--[ UxBoD ]-- wrote:
- Steven J. Douglas stev...@moij.biz wrote:
--[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
Okay, getting somewhere now ! I am now getting the following
--[ UxBoD ]-- wrote:
- Gordon Henderson gordon+aster...@drogon.net wrote:
On Fri, 1 May 2009, --[ UxBoD ]-- wrote:
Okay, getting somewhere now ! I am now getting the following :-
== Starting post polarity CID detection on channel 1
-- Starting simple switch on 'DAHDI/1-1'
Hi Jonas,
Maybe you can try leaving out bindport and bindaddr parameters first.
The port defaults to 4569 anyway. As for the bindaddr, you should be
using the IP Address of your interfaces. I am assuming you are using the
IP Address obtained from your router. If that is the case, then asterisk
Thanks to all who replied. The problem was due to a faulty NTU box from
the telco. It has been up for almost a week now without any downtime.
Regards,
Steve
Steven J. Douglas wrote:
Thanks for the tip, Harry. I will try that when I have exhausted all
avenue. My problem is that if I upgrade
with latest
DAHDI. In the DAHDI case I even had to use latest Subversion revision
due to some bug (but that was related to the TE121-cards I think).
Since then I haven't had any issues at all, so consider updating
Asterisk and Zaptel-DAHDI
2009/3/31 Steven J. Douglas stev...@moij.biz:
Hi
Hi guys,
I've been trying to get my ISDN-10 line up for the past few days, but
its been going up and down. I am using OpenVox D110P card on
asterisk version 1.4.21. It seems to me like a cable problem. I tried
using Ethernet straight cable (12, 45, 36, 78) and also a straight
cable where
/crossover+T1+cable
On Tue, Mar 31, 2009 at 12:37 AM, Steven J. Douglas stev...@moij.biz
mailto:stev...@moij.biz wrote:
Hi guys,
I've been trying to get my ISDN-10 line up for the past few days, but
its been going up and down. I am using OpenVox D110P card on
asterisk
Maybe your network is not ready when asterisk first fires up?
-steve
Yahya Mohammad wrote:
I'm running asterisk on Ubuntu 8.10. I have two 'register' lines in
iax.conf for registering with two remote servers. However only the
first one registers at system startup. I always have to issue an
Thanks! I'll give that a try.
Regards,
Steve.
Tilghman Lesher wrote:
On Monday 02 March 2009 00:27:00 Steven J. Douglas wrote:
Hi,
My asterisk segfaults a few times each day and the crash problem seems
weird. When I run gdb on the core dump, it almost always segfaults on
free
Hi Ken,
If you run ulimit -c on the command line and get a 0 output, then
you need to run ulimit -c unlimited on the command line.
-Steve
Ken D'Ambrosio wrote:
Asterisk segfaulted on me the other day; how do I tell it to generate a
core file so -- if it happens again -- I can attempt to
Hi,
My asterisk segfaults a few times each day and the crash problem seems
weird. When I run gdb on the core dump, it almost always segfaults on
free() or malloc(). When I run the back trace, I see something weird.
Here's one of the back traces.
#0 0x4017f87f in _int_free () from
Hi,
Have you tried using externip in your sip.conf? By setting the correct
localnet, any SIP packets that goes elsewhere will use the value in
externip. This might solve your problem.
Regards,
Steve
nik600 wrote:
On Sat, Feb 7, 2009 at 8:31 AM, nik600 nik...@gmail.com wrote:
hi
is it
Hi Lincoln,
The fact that you can hear and respond to the voice mail (even if its
for the first 20 seconds), means that your phone has received the OK
message properly. The problem is the missing ACK after receiving OK.
When asterisk did not receive the ACK after a few retries of the OK, it
Use different context for both users in sip.conf. In the context for
user 100300, include the context sip-trunk-100. For user 101300, include
the context sip-trunk-101.
Regards,
Steve
Nhadie wrote:
Hi,
Is it possible to detect where the call came from and route it out to
different sip
In your Read command, leave out the .wav extension in the file name.
Regards,
Steve
Artifex Maximus wrote:
Hi all!
I would like to make a service with recording sounds and playing back
to caller. I had wrote the script but it failed at Read statement with
file not found error. I have put
Hi Dimitar,
You can use the Read command in your 5051 extension to wait for a
response after the user answers the phone.
Regards,
Steve
Dimitar Dimitrov wrote:
Hi,
I have some troubles with early media with Zapatel TDM400P adapter. I
made a simple callback function wich works by followin
Don't use g729 in the iax.conf for the IAXY device. It doesn't support it.
Regards,
Steve
Adam Robins wrote:
I am using a Polycom SIP phone (ext 2042) to call an analog phone
connected via an IAXY (ext 2120). The analog phone rings, and when I
answer, I can hear the person speaking on the
100103 to use [sip-trunk-100-3] for outbound
calls. can i route it based from the source. TIA
Regards,
Nhadie
Steven J. Douglas wrote:
Use different context for both users in sip.conf. In the context for
user 100300, include the context sip-trunk-100. For user 101300, include
Hi Steve,
Thanks for the tip. But unfortunately it doesn't help. The Avaya is
passing on the MFC codes to the SIP phone when it answers the call. I
think the solution might be in the Avaya configuration to properly
convert the signaling.
Regards,
Steve
Steve Totaro wrote:
Answer() is the
Hi,
I'm trying to integrate my asterisk PBX to an Avaya PBX. I am using
chan_ooh323 from asterisk-addons.
I am able to make a call from SIP Phone - Asterisk - Avaya - Station
(phone) and vice versa.
I am also able to make a call from SIP Phone - Asterisk - Avaya - PSTN.
However I face
give your suggestion a try and see if it makes a
difference.
Thanks.
-Steve
David fire wrote:
try a answer() before the dial(sip/xxx)
and if you are using originate try local/ and start whit and answer()
2009/1/22 Steven J. Douglas stev...@moij.biz mailto:stev...@moij.biz
Hi
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