Thanks for all the replies.
So it seems auto-configuration code is a feature for ITSP, not for system
integrators looking for an easier way to configure each DECT base.
Too bad, as I'm sure this auto-configuration feature relies on standard
protocols we could play with (DHCP, TFTP, HTTP, ...).
Re
Hello,
I have been facing an issue that voice is getting distorted sometimes in MOH
(MusicOnHold) application.
I have checked and confirmed that lame is properly installed, even tried
native formats (ulaw, alaw, gsm), but the randomly seen distortion in MOH
can't be eliminated.
I came to know abou
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
--
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
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
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
> Hi!
>
> > client listens to me normally. The problem is when I will transfer this
> > connection, the call is m
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"
[/root/dah
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 (al
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
tw
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 tr
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 wrote:
> Hi,
>
> I can configure multip
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
wrote:
> Greetings list,
>
> Whilst running through a routine check of some CDRs, I've noti
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:
>
> /var/tm
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 B
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
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 wrot
"Asterisk" runs fine in a Virtual environment; it is (some) functions that
depend on "real" timing that may (will) give you fits.
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join u
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 server
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 "c
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 be
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 g
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)
Dial(SIP/x...@192.168.0.12:
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
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 call)
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
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:
/var/tmp/portage/net-misc/dahdi-2.3.0.1/work/dahdi-linux-2.3.0.1/drivers/dahdi/zaphf
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
Nobody uses chan_local
2010/7/16 Mickael Monsieur
> 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
>Dial(Local/exten
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
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.
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 typing
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 re
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.
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 wrote:
> On Tue, Jul 20, 2010 at 07:12:36PM
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
/lib/modules/2.6.29.6-smp/dahdi/dah
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
> wo
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 proble
Hi,
On Tue, Jul 20, 2010 at 2:14 PM, Olivier 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 information a
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
--
_
-- Bandwidth and Colocation Provided by h
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 dedicate
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
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 sip
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
> Chann
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