FOSDEM - Real Time Communications devroom CfP
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Hello Users,
We have Thirdlane Multi tenant PBX system in production. Asterisk version
is 1.6.2.15.
Attendant transfer is working, but blind transfer is not working with
Grandstream (gxp-2000) phone.
We have read from google that it is a bug in Asterisk 1.6.2.15.
We saw the below links:
One of our users recently had a powerfail while connected to our meetme
gateway. (Asterisk 1.4.17 on debian 4.0)
Through the course of it, asterisk never hung up. His system came back
up, and started sending ICMP port unreachables, but the stream went on,
flooding him with silence media stream
Hello,
I'm having an issue with caller ID in voicemail that I'd appreciate
any input on.
I have two sip peers defined as extension 100 and 101 each with
separate voicemail accounts. Each sip peer has its own DID number,
which is established via cid_number = 6021231234.
When a call is
Title: Stylish Vacation
You have told us you would like to receive exciting email offers from us.
The sun is getting hotter, the days are getting longer, and summer
Dear asterisk-us...@lists.digium.com!
Lovers package at discount price!
Discount price store: ID 406858
http://tba.dojmoquj.cn?faz
Pfizer is a licensee of the TRUSTe Privacy Program.
© 2001-2008 Pfizer Inc. All rights reserved.
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To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
I am attempting a fresh install of ASTCC on Ubuntu. Getting install
invalid user as bellow. Has any one seen this? Can some one help out?
/usr/src/astcc# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc
/var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc-admin.cgi
echo | ./astcc.agi /dev/null
Detected dry run!
./astcc-admin.cgi /dev/null
DBI connect('database=;host=','',...) failed: Access denied for user
'root'@'localhost' (using password: NO) at ./astcc-admin.cgi line 69
install -m
invalid user as bellow. Has any one seen this? Can some one help out?
/usr/src/astcc# make install
mkdir -p /var/www
mkdir -p /var/www/html/_astcc
mkdir -p /var/www/cgi-bin/astcc-admin
chmod 755 ./astcc.agi
chmod 755 ./astcc-admin.cgi
echo | ./astcc.agi /dev/null
Detected dry run!
./astcc-admin.cgi /dev
Hi
I have a asterisk with many phones (type=friend)
When I issue the command sip reload some of the phones become unreachable
and they come back just after.
I guess that the sip.conf file is too big and asterisk takes too much time
reloading the entire file.
Is there a way to avoid this
Hi everyone,
I'm completely new to Asterisk and before I buy any card, I would like to ask
for some information.
1. We'll be using analog PSTN phone lines. Is there anything that I should ask
the telecom company before I buy the card? What I mean is whether the card will
be compatible with
Hi everyone,
I'm completely new to Asterisk and before I buy any card, I would like to
ask for some information.
1. We'll be using analog PSTN phone lines. Is there anything that I should
ask the telecom company before I buy the card? What I mean is whether the
card will be compatible with
Dear Asterisk People,
I am having problems putting a SIP image on a 7970. I was wondering if anyone
can help?
First problem is the phone is running version
Load IDJar70.2-5-47-17.sbn
Boot Load ID7970_64054100.bin Version5.0(0.6S)
So I did read that you couldn't simply put the latest SIP
Is anyone successfully passing Outbound CallerID to Teliax?
If so can you please tell me how.
Thanks in advance!
Dan
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
We would like to use Asterisk to deploy VoIP to our broadband internet
access customers.
Which VoIP providers (that are reliable stable) provide wholesale
DID's?
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Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
I have a copy of RH9 and would like to build a Asterisk box for my
office but I do not want to load any unnecessary software.
Can someone provide me a list of required items above a minimal
install. Thanks in advance.
Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
]
[mailto:[EMAIL PROTECTED] On Behalf Of Plexicomm
Admin
Sent: Monday, July 04, 2005 5:06 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Asterisk, RH9, minimal install
I have a copy of RH9 and would like to build a Asterisk box for my
office but I do
We do colocation, but we are on the east coast. What are your specific
needs? -Dan
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Sahil
Gupta
Sent: Monday, July 04, 2005 10:10 PM
To: asterisk-users@lists.digium.com
Cc: asterisk-biz@lists.digium.com
Here are a couple of items I hear people asking for regularly.
- Multi-tenant functionality
- Allow users to change their own preferences via web (call forwarding, MoH,
etc...)
