On 06.06.2013, at 15:05, Jonas Kellens 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 hangup cause seems to
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 wrote:
> 2013-06-06 22:21, motty cruz skrev:
> > Hello All,
> >
> > I upgraded Asterisk 1.8.10 to A
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.
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 see
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] MeetMe("SIP/101-0
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
2013/6/6 jg
> 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 h
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, jorg
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
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 f
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 a
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 K
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:
>
> https://wiki.asterisk.org/wiki/dis
what is host architecture ?
try to install ubuntu x86 not x86_64.
On Thu, Jun 6, 2013 at 5:12 PM, 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 first is the one from "Asteris
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
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 passwor
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
whi
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 SIP-peer
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 t
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
L
On Thu, Jun 6, 2013 at 9:56 AM, Tzafrir Cohen wrote:
> 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?
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?
> (usb-:
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
--
This email is made from 100% recycled electrons
--
__
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
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