Hi,
I have been headbanging with asterisk and Jack for a while, decited to
ask other linuxists for an advice.
The problem is that Jack is compiled from source (0.118) in /usr/local/, but
menuselect says XXX for it (cannot enable it). I need jack...
Otherwise I will inotify Monitor WAVs, what is
Motiejus Jakštys desired@gmail.com wrote:
Hi,
I have been headbanging with asterisk and Jack for a while, decited to
ask other linuxists for an advice.
The problem is that Jack is compiled from source (0.118) in /usr/local/, but
menuselect says XXX for it (cannot enable it). I need
Tried the following, both did not work:
jack:
./configure --prefix=/usr make sudo make install
./configure --disable-xmldoc make menuselect - same problem (XXX app_jack)
Jack installed in /usr/local/
./configure --prefix=/usr/local make menuselect - same problem (XXX app_jack)
ran make
Opened a bug report for it:
https://issues.asterisk.org/view.php?id=17402
2010/5/26 Motiejus Jakštys desired@gmail.com:
Tried the following, both did not work:
jack:
./configure --prefix=/usr make sudo make install
./configure --disable-xmldoc make menuselect - same problem (XXX
Hello,
I'm in a bit of a fix. We have a particular Windows based softswitch which is
has its SIP and H323 ports hardcoded to listen on a particular IP address. The
problem is that the ISP is having major issues and we can no longer depend on
them for service. The softswitch will not listen on
Assume previous IP is LAN. Forwarding public IP ports to LAN is
straighforward. However, with SIP headers you will (don't know H323)
have to modify outgoing SIP headers: replace LAN ip with WAN ip.
For callers you have to substitute RTP destination IP
For callees you have to substitute RTP source
Try a Cisco ASA. It will rewrite the headers if configured properly.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Motiejus Jakštys
Sent: 26 May 2010 14:17
To: Asterisk Users Mailing List - Non-Commercial
Is there a tool that will allow me to automatically change sip headers in
realtime?
--- On Wed, 26/5/10, Motiejus Jakštys desired@gmail.com wrote:
From: Motiejus Jakštys desired@gmail.com
Subject: Re: [asterisk-users] Help with IP Routing
To: Asterisk Users Mailing List -
Hello,
if the remote side (the public IP side) is capable to do
something like asterisk's nat=yes (in sip.conf), than
a mascerading router (like every cheap DSL router) would
do enough NAT do let SIP work.
If the remote side does not support that nat-hack (which
is not SIP standard), than you
- Nivin Kumar nivinkuma...@yahoo.in wrote:
Hello,
I'm in a bit of a fix. We have a particular Windows based softswitch which is
has its SIP and H323 ports hardcoded to listen on a particular IP address. The
problem is that the ISP is having major issues and we can no longer depend on
Nivin Kumar schrieb:
Is there a tool that will allow me to automatically change sip headers
in realtime?
Hi,
imho changing the SIP headers will not be sufficient, since
the old IP addresses are now private IP addresses (only in
your network, outside, there are still public, but pointing
not to
Hello everyone,
any help please
I have asterisk installed in our call centre with aheeva platform and
centos linux,
We have 2 access provider I have configured the
etc/asterisk/extensions.conf in order to do the routing of calls
exten = _0612.,1,Set(CALLERID(number)=520460587)
Didn't realize you were so sensitive. My apologies!
The switch in question is called VoipSwitch. It's ok...we use it mainly for
billing. Most of traffic is carried on Asterisk and handed off to this
voipswitch for billing purposes. I've added OT in the subject. I posted it
here because I
Yes, the both extensions are SIP.
The problem to get the "core show channels" output its happen too
fast, so I cant get the output at the moment of the call...
I have the log of CLI output, with all types log enables (WARNING,
NOTICE, DEBUG), but nothing of unusual in the log shows.
Hi,
This is a bit off-topic, but still related to telephony. Is there a
barebones TAPI driver that exists that would allow me to call up a command
line with, as parameter, the number to dial.