We are two programmers who are passionate for Asterisk and we will be
dedicating the next three months towards
Here are a couple of items I hear people asking for regularly.
- Multi-tenant functionality
- Allow users to change their own preferences via web (call forwarding, MoH,
etc...)
We are two programmers who are passionate for Asterisk and we will be
dedicating the next three months towards
Do you have mpg123 installed?
Is there a .mp3 file available to play in your /var/lib/asterisk/mohmp3
directory?
-daryl
-Original Message-
From: chawki hammoud [EMAIL PROTECTED]
To: Asterisk-Users@lists.digium.com
Cc:
Date: Thu, 19 May 2005 06:03:55 -0700 (PDT)
Subject:
I am using a UTStarcom ATA. I am having the same experience. I can hear the
music via firefly softphone if I call a test MoH extension directly, but
neither the hold button on the softphone, or the hold/flash buttons on a
regular phone (connected through the ATA) seem to work.
When I call a
Can anyone here help me understand what
I missing with this setup. I want to use Asterisk as a feature server only,
speaking only SIP (no IAX), and use SER for registration to minimize
necessary bandwidth.
SIP-phone --SER -- *
-- PSTN Provider -- Regular-phoneRegular-phone
-- PSTN Provider
objective: users connected to box A can dial the extension number of
users connected to box B
boxA at location 1: works fine for internal lan users using the
firefly softphone
boxB at location 2: works fine for internal lan users using the
firefly softphone
Both the boxes have a IAX trunk
after two days of experiments finally decided to go with sipura 2001.
I was wondering to support a 50 people call center do i need 25 sipura
2001 or 50 of these ?
t
On Thu, 24 Mar 2005 16:39:25 -0500, Dana Olson [EMAIL PROTECTED] wrote:
On Thu, 24 Mar 2005 13:03:46 -0700, JD Austin [EMAIL
. Or are there firmware/voice quality/asterisk
integration issues to use both FXS ports on a sipura 2001
simultaneously.
t
On Fri, 25 Mar 2005 14:52:48 -0600, Michael Graves [EMAIL PROTECTED] wrote:
On Fri, 25 Mar 2005 12:46:41 -0800, Sys Admin wrote:
after two days of experiments finally decided to go
if some one was to create a open source IAX client as good/better then
skype, even then a asterisk IAX based network will not be able to
compete with skype. Since asterix is a centralized server regitration
network it can not grow as big as a skype P2P network can grow,
t
On Fri, 25 Mar 2005
would be more then $300,
whats up with that ?
t
On Fri, 25 Mar 2005 15:21:16 -0600, Michael Graves [EMAIL PROTECTED] wrote:
On Fri, 25 Mar 2005 13:14:12 -0800, Sys Admin wrote:
each of the desks already has two rj45 network ports so it makes sense
for me to put a sipura 2001 at each
You think way to small. :(
On Fri, 25 Mar 2005 15:55:54 -0600, Steven Critchfield
[EMAIL PROTECTED] wrote:
On Fri, 2005-03-25 at 13:18 -0800, Sys Admin wrote:
if some one was to create a open source IAX client as good/better then
skype, even then a asterisk IAX based network
just called digium using firefly softphone connected to a asterisk
server using IAX2 they said that the IAXy device is not in stock and
the earliest expected arrival is after a month.
On a dell insipiron 600m laptop with 512 MB RAM each time i maximize
or minimize even a small application like
2005 10:39:54 -0800 (PST), Robert Hajime Lanning
[EMAIL PROTECTED] wrote:
Because the video driver is a kernel thread and not allowed to lag.
That would cause framerate issues with games. :)
oh winderz...
quote who=Sys Admin
On a dell insipiron 600m laptop with 512 MB RAM each time i
so seems like the verdict is go IAXy with a IAX only network ? Most of
the problems of the IAXy device seems like will be fixed with firmware
updates and wont require a hardware update..
this way we get the advantage of a Hardphone (human factor, just feel
good to talk on a real phone) with all
couldnt agree with u more !!
On Wed, 23 Mar 2005 11:15:55 -0800, Robert Goodyear [EMAIL PROTECTED] wrote:
Confidentiality Notice
The information contained in this electronic message and any
attachments to
this message are intended
for the exclusive use of the addressee(s) and
is some one from digium reading this thread. !!
Looks like they have a ready and a big market for this device. And all
they need to do is invest say 6 man months of development effort :)
come on digium do it !!
