What read/write rights do I need to issue this sip prune realtime peer
command in manager.conf ??
Jonas.
Jonathan Thurman wrote:
If you have a web interface for updating information you could always use AMI
to issue the prune/reload after committing the changes.
-Jonathan
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You can call sip:200...@login.zipdx.com in g722 wideband
Hello.
There are so many sound files in /var/lib/asterisk/en. Is there an easy
way to let me play back all of them one by one while I am watching CLI
to see the current file name?
Thanks for help.
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Jian Gao
IT Technician
SJ Geophysics Ltd. http://www.sjgeophysics.com
What I did what pick up the wav version to my mac and then I can play any of
them I want in quicktime or any other audio player. Easier for me than cooking
up some asterisk way.
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CfMC
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On Apr 23, 2010, at 10:11 AM, Jian Gao
On Fri, 23 Apr 2010, Jian Gao wrote:
There are so many sound files in /var/lib/asterisk/en. Is there an easy
way to let me play back all of them one by one while I am watching CLI
to see the current file name?
No.
How about:
for F in /var/lib/asterisk/en/*.wav
This is tested in 1.4.30
#!/usr/bin/perl
#
# allfiles 1.0
#
# reads all of the files in /var/lib/asterisk/sounds/en and plays on the
line
# while showing name on console (CLI)
# save as at: /var/lib/asterisk/agi-bin/allfiles.agi Be sure to chmod +x
it!
#
# Invocation Example:
# exten =
Thanks for all the help.
For now I am going to use Jim's method - get the wav files to my PC.
Since I don't have wav sound files installed in my * box (we use g729),
so I went to Digium site downloaded the gz sound files. After unzip them
I found there is a core-sounds-en.txt and
I opened a ticket about this:
https://issues.asterisk.org/bug_view_advanced_page.php?bug_id=17217
Remove -c on the init script of asterisk, line 85. Should help.
I was trying it with a xen guest.
Awesome, Thanks.
[Kelvin Chan]
NOTICE: This communication is intended only for the use of
Hi List,
i have to put an * between two other SIP gateways and due to some
circumstances, i have to use sip over tcp. With 1.6.2.6 this is working
fine: sip gw A (deverto4) sends the call, i hand it over to sip gw B
(ocs) and that's about it. In the other direction however (ocs - me -
I don't think RTP can be sent over TCP at all, it would defeat the whole
purpose of RTP. Even if you somehow manage to do so, voice quality will go
down the drain.
Zeeshan A Zakaria
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On 2010-04-23 3:27 PM, ad...@3a.hu wrote:
Hi List,
i have to put
SIP is just the control protocol, and can be negotiated over TCP or UDP. The
actual payload is done over RTP, which is a UDP-based protocol.
If you had to add firewall exceptions/PAT config for the TCP SIP traffic,
you'll also need to add the same for RTP traffic as well.
-- Nathan Clemons
On
Hello.
As I see, there is a lot of threads about jitter buffer... Maybe anybody
knows something about my case? Any help will be appreciate.
Thanks in advance.
-- Original message --
From: russian qwerty russian.qwe...@gmail.com
Date: 2010/3/31
Subject: Jitter Buffer and MeetMe.
Hi Guys,
On 04-23-2010 21:40, Nathan Clemons wrote:
SIP is just the control protocol, and can be negotiated over TCP or UDP. The
actual payload is done over RTP, which is a UDP-based protocol.
thanks, for both of you for pointing this out. i was obviously on the
wrong track here. since i
Not sure if this is the right place to ask, but what do we need to do to
get this patch merged? How can I help? I'm no dev, but I use LDAP with
Asterisk and I might be of some help.
Thanks guys.
--
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-- Bandwidth and
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El jueves 22 de abril del 2010 a las 14:33:01 -0300,
Philipp von Klitzing escribió:
Hi!
Hi, Philipp.
But it draws attention to me between the PC with softphone and the
telephone I see traffic ARP or ICMP that could make to try between
the
Hi all.
I am having lots of trouble with random calls dropping after 20
seconds, and I finally managed to capture a full sip trace. I'll paste
it in full below, but I'll give a summary first. It seems that
Asterisk is not recognizing the ACK messages that it receives from the
Grandstream ATA.
On Fri, Apr 23, 2010 at 9:14 PM, Daniel Bareiro daniel-lis...@gmx.netwrote:
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El jueves 22 de abril del 2010 a las 14:33:01 -0300,
Philipp von Klitzing escribió:
Hi!
Hi, Philipp.
But it draws attention to me between the PC with softphone
Hi all.
I am having lots of trouble with random calls dropping after 20
seconds, and I finally managed to capture a full sip trace. I'll paste
it in full below, but I'll give a summary first. It seems that
Asterisk is not recognizing the ACK messages that it receives from the
Grandstream ATA.
I thought 1.6x could have more than 1 parking lot. I have searched
high and low and can't find any docs on it, samples or examples.
Was I wrong? I need a 2nd parking lot for the sales department and
thought I could have a default parking lot at 70 with slots from 71
to 74 and a 2nd one at 75
Hello asterisk gurus,
I'm developping a script that create call files
dynamicly from a database. Here the scenario
script move call file to outgoing dir to place the call
call is connected to [extension] which contains a playback app.While line
is ringing, playback
call-id doesn't match?
SIP/2.0 200 OK
...
Call-ID: 2117388659-506...@82.158.83.xxx
...
ACK sip:6615xx...@130.117.xxx.xxx SIP/2.0
...
Call-ID: 2117388659-506...@192.168.1.100
...
I'm not sure, but I think that the part after the '@' must also match
throughout the dialog. A Grandstream bug?
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