Hello everyone,
I see that the yum install freepbx from Digium repository actually
installs the latest FreePBX which is nice. However, I don't see the old FOP
in FreePBX anymore. Is there a way to install FOP or FOP2 through
repository?
Thanks,
--
We have had success with this:
https://sites.google.com/site/a2billing2asterisk/
On Fri, Mar 16, 2012 at 1:28 PM, Tahar .H harazta...@gmail.com wrote:
hi folks,
i was wondering if some one has a2billing script,which can be used to
install a2billing easly ?
thanks in advance
--
*
I would be more interested in a system where quality routes are tested with
different providers because rate really doesn't matter if a call can't be
placed or if a destination is a fake one. We have seen many fake
destinations with top tier providers but they had the best rates so the
strategy to
A2Billing doesn't do that. A2Billing in fact has a lot of shortcomings one
of which is this exact issue.
I would suggest running rate sheets against each other for finding true LCR
and then only uploading the rates that are cheaper into the system. In most
cases there are not such high
Our experience has always been, with all versions of Asterisk, that when
you do an originate command from Asterisk CLI then it stops showing CLI
verbose and one has to open a new terminal to see the verbose. Logging
status is in disarray in all version. These behaviours breaks security
tools
Downgrade actually worked fine in this instance - from 1.8.9.2 to 1.8.9.1
and I concluded it wasn't Asterisk that was the issue. Thanks Patrick.
It would be great to keep this feature stable. As it helps in case of
regressions, etc...
On Mon, Feb 27, 2012 at 10:09 AM, Kevin P. Fleming
jpar...@digium.com wrote:
On 02/23/2012 10:09 AM, Ast Coder wrote:
Hi,
I have followed instruction
on
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisitesto
add Digium Asterisk repositories but doing a, yum search asterisk only
shows
me Asterisk
Hi,
I have followed instruction on
https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages#AsteriskPackages-Prerequisites
to
add Digium Asterisk repositories but doing a, yum search asterisk only
shows me Asterisk 1.4, 1.6, and 1.8. There is no Asterisk 10 and yum
install asterisk10 fails
Hi everyone,
I got HTTP AMI working fine here. For example this dials 1-415-999- and
then sends to Extension @from-internal:
http://192.168.0.100:8088/asterisk/manager?action=commandoriginateDAHDI/g0/1415999extension@from-internal
However, I want to have some control over this
India TRAI rules doesn't allow for CLID setting. They are backwards minded.
If you ever get them to do it let me know ;)
-Bruce
On Mon, Feb 13, 2012 at 8:18 AM, Steven Howes steve-li...@geekinter.netwrote:
On 13 Feb 2012, at 12:06, virendra bhati wrote:
You can't set callerid for outgoing
Hi Sammy,
Yes, that's what I have brain-stormed as well. I would very much appreciate
sharing the code as I can try to improve it. I will put a GUI to it so that
admin can insert campaign numbers and agents be able to see what percentage
of the campaign is done.
Can you share it with me already?
in setting up 79xx on sccp, with sccp-b library, and tftp
server, which part is the main problem for you?
best
On Wed, Feb 16, 2011 at 3:10 PM, Andrew Latham lath...@gmail.com wrote:
On Wed, Feb 16, 2011 at 7:32 AM, ast guy ast...@gmail.com wrote:
Hi,
Anyone who has deployed Cisco 7945G
Hi,
Anyone who has deployed Cisco 7945G phone with asterisk, kindly share your
experience.
/ag
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New to Asterisk? Join us for a live introductory webinar
lolz, I agree. Better to spend more and use it for some time. It is not a
big installation about 4-5 extension so can spare the budget for it easily.
/Khurram
On Sun, Feb 13, 2011 at 5:45 PM, Michael Graves mgra...@mstvp.com wrote:
On Sun, 13 Feb 2011 09:28:19 -0700, Joel Maslak wrote:
My
Thanks Gordon, Grandstream is in my purchase list : )
/ag
On Sun, Feb 13, 2011 at 10:58 AM, Gordon Henderson
gordon+aster...@drogon.net wrote:
On Sat, 12 Feb 2011, ast guy wrote:
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking
Probably I will go with cisco 7945g I hope its support is good with
asterisk. Have you used it ? is it simple in configuration?
