On 10/05/2011 11:21 PM, A. M. Hoffmeister wrote:
Am 05.10.2011 20:42, schrieb Marek Cervenka:
hello,
is there some way to notify people in the same pickup group about call
from caller to callee?
i.e. i have call from 111 to 222
there are 222,333,444 in the same pickup group
333,444 see
Search for dialog-info pickup
-Original Message-
From: Marek Cervenka cerv...@fpf.slu.cz
Sender: asterisk-users-boun...@lists.digium.com
Date: Fri, 07 Oct 2011 09:47:45
To: Asterisk Users Mailing List - Non-Commercial
Discussionasterisk-users@lists.digium.com
Reply-To: Asterisk Users
Hi
Since upgrading to 1.8.4.3 my callers no longer hear busy tone when I
use playtones().
Here is the CLI output on such a case
http://pastebin.com/TMBFhngh
Any ideas anyone?
Regards
Jon
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-- Bandwidth and Colocation
Hi,
I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken
from deb http://packages.asterisk.org/deb lucid main) including dahdi
from this same repository. No FFA involved.
On incoming calls (only SIP, no telephony card), fax detection is
working but reception failed with
Hi,
my asterisk 1.8.7 is working well with phones (SNOM, Gigaset 620 and
GrandStream) connected from the lan
I now want to connect a snom320 from outside but it failed, having always
[Oct 7 14:48:04] ERROR[3870]: netsock2.c:94 ast_sockaddr_stringify_fmt:
getnameinfo(): ai_family not
Hi,
I am getting this error message while executing the module load
chan_dahdi.so.
astrisks*CLI module load chan_dahdi.so
Unable to load module chan_dahdi.so
Command 'module load chan_dahdi.so' failed.
== Parsing '/etc/asterisk/chan_dahdi.conf': == Found
== Parsing
It is likely you have an error in your /etc/asterisk/chan_dahdi.conf
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of michael k
Sent: Friday, October 07, 2011 9:24 AM
To: Asterisk Users Mailing List -
Hi,
This is my /etc/asterisk/chan_dahdi.conf file.
[root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf
; Copied from DAHDI Module of FreePBX
[general]
#include chan_dahdi_general.conf
[channels]
; include dahdi groups defined by DAHDI module of FreePBX
#include
remove the c argument
Kristijan
2011/10/7 Administrator TOOTAI ad...@tootai.net:
Hi,
I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken from
deb http://packages.asterisk.org/deb lucid main) including dahdi from this
same repository. No FFA involved.
On incoming calls
Hello!
I want to thank everyone who helped me out with tips for load balancing
asterisk machines in a cluster.
I have encountered a new problem that is related to SIP diversion headers in
the INVITE.
I make calls through the manager interface and now want to add a
SIP-Diversion header that
Try run your outbound leg through a Local channel.
--
This message was painstakingly thumbed out on my mobile, so apologies for
brevity, errors, and general sloppiness.
Alex Balashov - Principal
Evariste Systems LLC
260 Peachtree Street NW
Suite 2200
Atlanta, GA 30303
Tel: +1-678-954-0670
Fax:
You can dial a local channel which executes a dial plan that does what you want.
Channel: Local/dial_number@cfmc_cdi_private
This will use exten dial_number in the cfmc_cdi_private context.
If you add something like this to the originate packet
Variable: CfMC_Use_CID=5419712513
You can use
Can anyone suggest an ITSP with Singapore DIDs and local Singapore
termination?
Cheers,
j
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New to Asterisk? Join us for a live introductory webinar
Please paste the configurations in the #included files as well.
On Fri, Oct 7, 2011 at 7:07 PM, michael k mich...@inapp.com wrote:
Hi,
This is my /etc/asterisk/chan_dahdi.conf file.
[root@astrisks asterisk]# cat /etc/asterisk/chan_dahdi.conf
; Copied from DAHDI Module of FreePBX
On 10/07/2011 07:46 AM, Administrator TOOTAI wrote:
Hi,
I setup my first stock 1.8.7 asterisk (Ubuntu LTS 10.04 packages taken
from deb http://packages.asterisk.org/deb lucid main) including dahdi
from this same repository. No FFA involved.
On incoming calls (only SIP, no telephony card), fax
On Thu, Oct 6, 2011 at 11:25 AM, Kyle Sexton k...@mocker.org wrote:
I'm looking at the Cisco AS5400XM to convert some incoming T1s to SIP
signaling. Has anyone had any experience with these devices? The
feature cards that Cisco sells can be a little confusing. I'm
thinking something like
On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use sip set
debug on, rtp set debug on and udptl set debug on
No NAT involved and I shut off IPTables. Still no luck. Debug shows
the SIP invite, RTP
On 10/07/2011 02:20 PM, James Sharp wrote:
On 10/07/2011 12:27 AM, Nasir Iqbal wrote:
Check firewall and NAT settings!
Monitoring sip and media flow from asterisk cli can help, use sip set
debug on, rtp set debug on and udptl set debug on
No NAT involved and I shut off IPTables. Still no
for which user/number sip reinvite is for? ooh! you are running a direct
application without any dialplan or user, may be that is the cause! I think
you should first write fax dialplan, reload asterisk and test again with
originate but this time with extension not application.
Nasir Iqbal
ICT
On 10/07/2011 03:06 PM, Nasir Iqbal wrote:
for which user/number sip reinvite is for? ooh! you are running a direct
application without any dialplan or user, may be that is the cause! I
think you should first write fax dialplan, reload asterisk and test
again with originate but this time with
On 10/07/2011 04:04 PM, Kevin P. Fleming wrote:
First, we can see that Gafachi's T.38 implementation still has some
breakage in it (although the problems are ones that Asterisk has been
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has
On 10/07/2011 03:29 PM, James Sharp wrote:
On 10/07/2011 04:04 PM, Kevin P. Fleming wrote:
First, we can see that Gafachi's T.38 implementation still has some
breakage in it (although the problems are ones that Asterisk has been
taught to deal with). In its 200 OK to the T.38 re-INVITE, it has
On 10/07/2011 04:42 PM, Kevin P. Fleming wrote:
You shouldn't be *receiving* CNG, as you are the calling endpoint.
You're right. Hadn't even thought about that.
If you are seeing UDPTL packets containing T.38 CED, V.21 preamble, DIS,
etc. then something is badly wrong.
... and, that
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