has 4 BRI ports and is connected to
Point-to-multiPoint lines, in a country where telco often drop lines
for energy savings.
I think the dropped D-channel issue should be handled by a very recent
DAHDI. If there are
Hi,
I'm used to add OSLEC source code into asterisk and use it as default echo
canceller.
Currently, I'm proceeding this way:
download the kernel source code roughly matching the kernel version I'm
targeting with command such as wget
http://kernel.org/pub/linux/kernel/v3.0/linux-3.2.46.tar.bz2;
On 06/06/2013 05:55 PM, Olivier wrote:
Hi,
I need to rebuild a system which has 4 BRI ports and is connected to
Point-to-multiPoint lines, in a country where telco often drop lines
for energy savings.
I think the dropped D-channel issue should be handled by a very recent
DAHDI. If there are
Awesome Bakko, I did what you say and now I have Asterisk 11 up and
running.
Thanks a lot!
On Thu, 06 Jun 2013 11:09:48 -0500, Bakko wrote: Hello,
enter in make menuselect - Compiler flags and disable BUILD_NATIVE
option; then recompile Asterisk
Regards
El 06/06/2013 10:12,
2013/6/6 jg webaccou...@jgoettgens.de
On 06/06/2013 05:55 PM, Olivier wrote:
Hi,
I need to rebuild a system which has 4 BRI ports and is connected to
Point-to-multiPoint lines, in a country where telco often drop lines
for energy savings.
I think the dropped D-channel issue should be
Googling, I found this one-year old thread about the same subject
http://thr3ads.net/asterisk-users/2012/06/1950384-Asterisk-10-1.6.1-and-Dahdi-Libpri-compatilities-in-BRI-PtmP
If my reading and understanding is correct:
1. Briging layer 1 down is specific to PtmP (PtP is not concerned).
2. When
Hello All,
I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't get
meetme feature to work when dial meetme extension, can you please help?
It always worked before, also I do not have dahdi installed on this
machine, never did.
-- Executing [104@sipphones:1]
I don't see where your problem is. Can you get a pcap trace of the
D-channel? There's not that much, so you could run it overnight.
You should see the RR frames, maybe a few of SABMEs. I am wondering
whether the RR frames will stop at some point. When things need to
reinitialize you should
2013-06-06 22:21, motty cruz skrev:
Hello All,
I upgraded Asterisk 1.8.10 to Asterisk 1.8.22 since upgrading I can't
get meetme feature to work when dial meetme extension, can you please help?
It always worked before, also I do not have dahdi installed on this
machine, never did.
Thanks Johan,
I did noticed /etc/dahdi so you're right it was installed on one point, I
re-install dahdi and problem went away.
Thank you very much!
On Thu, Jun 6, 2013 at 1:45 PM, Johan Wilfer li...@jttech.se wrote:
2013-06-06 22:21, motty cruz skrev:
Hello All,
I upgraded Asterisk
On 06.06.2013, at 15:05, Jonas Kellens jonas.kell...@telenet.be wrote:
Hello,
when picking up an incoming call from one ip phone on another ip phone, the
call terminates after about 5 to 10 seconds.
When reading out the hangup cause variable in the h-extention of the
dialplan, the
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