/Khurram
On Sat, Feb 12, 2011 at 1:47 PM, Andrew Latham lath...@gmail.com wrote:
On Sat, Feb 12, 2011 at 9:31 AM, ast guy ast...@gmail.com wrote:
Hi,
I have
Hi,
I have been out of touch with asterisk for quit some time and needed some
recommendations. I am looking for SIP hardphone that works well with
asterisk server.
Pls suggest.
cheers
/ag
--
_
-- Bandwidth and Colocation
of Asterisk. It works over HTTP now and
with an FTP or TFTP proxy can work over multiple protocols at once.
Read More:
https://wiki.asterisk.org/wiki/display/AST/Phone+Provisioning+in+Asterisk
I added example snom support and will have to start a review board for
adding Cisco, Aastra and others
Hi,
I have made a fresh install of asterisk-1.6.2.10 and when I register my
soft phone it gives following error. Rest are default configurations.
32.454370 MY_IP - ASTERISK_SERVER_IP SIP Request: REGISTER
sip:ASTERISK_SERVER_IP
32.454505 67.19.43.202 - MY_IP SIP Status: 401 Unauthorized(0
Hi,
I am facing terrible issue regarding no audio/voice on both sides. I am
using g729 codec on two machines and carrier also supports g729 codec. I can
see the RTP traffic flowing but there is no audio.
Call is going from Server 1 to Server 2. I can see the established SIP
channels on Server but
Hi,
I am using codec g729 on two asterisk machines, but when call is forwarded
from asterisk server 1 (32bit) to server 2(64bit). server 1 outputs
following error and there is no audio. Also the IVRs being played have
choppy voice.
Insufficient information for SDP (m = 'audio RTP/AVP 18
On Tue, Nov 24, 2009 at 10:49 PM, Miguel Molina mmol...@millenium.com.cowrote:
ast guy escribió:
Hi,
I am using codec g729 on two asterisk machines, but when call is
forwarded from asterisk server 1 (32bit) to server 2(64bit). server 1
outputs following error and there is no audio
This one is manual way of doing it.
You can get more details at page
http://www.voip-info.org/wiki/view/Asterisk+Starting+and+Stopping
It has provided the init.d scripts where you can automate the process.
/ag
On Sat, Nov 14, 2009 at 7:44 AM, Yawar Hadi yawarh...@gmail.com wrote:
cli stop
Hi,
I am trying to install asterisk-1.2.34 but facing following issue. I have
gone through it and found that there are files in /usr/lib
libpq.a libpq.so libpq.so.4libpq.so.4.1
make[1]: Entering directory `/usr/src/asterisk-1.2.34/cdr'
gcc -shared -Xlinker -x -o cdr_pgsql.so
(alternative title - what did I do wrong? or suggestions to make this
work)
Thought I'd try 1.6 beta5 (and 1.4.18 didn't want to compile vpb
/usr/lib/gcc/i386-redhat-linux/4.1.2/../../../../include/c++/4.1.2/i386-redhat-linux/bits/gthr-default.h:48:
error: â does not name a type )
1.6 did
Hi,
I have register a sip user to sip server. I can see after registration * is
sending periodic SIP Request: OPTIONS messages to server. but it's not
getting back any response that should be SIP 200/OK as the documents say.
3130.299707 192.168.2.113 - 58.ab.cd.ef SIP Request: OPTIONS sip:
So that´s why I´ve always get a red bar on home screen of the Trixbox?
Phisical memory is always at top most use, near 100% (green bar turns red on
high level of memory use), and below it there is Kernel / Application,
Buffers, Cached memory uses.
tks,
On Feb 13, 2008 12:51 PM, Atis
why don't you write an AGI which talks to asterisk manager API 5038 port and
executes the asterisk commands. You execute asterisk command via agi not
using system command
-ag
On Feb 11, 2008 11:24 AM, Rob Hillis [EMAIL PROTECTED] wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi guys,
Hi,
I have a scenario that * Server A ( behaving as client) has sip peers, P1,
P2, P3 with different contexts. Peers register to another * or any other SIP
server. Using realtime * I am able to create a peer entry in sip buddies
table and a register statement in sip.conf on client side and it's
? It's unnecessary.
I believe that Dial(SIP/gs102/1234) will achieve what you want.
ast guy wrote:
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...
Dial(SIP/gs102:[EMAIL PROTECTED]);
User on sip server
Hi,
I have installed asterisk real time and sip buddies information is being
stored in DB. Now I have a question,
Asterisk Realtime Server -A
Third party SIP server-B
Question: Is there any configuration in * RT that it can register with
defined sip user on Server-B
I was only able to find sip
Hi,
I'm trying to call a SIP server while providing the SIP server
username/password in dial string but it's not working ...
