On Mon, Jul 19, 2010 at 05:06:37PM -0400, Jose P. Espinal wrote:
Hello list,
I'm facing a little issue with dahdi attempting to load the OSLEC echo
canceller into my current kernel.
After compiling dahdi 2.3.0.1 with OSLEC support, I get the following
error when set 'oslec' as the
On Mon, Jul 19, 2010 at 09:19:55PM +0200, mattias wrote:
Ok
How to test on the cli
As i say
I running elastix and yes i know there a mailing list about elastix but the
people there only point me to the book about elastix
What version of asteris is it? What is the output of:
ls
Hi,
I'm trying to use Asterisk to place Automated Voice Calls.
A verbose log from Asterisk CLI taken when I place a call in the spool
directory looks like this:
-- Attempting call on SIP/MTN-NEW/my-number for application
MP3Player(/myfile) (Retry 1)
== Using SIP RTP CoS mark 5
In your sip.conf, there is no mention of your sip provider's IP address,
username and secret (password). Even if the provider doesn't have username
and secret requirements, there should at least be his IP address somewhere
in your sip.conf. Do they require registration? You should ask them what
Greetings list,
Whilst running through a routine check of some CDRs, I've noticed that the
originating channel's accountcode isn't preserved on creating a local
channel. For example, if we start with:
exten = 123,1,Set(CDR(accountcode)=foo)
exten = 123,n,Queue(bar,nrtw,,,)
And the queue 'bar'
Hello,
Gigaset C470IP and others offer auto-configuration through an
auto-configuration code.
(see manual here
http://gigaset.com/hq/en/cms/PageCustomerServicesDownloadsManuals.html)
Has someone managed to get more information about it ?
Is this feature available for enterprises or is it
Hi,
I can configure multiple source IP addresses on a Ethernet interface.
Is it possible to configure asterisk to bind to a different source IP
address for each peer?
Thank you,
AC
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Hi,
On Tue, Jul 20, 2010 at 2:14 PM, Olivier oza_4...@yahoo.fr wrote:
Hello,
Gigaset C470IP and others offer auto-configuration through an
auto-configuration code.
(see manual here
http://gigaset.com/hq/en/cms/PageCustomerServicesDownloadsManuals.html)
Has someone managed to get more
Hi,
I am running Fedora 7 VM. On an earlier configuration with zaptel and
Asterisk 1.4.21 , meetme worked alright. I upgraded to Dahdi and Asterisk
1.4.26, and the result is choppy sound via Meeme, while a simple Musiconhold
works OK with descent audio quality. So I am sure its a Dahdi_dummy
On Tue, Jul 20, 2010 at 07:12:36PM +0530, Mr architect wrote:
Hi,
I am running Fedora 7 VM. On an earlier configuration with zaptel and
Asterisk 1.4.21 , meetme worked alright. I upgraded to Dahdi and Asterisk
1.4.26, and the result is choppy sound via Meeme, while a simple Musiconhold
works
Hi Tzafrir,
Maybe I did not understand very well your answer, but just in case,
doing a 'locate dahdi_echocan*' on the Asterisk box, gives me the
following output:
...
r...@slackbox:~# locate dahdi_echocan*
/lib/modules/2.6.29.6-smp/dahdi/dahdi_echocan_kb1.ko
Linux version 2.6.21-1.3194.fc7 (
kojibuil...@xenbuilder4.fedora.phx.redhat.com) (gcc version 4.1.2 20070502
(Red Hat 4.1.2-12)) #1 SMP Wed May 23 22:35:01 EDT 2007
Dahdi-linux-2.2.02
Hope this helps..
On Tue, Jul 20, 2010 at 7:32 PM, Tzafrir Cohen tzafrir.co...@xorcom.comwrote:
On Tue, Jul
Hi,
I set my list to subscribe to digest and I can't see how to reply to
your reply without starting a new thread.
There is no need for SIP username and password because the provider
authenticates me on my IP address.
I thought that host=192.168.34.1 would be the sip provider IP address.
Hello Jose.
