Displaying 6 results from an estimated 6 matches for "rakar".
Did you mean:
radar
2005 Sep 22
12
custom ring tone
Few weeks back local telco introduced option of custom ring tones. I am
not talking about your phone ring tones but about ring tones you hear in
your headset while phone is ringing on the other end.
If I understand correctly, ringing tone is generated localy on asterisk
if you are connected to phone network with E1/T1 connection. Which means
that instead of regular beep-beep tone we could send
2004 Apr 24
0
Messengers calls dropped (SIP problem?)
...nger and then call is dropped always in
25th second of the call. Any ideas what I did wrong?
here is my messenger sip.conf portion;
[marko]
type=friend
reinvite=no
username=marko
host=dynamic
mailbox=1300
here is my cisco 7905 sip.conf portion;
[123]
type=peer
reinvite=no
callerid= "Marko Rakar"
username=123
secret=1234
dtmfmode=inband
careinvite=yes
host=dynamic
defaultip=192.168.3.52
incominglimit=2
outgoinglimit=2
here is a part of my sip debug file
9 headers, 0 lines
Sending to 192.168.3.54 : 14250 (non-NAT)
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.54...
2004 Apr 03
2
Ztdummy - is it requirement?
I am interested to learn if I need to have ztdummy installed if I do not
have any zaptel hardware in my machine?
I have found a lot of references with RTP problems which were related to
RTP timing (or lack of it).
My problem is that voice coming from SIP hardware is OK, but voice going
from asterisk to SIP hardware is choppy, full of noise or completely
cut-off. Am I going to solve my problem
2004 Apr 02
0
SIP call troubleshooting
Can someone help me what went wrong with this call?
This call was initiated from dev/ttyI0 device on my asterisk server to
mediatrix unit. Mediatrix unit user received the call and call started.
I can hear them OK but they can not hear me correctly (cut-off sound,
noise). Call was finally hunged up.
Can anyone point out if there was something wrong?
-*CLI> sip debug
SIP Debugging Enabled
2004 Apr 05
3
ZAP channels
I have made bri-stuff.0.0.2rc19 to work (I think) but I can not achieve
any in-dialing nor I can dial out;
this is what I have from "pri intense debug span 1" command
----------
*CLI> pri intense debug span 1
Enabled EXTENSIVE debugging on span 1
-- Executing Playback("SIP/201-a862", "tt-weasels") in new stack
-- Playing 'tt-weasels' (language
2004 Aug 19
9
bridging and internet
(I''m not a member of the list at the moment so please answer this e-mail CC to
my personal address. Thank you all)
I am part of a community network in Buenos Aires and I''m now trying to
set up a bridge between my local net and the community net.
The problem is that appart from the bridge between these I need to share
an internet connection and the cable modem assigns me a