Dial(SIP/gs102:[EMAIL PROTECTED]);
User on sip server (192.168.2.81):
[gs102]
disallow=all
allow=ulaw
allow=alaw
type=friend
username=gs102
secret=test
host=dynamic
Hi,
Let me explain what I'm looking for a solution using asterisk.
I have one third party SIP based server (A) and on Asterisk server (B).
1. Extension-1 -- Server A calls Server B.
2. Server B does some processing and calls/sends back to Server A ---
Extension-2
3. SIP session has been
there is no /proc/zap folder .. can you tell how can I create /dev
nodes. I have tested the same configurations on FC5 and these device
links were created ...
drwxr-xr-x 2 root root 160 Jan 17 10:59 .
drwxr-xr-x 13 root root 3640 Jan 17 11:00 ..
crw--- 1 root root 196, 1 Jan 17
*Walter Willis,
*Thanks a lot, got the commands from zap Makefile and it worked, now can
create conference room, my question still stands why it didn't create
itself. Will go through make file to get an answer to that.
Anyone else facing the issue can resolve by running following commands
Hi,
I'm using zaptel-1.2.22.1 with asterisk-1.2.10 and following steps to
make zaptel working...
OS is gentoo linux 2006.1
Hardware:
-
:05:01.0 Communication controller: Tiger Jet Network Inc. Tiger3XX
Modem/ISDN interface
Subsystem: Unknown device 8085:0003
Flags:
Hi,
Can any one please guide how do I handle the SIP phone-context URI
parameter. I got following traces..
9.191690 IP_A - IP_B SIP/SDP Request: INVITE
sip:12599;[EMAIL PROTECTED]:5060;user=phone, with session
description
9.191942 IP_B - IP_A SIP Status: 404 Not Found
9.195656 IP_A -
This 2 line code is doing what I wanted.
exten = 200,1,voicemail(200)
exten = 200,2,Hangup
What I've been told is that they want the 20 year old phone system to
light up the message bulb. (yea, a filament bulb, not an LED) To do
this you pick up on the line that goes into Asterisk and do a:
So I've got a Voicetronix card and it looks like the kernel driver works.
Other than the 0's for ID info.
vpb: Driver Version = 4.0
vpb: major = 251
vpb: tmp [0xfc8fec00] dev-res3 [0xfc8fec00]
vpb: tmp [0xfc8c] dev-res2 [0xfc8c]
vpb: 1WS Write cycle
vpb: Manufactured 00/00/
vpb: Card
Hi,
Just need to confirm whether dial() command provided options
h: Allow the callee to hang up by dialing *
H: Allow the caller to hang up by dialing *
work for SIP channels as well ?
-ag
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Hi,
It's really weired issue,I'm facing with asterisk-1.2.10 version. I
see SIP call sessions stuck in asterisk for hours and then somehow get
released. There happens to be an issue with BYE/CANCEL release msgs
b/w sip entities. Has anyone faced this issue before also rtptimeout
option given in
Hi,
I would like to know what is Asterisk Dynamic Loader. Let me
explain what I'm about to ask.
I have three Asterisk servers running my in-house built app_xyz.so
application. Now what I do to save time is compile application on one
server and scp app_xyz.so on rest of servers. All servers have
Hi,
I'm running Asterisk 1.2.10 on gentoo linux and facing strange kind of issue.
1. chan_sip.so takes about 10 secs to load up when asterisk starts.
2. When I dialout using SIP it takes 20 secs to output -- Called SIP
[EMAIL PROTECTED] and get ring back from B party...
Is there any config that
Hi i just setup asterisk and tried to install the digium g729 codec.
it works ok using stable but with SVN i get a core dump with the error,
'missing mod_data for codec_g729a.so'
i had a quick look though the archives but all i came up with is this,
Hi,
I have downloaded iaxcomm version iaxcomm-lin-1.0rc3, when I try to
execute it it gives following error.
# ./iaxcomm
Error wxWindows Fatal Error : Couldn't Initialize IAX Client .
any idea what's going wrong ?
-ag
___
--Bandwidth and Colocation
I was going through Zap channel page on voip-info.org
Zap/channel-instance
channel is the channel number and instance is a number from 1 to 3
representing which of up to 3 logical channels associated with a
single physical channel this is.
for what purpose logical channels are used?
