Displaying 20 results from an estimated 9000 matches similar to: "Install just to play with experiment"
2010 Aug 04
2
How to record a file and play some other file at the same time
Hi,
I have an xlite registered with asterisk server. When i dial a number AGI is
invoked. and in this we are running to threads one to record files and one
to play files. So i dialed the extension and i started recording and playing
at the same time. On the xlite i m getting an indication when recording my
voice and at the same time i could see playing the other file too. But in
the directory
2010 Dec 22
4
Asterisk hangs up call after 20s
Hello
I have an Asterisk 1.4 server and two XLite softphones, where
Asterisk and the local XLite phone are located in a LAN behind a NAT
router, and the remote XLite phone is located elsewhere on the Net
behind its own NAT router:
http://img252.imageshack.us/img252/3749/asterisknat.png
I'm having the following issue: When the _local_ XLite calls out the
remote XLite, everything works fine;
2005 Feb 03
2
Odd behaviour between Grandstream and Xlite
Hi,
I've got an Asterisk box with grandstream and xlite clients on it.
No here's the thing:
- I grey out all the codecs on the Xlite except for GSM
- I call the Grandstream from the Xlite, the Xlite uses the GSM codec
and the Grandstream uses ulaw, with Asterisk doing the conversion,
everything fine
- I call the Xlite from the Grandstrea, the Xlite ends up using the
ulaw codec as
2006 Dec 28
1
one way rtp stream (Sent alwax to 127.0.0.1)
Asterisk version 1.2.14
I use snom190 and xliteV3 as sip phones.
asterisk send the rtp stream never to the xlite softphone.
Any hits for me?
*CLI> rtp debug
RTP Debugging Enabled
-- Executing Dial("SIP/xlite-007918f0", "SIP/snom") in new stack
-- Called snom
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 is ringing
-- SIP/snom-00797110 answered
2006 Mar 16
1
Newbie needs audio help
My first Asterisk install: Debian sarge with the 2.6 kernel, and two
X-Lite soft-phones. I followed the online how-to documents and was
calling between the two soft-phones and calling the demo system with
no problems and had full audio. I then went on to configure the
TDM400P's two FXS modules. I got into that a ways and was having some
success, but no dial-tone when I was off the
2011 Feb 24
1
Using a Virtual IP Line
Hello!
I bought a virtual IP line to my ISP to use with my asterisk but when I try to connect it to my ISP tells me I can not use and I can only use with a softphone that gives me, xlite ready configured.
I use ngrep to see what information sent on xlite for communication, the User-Agent was changed so I change the User-Agent to my asterisk to the same as saying the xlite but still does not work.
2011 Mar 17
1
[1.6.2.5] Asterisk can't find MOH file
Hello
I thought I had things set OK to have Asterisk play FR files for
prompts and MOH, but for some reason, it still can't find them:
============ ll /var/lib/asterisk/sounds/
drwxr-xr-x 2 asterisk asterisk 4096 2011-01-21 16:18 custom/
drwxr-xr-x 10 root root 61440 2011-03-17 14:21 fr/
Note: fr/ contains core + extra + moh as downloaded from here:
2005 Mar 27
1
Asterisk and XLite on same machine (OSX)?
Dear all,
I have tried to run an asterisk instance together with XLite on a single
machine (a PowerBook).
The intent is to take advantage of IAX connections to easily cross NATs
while traveling.
While the IAX setup proved 'easy', just having to fiddle a little with
working configs at both sides, I did not succeed so far in getting XLite
to connect to the local Asterisk server, AND be
2006 Feb 26
2
Skype vs. an Xlite registered to Asterisk
I have a bunch of road warriors who I've set up with Xlite clients.
Unfortunately
the sound quality has been intermittent at best. Sometimes it's great other
times completely unusable. When it's bad one usually hears harsh static
when the other party speaks or their voice gets "clipped" to static if they
speak too loudly.
Many of these users have migrated to Skype ? much
2010 Dec 06
1
[3102] How to rewrite CID name + number?
Hello
I use the Linksys 3102 to connect Asterisk to a POTS line, and XLite
on XP as an SIP client:
http://img694.imageshack.us/img694/1421/3102asteriskxlitecid.png
The problem is that by default, Asterisk doesn't rewrite the CID name
+ number in incoming calls, so that XLite displays whatever name I
used in the 3102 and the extension the 3102 uses to register with
Asterisk.