I've found the same problem on some servers and I solved it renaming (or
deleting) the echo.ko driver already present in the binary kernel
distribution:
In my system is something like:
/lib/modules/2.6.27.45-0.1-default/kernel/drivers/staging/echo/echo.ko
Hope this helps you.
Best
sorry for the typo mistake. the actual dial string that I used is like this
Dial(SIP/x...@192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/x...@192.168.0.12:64290-0966ab80,30,rtT)
it is not
Dial(SIP/192.168.0.20:5062-096afee8,30,rtT)
Dial(SIP/192.168.0.12:64290-0966ab80,30,rtT)
it was just a
On Tue, Jul 20, 2010 at 07:45:56PM +0530, Mr architect wrote:
Linux version 2.6.21-1.3194.fc7 (
Any chance you could try something newer?
kojibuil...@xenbuilder4.fedora.phx.redhat.com) (gcc version 4.1.2 20070502
(Red Hat 4.1.2-12)) #1 SMP Wed May 23 22:35:01 EDT 2007
Dahdi-linux-2.2.02
This host=192.168.34.1 is where you'll put your provider's IP address.
Currently you are using some local address which is not your provider's IP
address. Where did you get it from? Call your providrr and ask them the IP
address of the server where you'll be sending your calls.
Zeeshan A Zakaria
Nobody uses chan_local
2010/7/16 Mickael Monsieur mickael.monsi...@gmail.com
Hello
I just coding a AGI script for billing.
- For external calls, I pass the call directly on a trunk. I do :
Dial(trunk1/extension) - OK !
- For internal calls (shortcode, others users ...) I am
As the auto provision codes are provided by a server run by Gigaset I
suspect that they are only making them accessible to ITSPs. I know that
Tony Stankus, product manager at Gigaset US, helped get SIPGate US
established in there. He seemed open to helping any bone fide service
provider get on the
Greetings list,
I've compiled and installed dahdi countless times on standalone machines,
but recently I've been trying to compile Dahdi in a Xen DomU without much
success. The errors I'm seeing are as follows:
Hi,
No that is the correct address. I know it is an internal IP.
We have our machine hosted in racks at our SIP providers data center.
They've patched a new port to our cabinet and linked that to a gateway
(172.28.20.105).
As long as we use that gateway (and the IP address they assigned to
Hi to all, I have a strange behavior in my asterisk server.
I have a queue for 5 agents, the calls enter the queue an go to the agents
normally, but if I need to transfer or dial directly to an agent extension
that is already in a call, the pbx hung up the actual call (not the
transferred
If you add qualify=yes to the setting in sip.conf it will send a sip
message to the peer every 60 seconds to check if it is alive.
If you try to make a call while the peer is not alive it will fail
immediatly rather than the caller hearing silence while your box waits
for a reply timeout.
Andy
sorry for typo mistake in my last post. as from my orignal post two
registration of the same user are as follows
SIP/x...@119.68.0.90:5060
SIP/x...@202.16.34.10:5678
so dial command with unique-id i want to use will be
Dial(SIP/x...@192.168.0.20:5062-096afee8,30,rtT)
Hi,
I am trying to write the regserver value into my database using ARA but the
field keeps empty.
Afaik all that needs to be done to make it work is having a db field called
regserver, the var systemname set in asterisk.conf and
rtsavesysname=yes in sip.conf.
But the regserver is not
For a quick and dirty view, from your asterisk box, do:
tcpdump host 192.168.34.1
and make a test call. For a pcap file you can read with wireshark,
instead do
tcpdump host 192.168.34.1 -s1500 -w FILENAME.pcap
where FILENAME is whatever you think is meaningful. This will show you
what's
Hi!
Nobody uses chan_local
Absolutely nobody. Except you. ;-
Maybe this will help you: Search for Asterisk timing, consider to not
run Asterisk in a virtual environment, and do not run X on the same box.
Makre sure to turn off silence suppression in your SIP client(s).
Search for
Hi,
Thanks, I added that. I'll ask my network provider if they received
these message tomorrow morning. That will narrow things down to either
an Asterisk configuration or a network routing issue.