Hi, I got this message in the asterisk console while sending the dtmf from a phone.Dec 8 14:55:50 WARNING[29098]: codec_ilbc.c:163 ilbctolin_framein: Huh? An ilbc frame that isn't a multiple of 50 bytes long from RTP (4)? Please help me to solve this.Thanks Jibu
Yahoo! India
Eric Wieling aka ManxPower [EMAIL PROTECTED] uttered the following thing:
Dean Collins wrote:
I've just gotten off a skype conference call and it pisses me off that
the quality of skype is higher than my asterisk calls.
Is there such a thing as a super high bandwidth codec?
Asterisk does
user monitor application
-Wix
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http://lists.digium.com/mailman/listinfo/asterisk-users
Its not a moderator issue, it is a bounce issue, Mailman can be setup to
deal with this. However, if this guy bounces messages, just remove him
from the list.
On Sat, 11 Dec 2004, Leif Madsen wrote:
On Sun, 12 Dec 2004 04:36:56 +1300, Matt Riddell
[EMAIL PROTECTED] wrote:
Wilson Pickett
root/admin
On Fri, 3 Dec 2004, Kanuri, Seshu (Company IT) wrote:
Tell me which one can get me access to the LinkSys Linux using SSH?
Does Satori has this feature? I am not so concerned with Voice Shaping
and QOS at this time, but more interested in converting this into a
Linux box that is
Hi all,
I know that this has been passed around before, and I know that it
happens about every 3 months or so, but evertime the answers change, so I
thought I would pass it around again.
A company I work for has 3 incomming lines and 4 phones. They require
voicemail and MOH.
Their phone
On Mon, 29 Nov 2004, Gregory Junker wrote:
My question is would you guys setup an anolog system or VOIP for the
phones. There is not a local VOIP provider in our area, so we can not
port the 3 pots lines.
I would use a 3-port FXO card (for the incoming lines) and a 4-port FXS
card (for
don't waste money :)
I would never think that spending money with the people who make asterisk
is a waste.
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Hi all,
I have recently installed a Wildcard E100P. Everything seems find except
that once in a while I would experience dropped call. I looked at the log
and found that when the call was dropped there was red alarm and the entire
channel in the span was reseted. If there was any call during
If you look at their website it clearly says IAX2
http://ipphone.eezeephone.com/
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Ron,
You hit the best application of GPL. Encryption program can be completely
open source without losing the privacy of the encrypted data. You can
either use one way encryption such as any UNIX password system or
public-private key pair as PGP. They are all open source, but it is
On Tue, 12 Oct 2004, Jim Van Meggelen wrote:
The bounty seems to be at $3000 so far:
http://www.voip-info.org/tiki-index.php?page=Asterisk%20bounty%20SMDI
I do not know for sure, but it seems easier enough to write this as
another application that just scans the
If you are going to buy it new. It should not be a problem at all. Just
order the SIP software with the phone.
Your order should be somthing like this.
CP-7905G - Cisco IP Phone 7905G, Global
SW-SMH-UL-7905 - SIP or H.323 license for single 7905 IP phone
CON-SNT-CP7905 - 8x5xNBD Svc, Cisco IP
Hi all,
I am new to asterisk. Is there any way to get dtmf in
between an active conversation. For eg. If phone1 has
made a connection with phone1, then is it possible to
handle all the dtmf send between them in
extensions.conf.
Thanks
Jibu
I know this has been gone over before but
It seems that most of the traffic to this list is the same 10 questions
being asked over and over again. What if someone (with some programing
skill) wrote a script so that when someone posted to the list, it would
search the wiki and google and
About twice a week we have a caller that comes in and hangs up on
voicemail. We have 2 x100ps each with their own irq. When the caller
hangs up asterisk does not release the line. The line rings busy,
sometimes I can do a soft hangup Zap/1 and release the line sometimes I
have stop
On Mon, 26 Jul 2004, Jeremy McNamara wrote:
Ok, then someone has to step up and make absolutely sure Asterisk has
valid up-to-date documentation that newbies know how to find. Then the
link(s) to that documentation should always show up in the topic of said
IRC channels.
In my book,
On Mon, 19 Apr 2004, Rich Adamson wrote:
Maybe a couple of us should write a whitepaper for beginners on the topic.
Yes Please do
Rich
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Please send me a copy
On Thu, 1 Apr 2004, Leo Ann Boon wrote:
Figured this might be interesting to some on the list. I've written a
small tool to combine the 2 res_monitor created wav files into a single
compressed (IMA ADPCM 8KB/s) stereo file. During playback, you can
select the party
You need to turn register globals = on
in your php.ini file, HOWEVER, this can make things VERY insecure if you
have other php scripts that are not thought out right.