How can I tell
2005 May 16
2
NAT and sip issues
I have an asterisk server behind NAT - no audio on the test external calls I
have tried making so far.
Read http://www.voip-info.org/wiki-Asterisk+SIP+NAT+solutions - No solution
evident from there, sounds like I have case 9. I would have thought that all I
would have to do is port foward and have the external IP on the asterisk server,
which I have done
I have fowared 5060UDP, 8000UDP, and
2006 Feb 01
3
XLite dtmf issue?
Hi,
I'm wondering if anyone has experienced an issue with the XLite
softphone and asterisk accepting dtmf? I can listen to my voicemail
perfectly from my hardphone. However when I dial the voicemail number
from my XLite softphone and enter the password at the voicemail prompt,
an error appears vm-incorrect and I get an "Unable to read password"
message on the asterisk console. Has
2010 Feb 25
3
X-Lite won't register
Beginner to Asterisk, but not beginner to VoIP
FreePBX front end running on a dell 1550 and XLite running on a different Woindows XP box
Both boxes connected via switch on same subnet. No NAT involved
On FreePBX I created a new extension 1001 with a SIP password of 1001
On Xlite, username is 1001, password is 1001, authorization user name is 1001, and domain is IP of Free PBX
XLite tries to
2005 Jun 08
5
Xlite not communicating with Asterisk
Dear All,
I have downloaded the xlite version 2.0 for windows and I made the
following conf in the xlite itself as the document suggested in order to
make it work with Asterisk but still it doesn't work as a matter of fact
when I tried to make a tcp dump I can see no packets going between the
windows client and the Asterisk server at all, here is the my conf on
the xlite itself:
in the
2006 Mar 07
1
Setting Vaaibles
Helo List,
First I would like to apologize for my bad spelling as
well as that I did not search the wiki first. I only
have email access at the moment.
I am having trouble setting both variables and global
variables thru an extension.
I am using Asterisk 1.2.4 with Ztdummy on CentOS 3.4
with an Xlite softphone. I have two xlite phones on
diffent computers. One logs in as xlite1 and the other
as
2003 Dec 20
3
iconnect 480 unavailable msgs
Hi guys
i signed up to iconnect a few hours ago to try do some cool stuff. but im having a few problems. Im running asterisk and xp/xlite softphone.. both xp box and asterisk box are on public ips.
The problem is that when i ring anyone in the world it'll ring they'll pickup and i can hear them 100% perfectly/clearly.. but they cant hear me.. occasionaly they can hear something like a
2005 May 18
1
Agent Queues and Sending URLs
Hi guys,
I'm testing the sending of a URL to an XLite softphone when a call is
in queue. See the output of the CLI below:
-- Executing Queue("Zap/69-1", "q_sample|tT|http://
www.google.com/") in new stack
-- Started music on hold, class 'default', on Zap/69-1
-- outgoing agentcall, to agent '1000', on 'Local/
1000@agents-1b94,1'
2005 Mar 23
1
cannot dial any extension except xlite
hi all, was wondering if someone could assist with a slight problem
i'm having. I have asterisk setup with extensions 101 to 109 and am
using xlite, grandstream budgetone, polycom ip500 and a couple of
other phones. the problem is:
1. only the xlite extension (107) can receive calls.
2. all extensions can dial into voicemail and get mwi when msgs are received.
3. when dialing a non-xlite
2009 Nov 01
2
Tutorial for SIP user
Dear all,
I'm trying to setup a SIP call with XLITE using my asterisk PBX, but I have
trouble, I see on XLITE console:
Registration Error: 503 - Service unavailable.
Someone have a tutorial or a step by step description how to do that ?
Thanks in advance
--
Giancarlo Lombardo
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2006 Mar 15
2
Help with Gizmo from outside firewall
I've beaten myself bloody dealing with this one... No luck so far. In
summary, incoming calls from Gizmo establish, but neither get nor send
sound. Outbound calls to Gizmo work fine (well a bit choppy but work)
My thought is that the SIP connection is being made fine, but the RTP
is getting stopped / blocked / misdone somewhere.
Here is the thing:
Asterisk 2.5 on Linux
(No hardware