There is not really a caller, I'm trying to use Asterisk as an Automated
Voice Message
Asterisk runs fine in a Virtual environment; it is (some) functions that
depend on real timing that may (will) give you fits.
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Ah genius :) I had tried tcpdump but kept getting a permission denied
error. When you suggested it I remembered to set AppArmor to complain
and so now I have a dump of my traffic. Thanks! Wireshark is
illuminating, I think this is a routing error.
On 20/07/2010 05:52 PM, tdensmore
Posting a sip debug will probably be helpfull aswell as you can see
exactly where the traffic is being sent and what the response was.
Andy Beak wrote:
Hi,
Thanks, I added that. I'll ask my network provider if they received
these message tomorrow morning. That will narrow things down to
You are getting congestion error message, which in your case only means
failed sip communication, or no sip communication at all. Settings on your
end are just fine.
Can you post the Dial command from your extensions.conf?
Zeeshan A Zakaria
--
www.ilovetovoip.com
On 2010-07-20 12:16 PM, Andy
On Tue, Jul 20, 2010 at 04:35:59PM +0100, Chris Bagnall wrote:
Greetings list,
I've compiled and installed dahdi countless times on standalone machines,
but recently I've been trying to compile Dahdi in a Xen DomU without much
success. The errors I'm seeing are as follows:
while setting up accountcode value try two underscores just before variable
name like '__accountcode'
for more google for Inheritance of Channel Variables
On Tue, Jul 20, 2010 at 5:05 PM, Chris Bagnall
aster...@lists.minotaur.ccwrote:
Greetings list,
Whilst running through a routine check
As per my knowledge Asterisk uses system network settings to pick
appropriate source address so you can not do it within Asterisk until you
configure Linux firewall / routes accordingly.
Please correct me if I am wrong!
On Tue, Jul 20, 2010 at 5:19 PM, AC a57m...@gmail.com wrote:
Hi,
I can
Good afternoon list.
I'm experiencing a problem with my SIP channel's. When I have an external
connection for one of my SIP carrier's, I can listen to the client and the
client listens to me normally. The problem is when I will transfer this
connection, the call is mute for the extension I have
On Tuesday 20 July 2010 12:37:30 Nasir Iqbal wrote:
while setting up accountcode value try two underscores just before variable
name like '__accountcode'
You cannot do that with CDR variables at this time, only channel variables.
--
Tilghman Lesher
Digium, Inc. | Senior Software Developer
Hi!
client listens to me normally. The problem is when I will transfer this
connection, the call is mute for the extension I have transfered. Only the
client hears normally.
I *think* there is/was an entry in the bug tracker on this. You might
want to search https://issues.asterisk.org (also
Hi Signorini,
I looked for the 'echo.ko' file and is not present but
the file 'dahdi_echocan_oslec' is.
At compile time, I see this:
...
WARNING: oslec_create
[/root/dahdi_linux-SlackBuild/dahdi-linux-2.3.0.1/drivers/dahdi/dahdi_echocan_oslec.ko]
undefined!
WARNING: oslec_free
This is the exit of core show version:
Asterisk 1.6.0.28 built by root @ AST on a i686 running Linux on 2010-06-28
12:21:24 UTC
Obg,
Rodrigo Lang.
2010/7/20 Philipp von Klitzing klitz...@pool.informatik.rwth-aachen.de
Hi!
client listens to me normally. The problem is when I will transfer
Rodrigo Lang schrieb:
Good afternoon list.
I'm experiencing a problem with my SIP channel's. When I have an
external connection for one of my SIP carrier's, I can listen to the
client and the client listens to me normally. The problem is when I
will transfer this connection, the call is
Hi Gareth,
Thanks for replying. Here is the SIP debug from the CLI. I assume that
the first two blocks are from having qualify=yes and the remaining are
from attempting to place a call.
Do you know what SIP/2.0 480 No Routes Found means? It looks like the
SIP provider cannot find my box.
Dear all;
I have and asterisk box receive a phone call from a VoIP carrier and
pass it to internal SIP clients,
it worked fine when using g711, when it comes to g729 call established
successfully and there is some rtp
flows but dead air on both side, any ideas?
Regards
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