On Wed, 24 Mar 2004, Peer Oliver schmidt wrote:
Wipeout wrote:
Another thing I had to do was changing the defines.php
You need a license to use SIP firmware on a cisco phone. If you are not
using one, and you are using the phone in production, then you are in
violation of the software agreement.
Michael
On Tue, 23 Mar 2004, Joe Dennick wrote:
I just bought about 20 new Cisco 7960s for $350 (USD). I
Removing the reply to all is a bad idea, How are you going to be able to
tell someone to search the list, if they don't reply to the list. People
will answer the same questions over and over again, use your mail software
and write some filters, or remove yourself from this list, and use the IRC
How can I get the latest CVS of chan_sccp
The way described on Zozos webpage seems not to work:
[EMAIL PROTECTED]:~ export CVSROOT=:pserver:[EMAIL PROTECTED]:/var/lib/cvs/
[EMAIL PROTECTED]:~ cvs login
(Logging in to [EMAIL PROTECTED])
CVS password:
/var/lib/cvs/: no such repository
cvs [login
I have used both VoicePluse and Nufone. I have to say that the support and
the service I have gotten from NuFone is second to none. They are quick
to respond, they had me up in no time.
On Wed, 17 Mar 2004, Matthew Marlowe wrote:
Try searching the forums or emailing sales, or calling
Hi,
I want to use my Cisco7960 with sccp. Current firmware version is 5.05,
chan_sccp Version is 0.2
I attached my sccp.conf. A lot of things doesn't work:
- speeddials are ignored, nothing is display
- Description in the top line of the display is jensp and not Jens-7960
- the first two line
SSH
On Mon, 8 Mar 2004, hank smith wrote:
is there a program that I can install on my linux box so I can configure the
pbx from the internet from my windows box so I don't have to work with
config files?
thanks
hank
- Original Message -
From: Steve Underwood [EMAIL PROTECTED]
To:
What about Zap Channels?
On Sat, 6 Mar 2004, Tim Sailer wrote:
On Sat, Mar 06, 2004 at 09:26:28PM -, Scott Stingel wrote:
cool! I like the light bulbs. (and the name)
They were something I had in an old image collection. I'll
hvae to find something a bit more 'modern'. :)
nice
I am running about 300 calls at the same time though pbx_spool.
I am getting these error messages. Where you see the Call failed to go
though reason 5, the call is dropped and never tried again. I looked at
the source code in ./pbx/pbx_spool.c and on line 199 I see the message but
I can't
Has anyone had good or bad experiance with http://www.oneunified.net. I
need a DID for incomming calls only. Nufone does not have service in my area(614-XXX)
:(
Anyone have worked with these people. Good comments bad comments.
Should we create a area in the WIKI for all of the VOIP providers
On Wed, 4 Feb 2004, John Todd wrote:
At 10:18 AM +0100 2/4/04, Andy Powell wrote:
lo,
Is there a single central location for code and applications other
than CVS? I'm talking about code that can't/wont be included in CVS
for various reasons? Does the wiki have this sort of thing? I've
Nufone is setup for it and it works great
On Wed, 4 Feb 2004, Stephen R.
Besch wrote:
Chris Clifton wrote:
The majority of sip to pstn gateway providers (vonage, voicepulse, and
others) appear to be setup for a one line only type of set up. Their web
sites seem to be heavily geared for
Why is there still MYSQL stuff in there, I thought we had to remove
that.
On Fri, 23 Jan 2004, Kannaiyan Natesan wrote:
If you are not users from mysql database then you can disable in the
makefile.
For this,
USE_MYSQL_FRIENDS=1
change it to
USE_MYSQL_FRIENDS=0
You won't get
Hello All,
I'm having a problem with g729 codec. Asterisk CVS-08/18/03 is running
without problems if i don't load this codec, but when i try i'm getting
this messages:
== Registered application 'DateTime'
[codec_g729b.so] = (Annex B (floating point) G.729/PCM16 Codec
Translator)
Cannot
I'm using version from
ftp.digium.com/pub/telephony/asterisk/g729/new_codec_binary
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de John Todd
Enviado el: jueves, 11 de septiembre de 2003 11:48
Para: [EMAIL PROTECTED]
Asunto: Re: [Asterisk-Users] Error
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