For exemple, Outlook integrates with TAPI, so that TAPI driver would allow
me to call my own app
salaheddine elharit wrote:
G2 is for the second provider and g1 for the first provider even I
configured the extensios.conf I have some calls passed from g1
instead g2
Any help please will be appreciated
Maybe if you asked a question, something could help. But, as it is
Doug, did you cancel your psychic friend's subscription? All programmers
are supposed to be able to determine intent without full information :)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Doug Lytle
Sent:
Danny Nicholas wrote:
Doug, did you cancel your psychic friend's subscription? All programmers
are supposed to be able to determine intent without full information :)
I had too! I'm on a budget and it was costing me more then my cable bill.
Doug
--
Ben Franklin quote:
Those who
GIYF - try this link
http://www.voip-info.org/wiki/view/Asterisk+TAPI
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Mike
Sent: Wednesday, May 26, 2010 10:00 AM
To: 'Asterisk Users Mailing List - Non-Commercial
On 26 May 2010 15:59, Mike l...@virtutel.ca wrote:
Hi,
This is a bit off-topic, but still related to telephony. Is there a
barebones TAPI driver that exists that would allow me to call up a command
line with, as parameter, the number to dial.
There is a command-line tool dialer.exe that
Thanks, will take a look. Althought none of those things seem to allow me
to call up my own handler for calls, does it? Or am I misreading?
Mike
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Danny Nicholas
Sent: Wednesday, May
Hello
I'm trying to install Dahdi through source code on a Fedora 13 host
to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv
isn't happy.
1. After successfully running make all; make install; make config, I
edited /etc/dahdi/system.conf thusly:
loadzone=fr
Has anyone written a better AMD than the default AMD? The existing AMD
works great but it has a few shortcomings...
I do know about Sangoma but am just looking for a better AMD module.
Thanks,
John
--
_
-- Bandwidth
On Wed, 26 May 2010 17:17:08 +0200, Vincent codecompl...@free.fr
wrote:
I'm trying to install Dahdi through source code on a Fedora 13 host
to use an OpenVox PCI card with a single FXO port, but dahdi_cfg -vv
isn't happy.
More information, as I investigate:
# vi
Skip the whole NAT scenario.
Put up an asterisk box with two network interfaces. One interface
connects to the real world on your new IP address from your new ISP.
The other interface can be on the same subnet as the windows box that
you can't change. Set up a SIP trunk to your Windows
Hello All
i have set all extensions for 2 providers in dialplan.conf and
extensions.conf
the problem is all numbers take the same provider
when i change the g1 with g2 all the phones numbers take the secend
provider
; Outbound dial context
[aheeva_ccs]
; If we are dialing out through
The Asterisk Development Team has announced the release of version
1.4.11 of libpri. This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/libpri/
This release contains many fixes and new features, among them being:
1.) Support for NT-PTMP BRI links,
Hi Motiejus!
If all else fails for the moment, it should be quite simple to move JACK.
Move all jack applications from /usr/local/bin to /usr/bin.
In /usr/local/lib move the dir jack and libjack* to /usr/lib.
That should be it for the moment. another thing is to hack the JACK
confiugre
I dont know, maybe I am missing it. I see nothing off the top of my head that
shows you attempting to dial out 2 different providers or fail between them.
Both times you have posted code I see a dial command set to go to a single Zap
Group, and no failure code or Prefix that determines how or
Something new to me. Recently installed a 1.4.30 box for a small office
with four POTS lines in a hunt (Digium TDM410P). Had the telco put a
call forward option on the main line of the hunt. They dial a feature
code from their desk phones (Polycom IP450) that results in forwarding the
main
What version of asterisk are you running.
What shortcomings are you experiencing in AMD?
What type of tuning have you done or settings are you using with AMD?
What are you doing after you run AMD on the call? If the call is human you do X
if its not you do Y? Are these AGI's or Goto's or???
--
On 05/26/2010 11:36 AM, Jeff LaCoursiere wrote:
So to the question - can the TDM410P somehow tell the difference between a
ring splash and an actual inbound call? I think in the meantime I will
send inbound POTS calls to an auto attendant that will eventually hang up,
but would love a
The ring splash is a long standing feature of call forwarding.
Of course somewhere in the Asterisk code a change could be made to extend
the time required to detect a valid ring.