How about making the firmware open source so we can hack on it ...
t
On Wed, 23
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for IAX vs SIP is there
any reason why i should use SIP anywhere !!
t
___
Asterisk-Users mailing
an update: since it might help others
I did the same make on another machine and it worked fine. So it seems
to be a problem with my tool-chain
t
On Sun, 20 Mar 2005 22:18:50 -0800, Sys Admin [EMAIL PROTECTED] wrote:
I am trying to install asterisk on fedora core 3 these are the steps i took
2 reasons for not using IAX:
A. CDR as part of the media
B. Not good hardphone / softphone
Problem A (CDR as part of the media) I am not worried about too much,
as long as the data is there some parsing will allow me to extract it
and then do what i want to do with it.
Problem B (Not good
btw if there is no good softphone for IAX which does G729, i could
possibly get my company to buy consulting time from say DIAX and make
dante develop it further,
t
On Mon, 21 Mar 2005 11:03:48 -0800, Sys Admin [EMAIL PROTECTED] wrote:
2 reasons for not using IAX:
A. CDR as part of the media
After 20 posts, in 2005 the ideal setup for a new installtion of a 50
user asterisk is:
Option1: IAX2 with softphone firefly
Option2: SIP with softphone
Option3: IAX2 with hardphones (which brand?)
Option4: SIP with hardphones.
Seems like we cannot come to a definite conclusion, poll ?
so the
I am trying to install asterisk on fedora core 3 these are the steps i took:
1. download asterisk-1.0.7.tar.gz
2. make clean and make install and then it gives me these errors:
{standard input}:9975: Error: symbol `i' is already defined
{standard input}:9978: Error: symbol `__result' is already
Hello:
I am looking for alternatives to Broadvoice. Comparability in price but
with much better services (stable) to Asterisk users.
Please email off-list.
Thanks,
Terrell S. Patrick
www.dealerbackup.net
(800) 651-4484
The best time to respond to a disaster is before it happens.
A
Received errors as follows.
[EMAIL PROTECTED] asterisk]# patch channels/chan_sip.c broadvoicesip.txt
patching file channels/chan_sip.c
Hunk #1 succeeded at 229 (offset 13 lines).
Hunk #2 succeeded at 308 (offset -2 lines).
Hunk #3 succeeded at 494 (offset 11 lines).
Hunk #4 succeeded at 489
Seth:
Received an error when applying patch.
$ patch channels/chan_sip.c broadvoicesip.txt
patching file channels/chan_sip.c
Reversed (or previously applied) patch detected! Assume -R? [n]
Apply anyway? [n]
Skipping patch.
17 out of 17 hunks ignored -- saving rejects to file
When using a PRI, after the remote party hangs up, asterisk tries to spawn
a call to the h extension. Is this normal behavior for a pri to try to
call the h extension to try to clean things up?
Call Comes In:
-- Executing Dial(Zap/1-1, SIP/[EMAIL PROTECTED]) in new stack
-- Called
Well, in lieu of dropping us, Broadvox has transferred us to their lab
switch (keeping our DID's in the process).
Now they're complaining that asterisk is sending a Silence-Suppression OFF
request of some sort.
There's no way to turn this on in asterisk is there? (Yes, I know it will
shoot
On Thu, 19 Aug 2004, Andrew Kohlsmith wrote:
Yes, that's what I've told them, too. Do you know of any software that I
can use as a proxy which does support this?
-Dan
On Thursday 19 August 2004 17:59, Dan Mahoney, System Admin wrote:
Now they're complaining that asterisk is sending a Silence
On Thu, 19 Aug 2004, Andrew Kohlsmith wrote:
SO what do the higher-end products use for timing?
-Dan
On Thursday 19 August 2004 17:59, Dan Mahoney, System Admin wrote:
Now they're complaining that asterisk is sending a Silence-Suppression OFF
request of some sort.
There's no way to turn
colors.
The fact that the lines in term.c don't consider screen to be
vt100-compatible is an interesting thing that I don't consider a bug but
merely an oversight.
-Dan Mahoney
On Wed, 2004-08-04 at 17:56, Dan Mahoney, System Admin wrote:
Hey all. I have a color-capable console (color ls works
Guys,
For what it's worth, after months of trying to troubleshoot issues with
them, and after paying them around $2500 for setup and a down payment
(it's unclear what of that will be refunded, if any) BroadVox --
http://www.broadvox.net/ -- decided to terminate our contract without any
valid
On Wed, 11 Aug 2004, Olle E. Johansson wrote:
Dan Mahoney, System Admin wrote:
You start up the phones, they register, all is good. They show up in sip
show peers like thus:
danm/danm65.125.237.91D N 255.255.255.255 5060 OK
(29 ms)
We pass a few calls in and out
Okay, this one is driving me nuts.
