Hello,
No, another installation haven't solved the problem!
It looks more like something related to the configuration in setting the
running environment!
Thanks. --
_
-- Bandwidth and
Hello,
I am trying to make my first call on Asterisk to succeed. I have Asterisk
1.8.10.1 running on Ubuntu machine.The configuration is quite simple just for
my first test, Trying to have a call between two X-lite sipphone. The
subscribers succeeded to register and the call is established, but
Choose suitable NAT settings from sip.conf
turn direct media in sip.conf or per peer off
On Thu, Sep 19, 2013 at 12:54 PM, Asmaa Ahmed asabatg...@hotmail.comwrote:
Hello,
I am trying to make my first call on Asterisk to succeed. I have Asterisk
1.8.10.1 running on Ubuntu machine.
The
On Thu, 19 Sep 2013, David Duffett wrote:
i am getting these errors in asterisk cli
-- Executing [01179553708@default:1] Set(SIP/-015b,
CALLERID(num)=xx) in new stack
-- Executing [01179553708@default:2] Dial(SIP/-015b,
SIP/01179553708@sipgate,30,trg) in new stack
remove content of /var/log/asterisk/messages /var/log/asterisk/messages
run asterisk and post content of /var/log/asterisk/messages to pastebin.
On Thu, Sep 19, 2013 at 9:39 AM, Asmaa Ahmed asabatg...@hotmail.com wrote:
Hello,
No, another installation haven't solved the problem!
It looks
It looks like the challenge response after INVITE is not been accepted.
Provide more detail.
$ sip set debug peer sipgate
--
==
Miguel Oyarzo
DevOps Engineer
http://www.linkedin.com/in/mikeaustralia
Linux User: # 483188 - counter.li.org
Melbourne, Australia
It looks like the challenge response after INVITE is not been accepted.
Provide more detail.
$ sip set debug peer sipgate
server*CLI sip set debug peer sipgate
SIP Debugging Enabled for IP: 217.10.79.23:5060
Really destroying SIP dialog '3ef8ff1a6ec360626af409b112b860ee@127.0.1.1'
Method:
Challenge authentication look good.
--- SIP read from UDP:217.10.79.23:5060 ---
SIP/2.0 200 OK
Are you sure this number format 01179553708 is accepted in that SIP trunk?
Some VOIP providers only accept international format.
--
==
Miguel Oyarzo
DevOps Engineer
On Thu, 19 Sep 2013, Miguel Oyarzo wrote:
Challenge authentication look good.
--- SIP read from UDP:217.10.79.23:5060 ---
SIP/2.0 200 OK
Are you sure this number format 01179553708 is accepted in that SIP trunk?
Some VOIP providers only accept international format.
when i use a softphone
you have insecure=port,invite in sipgate peer?
On Thu, Sep 19, 2013 at 12:26 PM, gpxctawjc...@irational.org wrote:
On Thu, 19 Sep 2013, Miguel Oyarzo wrote:
Challenge authentication look good.
--- SIP read from UDP:217.10.79.23:5060 ---
SIP/2.0 200 OK
Are you sure this number format
What you don't have mentioned yet is whether your outbound call reaches
the destination.
--
==
Miguel Oyarzo
DevOps Engineer
http://www.linkedin.com/in/mikeaustralia
Linux User: # 483188 - counter.li.org
Melbourne, Australia
On 9/19/2013 8:26 PM,
exten = 123,1,Set(TIMEOUT(absolute)=3600)
exten = 123,n,MeetMe(blah,d)
if you are using freepbx and you want to set timeout for all conference rooms
go here -http://dn.forceit.ru/asterisk-conference-timeout
--
_
--
Le 19/09/2013 05:01, David Duffett a écrit :
I believe registration is in place, otherwise inbound calls would not
work.
Yes, I didn't read carefully the original message, sorry.
[...]
--
Daniel
--
_
-- Bandwidth and
I'm trying to setup a pair of FreePBX-4.211.64 boxes using Pacemaker,
Corosync, and DRBD.
All the examples I've found so far use Heartbeat, but Heartbeat is not in
the repositories and doesn't want to compile from source.
Does anyone have a working configuration they can share or a tutorial
Hi,
Asterisk 11 doc says CDR(src) value is read-only (see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR).
For various reasons, I would appreciate to change its value so that it my
own presentation rules instead of telco rules.
Very often, I'm connected to telcos through
On Thu, Sep 19, 2013 at 9:02 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
Asterisk 11 doc says CDR(src) value is read-only (see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR).
For various reasons, I would appreciate to change its value so that it my
own presentation rules
Be careful with DRDB singe failing drive/corruption on one peers takes down the
other too...
Check out haast as well (at www.generationd.com) for a commercial asterisk
clustering solution.
Michelle
(GenerationD Systems)
From:
Hello Edwards
you can install fedora repositories and the HeartBeat from those
repositories.
If the failover is only for two servers, this is a good solution.
In the directory list, you have to add /etc/dahdi (is you use dahdi) and
/var/spool/asterisk
Regards
El 19/09/2013 08:58, Steve
Asmaa Ahmed wrote:
I am trying to make my first call on Asterisk to succeed. I have
Asterisk 1.8.10.1 running on Ubuntu machine.
The configuration is quite simple just for my first test, Trying to
have a call between two X-lite sipphone. The subscribers succeeded
to register and the
Is anyone aware of a way to replicate parts of the AstDB to another Asterisk
install?
For example, to export all CF entries on a FreePBX based system to another
system running FreePBX, I might do:
asterisk -rx 'database show' | grep CF
This gives me a list of data, which I can rsync to
Hi,
Am Mittwoch, den 18.09.2013, 14:29 +0100 schrieb
gpxctawjc...@irational.org:
Hello
i am trying to setup sipgate gateway
i can get incoming calls fine, but when i dial in and then try to dial
out i get this in asterisk command line
What Sipgate product are You using? At least in
2013/9/19 Matthew Jordan mjor...@digium.com
On Thu, Sep 19, 2013 at 9:02 AM, Olivier oza_4...@yahoo.fr wrote:
Hi,
Asterisk 11 doc says CDR(src) value is read-only (see
https://wiki.asterisk.org/wiki/display/AST/Asterisk+11+Function_CDR).
For various reasons, I would appreciate to change
I saw this thread which is very similar to my issue, though I cannot
solve mine with iptables.
http://lists.digium.com/pipermail/asterisk-users/2013-September/280429.html
Using asterisk 11.5, IAX does not seem to be able to receive any
packets.
My IP tables looks like this:
Hi All,
Could anyone tell me the real use of internal_ timing=yes option on
asterisk.conf file? I am using asterisk 1.4.22.
As per my understanding if we don't have any TDM card installed with
appropriate driver, we use internal_timing = yes to get the timing from ztdummy
/ztDahdi.
Is there
And here I thought I was back in the dark ages using 1.4.44!!
You had better consider moving up to a more current version before you get bit
real hard!
John Novack
Comp Aholic wrote:
Hi All,
Could anyone tell me the real use of internal_ timing=yes option on
asterisk.conf file? I am using
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