But, how about just unplugging the pots lines from the PBX with a quick
restore ability? Unplug lines at the NID, or
Hi List,
Our company has several small distributed offices we would like to
inter-connect with bridged VPN a single subnet (last example in
http://www.shorewall.net/OPENVPN.html). We have SIP phones in every
office (up to 5) so we can use SIP without any NATing and securely.
Max theoretical
Just set the POTS lines to answer after a second ring rather than after
the first. Problem solved.
On 5/26/2010 11:36 AM, Jeff LaCoursiere wrote:
Something new to me. Recently installed a 1.4.30 box for a small office
with four POTS lines in a hunt (Digium TDM410P). Had the telco put a
Strangely enough I have used this many times with our POTS from ATT. We get
ring splash, but didnt get a ghost ring into the system, just the valid ring
that was redirected to the VoIP lines after forwarding. Although I think i had
the Asterisk-GUI creating the dialplan on these systems, not
I use openvpn for VOIP traffic all the time. It's not a commercial
application, and only one simultaneous call usually on each vpn link,
but I even have a VPN client on a Linksys WRT-54g wireless router with
1 phone behind it - it works flawlessly, so it does not take a lot of
CPU to run a vpn
Thanks for the update. How to upgrade to the latest stable release without
compliling Asterisk again? Can you please explain and detail the commands?
We are running PBXinaFlash with LibPRI 1.4.10.1 which gives lots of
problems.
Thanks
On Wed, May 26, 2010 at 12:27 PM, Asterisk Development Team
Hello,
I have a few entries for sip peers in sip.conf with different name and
username, like
[TestSIPUser]
type=peer
host=dynamic
username=testuser
secret=1234
context=test_context
[TestNewUser]
type=peer
host=dynamic
username=newsipuser
secret=3456
context=test_context
When a call is made
I might be wrong, but I think that adding fullname=xxx to the context will
populate CALLERID(name)
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Deepesh D
Sent: Wednesday, May 26, 2010 12:18 PM
To:
On Wed, May 26, 2010 at 8:01 PM, Andrew Hakman andrew.hak...@gmail.com wrote:
I use openvpn for VOIP traffic all the time. It's not a commercial
application, and only one simultaneous call usually on each vpn link,
but I even have a VPN client on a Linksys WRT-54g wireless router with
1 phone
I have several Atom based boxes running OpenVPN and processing up to six
simultaneous calls over it with no issues. I am quite sure it could do
more. Load is still at .2 :)
j
On Wed, 26 May 2010, Andrew Hakman wrote:
I use openvpn for VOIP traffic all the time. It's not a commercial
On Wed, 2010-05-26 at 22:48 +0530, Deepesh D wrote:
When a call is made from any of these peers I want to get the username
of the peer.
for eg:- If a call is being made from 'TestSIPUser' then I want to be
able to get the value 'testuser'
I can think of two ways of doing this. The first is
-Original Message-
From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-
boun...@lists.digium.com] On Behalf Of Trevor Benson
What version of asterisk are you running.
1.6.0.28.
What shortcomings are you experiencing in AMD?
Was thinking that an asynchronous AMD
On Wed, 26 May 2010, Brent Davidson wrote:
Just set the POTS lines to answer after a second ring rather than after
the first. Problem solved.
Now that sounds like a good plan. But a quick look through the options in
zapata.conf don't show any kind of option for waiting before pickup.
- Jeff LaCoursiere j...@jeff.net wrote:
On Wed, 26 May 2010, Brent Davidson wrote:
Just set the POTS lines to answer after a second ring rather than
after
the first. Problem solved.
Now that sounds like a good plan. But a quick look through the
options in
zapata.conf don't show
Pressing hold on the telephone set may not be sending hold to Asterisk
to trigger the correct action. You can verify this from the CLI.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur hasan
Sent: Wednesday, May 26,
Hello
Yesterday, i brought linksys PAP2 and have success with that. The only thing
that does not go well is the music on hold. When i press 'hold' button from the
telephone set instead of playing the music on hold that i have setup in
Asterisk, Telephone Set plays its own MOH. Is there any
Hello Group,
some strange problem i have on my setup.