I have a fedora core 1 machine running asterisk from CVS. Built last
week. I have a couple of snom phones with the latest firmware.
Here's the issue, it's a wierd one.
You start up the phones, they register, all is good. They show up in sip
show peers like
On Tue, 10 Aug 2004, Todd Lieberman wrote:
Looks like a firewall issue too me.
Some of the snoms are behind NAT. However, my test one was on the same
subnet, and exhibited the same problems.
The asterisk box has firewalling disabled. A firewall issue, I would
think, would not cause a
On Wed, 4 Aug 2004, Steven Critchfield wrote:
On Wed, 2004-08-04 at 17:56, Dan Mahoney, System Admin wrote:
Hey all. I have a color-capable console (color ls works, and I can run
any color-smart program like naim and bitchX), but for some reason the
color in the console for asterisk, whether
Hey all. I have a color-capable console (color ls works, and I can run
any color-smart program like naim and bitchX), but for some reason the
color in the console for asterisk, whether started with -c or
safe_asterisk, isn't working for me.
Any ideas as to why?
I don't think it's my termcap,
Hello All!
I have successfully set up my Cisco 5350 for use with *!
Through direct-inward-dial i have all my users dialing my number placed
in Asterisk.
But I have a problem - one way sound (it IS NOT a codec issue):
When I call the 5350, it connects to the Asterisk, and then to the
Hello, everybody!
I need help connecting my Cisco AS5350 to *.
What i want to do is forward all incoming calls coming from the E1
connected to the AS5350, to my * server, using SIP.
How could this be done?
Greetings,
Doichin Dokov
NetOne - Silistra
Ping times (latency) and bandwidth are really not related unless you are
filling the pipe. Your ping times are too high. My understanding is that
anything over 100ms is not good. Your problem probably lies with too many
hops or slow or overburdened router along the path.
From a windows box run
When an NBX100 is upgraded a .tar file is uploaded and installed on the box.
Inside that tar file is the firmware for the phones which is downloaded when
the phone boots. If someone can provide the last SIP firmware I will
replace the phone firmware in the tar file with the SIP code and see if
works with voicepulse too
- Original Message -
From: Jeremy McNamara [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, February 09, 2004 6:39 PM
Subject: Re: [Asterisk-Users] Dialing 800 numbers with VOIP
Matt Lawson wrote:
Hmm. Both Voicepulse and Nufone don't seem to be
Bogen
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Sunday, January 18, 2004 3:35 PM
Subject: Re: [Asterisk-Users] Zone Paging
Alfred R. Nurnberger wrote:
There are a number of paging interfaces available which connect to a
regular
phone
I just installed the new version of SJPhone and it
appears that it cannot work with * anymore?
everything is free or the cost of shipping if you think...
dont worry, newbs will land at my forums but i still wanna know if i can cut
and paste FAQs and the like. I plan on it so sue me, rofl.
- Original Message -
From: [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January
Just refund the guy his money...
- Original Message -
From: John Brown (CV) [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 2:46 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
For the list,
Mike received a partial order shipped 15-Dec, SN ending
Number, Auto Attendent answers
Pressing 0 takes you to a operator
M-F 9-5 MDT (GMT-7)
orders at chagres dot net gets email into the order admin
which replies within 1 biz day
and you should get a auto reply
Sorry, but how can you ID his inbound packets?
- Original Message -
From: admin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, January 10, 2004 3:17 PM
Subject: Re: [Asterisk-Users] Chagres Technologies, Inc
Just refund the guy his money...
- Original Message
I work for an interconnect that sells 3com and NEC. When I made this
project my own and followed through to show my boss, he said, this is going
to ruin our industry
If that is the case then so be it. Same with mp3s and the music industry.
Had they embraced the technology, everyone could be
Great interview with Nicolas-Peter Pohland CEO of SNOM
Chief Executive Officer
- Original Message -
From: Steve Totaro [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Saturday, December 06, 2003 1:01 PM
Subject: FW: snom in Wallstreet report
-Original Message-
From: Robert
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