If a second caller entering a meetme conference dropping the first one.
my setup is using asterisk 1.6.2.6 and dahdi 2.1.1.1 with realtime, the
conference room numbers storing in a mysql database.
the calls came from a sip provider. there are
Hi,
I've noticed that if a phone goes UNREACHABLE while it is Ringing,
when the phone comes back, Asterisk will not clear the channel that
was created, so it still thinks it is in the Ringing state.
The only way to clear this is to do a soft hangup on the SIP channel
or to restart Asterisk.
Hello
Ya you are right moh in Asterisk is not triggered. Is there any solution to
that ?
Regards
Taimur Hasan
-THQ- !!!ONE
From: da...@debsinc.com
To: asterisk-users@lists.digium.com
Date: Wed, 26 May 2010 13:39:00 -0500
Subject: Re: [asterisk-users] Music on Hold
On Wed, May 26, 2010 at 7:28 PM, Julien Claassen jul...@c-lab.de wrote:
Hi Motiejus!
If all else fails for the moment, it should be quite simple to move JACK.
Move all jack applications from /usr/local/bin to /usr/bin.
In /usr/local/lib move the dir jack and libjack* to /usr/lib.
That
Either teach your operators to press a new sequence that will send hold to
asterisk or reprogram your phone. I know how to do with Polycom phones, but
not linksys.
_
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of taimur
You would need to see if there is a hook flash hold.
Try playing with a hook/flash ( ie do a flash, wait, then hangup phone, it
may send the onhold message ) it may also ring back.
Or you will have to park the call
Hook flash , Dial 700 (if that's your park extension), hangup, then
On Wed, 26 May 2010 17:30:08 +0200, Vincent codecompl...@free.fr
wrote:
More information, as I investigate:
For those having the same issue, here's what I learned:
1. In /etc/modprobe.d/dahdi.blacklist.conf, blacklist the netjet
driver:
blacklist netjet
2. To configure Dahdi, edit
Hi Motiejus!
I of course menat prehacking the installation script. I can unpack my
jack-tarball or get a new svn and see, what I did. It was some time ago. And I
usually don't have to hack the installation prefix.
Information: I'm running my asterisk and JACK on a simple desktop system (no
On 5/26/2010 1:16 PM, Tim Nelson wrote:
- Jeff LaCoursierej...@jeff.net wrote:
On Wed, 26 May 2010, Brent Davidson wrote:
Just set the POTS lines to answer after a second ring rather than
after
the first. Problem solved.
Now that sounds like a good
hello,
which phone do you have behind the pap2 cause the hook flash time
sometimes could be set in the phone and then it will work with the pap2
also.
you should have a look at spaconfig.de (its a german website) but the
default parameters in sip and regional conf, may help you.
best regards
Hi Julien,
yes, I am thinking about implementing something better than asterisk,
but it is a future talk. It really lacks support for flexibility :S
I would be grateful if you found the hacked configure script and
sent it to me :-)
I did not really understand the part about C program, could you
Hi Guys,
is it possible that the silence joining with the Option q in a MeetMe room
damaged ?
I updated to Version 1.6.2.7 (before 1.6.2.6) and now my silence Orginates to
Play Voice into a Meetme Room will play a bleep after a Success Orginate
I Orginate with this simple AMI request.
Action:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Fri, May 21, 2010 at 23:35:14 -0300, Tim Nelson wrote:
Greetings!
Hi, Tim!
I had the opportunity to test a Sangoma A200 card and I have some
doubts that I would like to consult:
I tried to detect the card and I had no success using the
Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
On Sun, May 23, 2010 at 04:54:38AM -0400, cov...@ccs.covici.com wrote:
Hi. I am having problems with dahdi using kernel 2.6.32. I am using
2.6.30 kernel and it works fine -- here is the output of dahdi_cfg -vv
++ dahdi_cfg -vv
DAHDI Tools
Anyone can comment on this please?
Is it right to assume that if you own a PRI Caller ID always comes through
even if customer used *67 feature to block their CLID?
I understand that is true of calling a Toll-Free number.
Does Asterisk or LibPRI somewhere in the code abide by some standard to
Dear Supports,
I was attempting to install BRI Card(OpenVox B800P) with wcb4xxp in NT mode
.But I can not make it worked!
Could you please give me some hints? Thanks in advance!
Here are my environments:
--
CentOS-5.3
Kernel-2.6.18-164.el5
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