Hi all,
How can I get the RTP port one SIP client is using for sending/receiving RTP
flow? Can I obtain it in from SIP_HEADER of something like that in the dialplan?
Thank you!
Isabel
Este mensaje se dirige exclusivamente a su destinatario. Puede consultar
thank you for your response after module unload app_queue.so
module load app_queue.so i can do this operation, for the internal
extension
now i have another issue related to the same queues
i have 2 providers
for the first provider
exten = 800,1,AgentLogin()
exten = 52046,1,Answer()
You can use tcpdump portrange 1-2 udp
2011/10/20 Andrew Higgs andrew.m.hi...@gmail.com
Hi Isabel,
Could you not just filter out after the fact using something like
Wireshark?
Regards
On Thu, Oct 20, 2011 at 3:28 PM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Dear all,
** **
Hi all,
Is there a way to read in the dialplan a macro output parameter?
For instance, in the following macro I would like to know the pid of the Linux
process for killing it when hanging up.
[macro-capture]
exten = s,1,NoOp(Caller IP = ${ARG1})
exten = s,n,NoOp(Filename = ${ARG2})
exten =
Hi,
Any video softphone that will send the video codec in the first INVITE, in
eyebem and some other phones like Ekiga first we are getting audio and then
there is a button SEND VIDEO, if we click that the re-invite is going with
video codec, whereas i need to send the video at first invite
Hi,
I tried to establish NCS call b/w 2 endpoints of same PacketCable MTA
using asterisk-10.0.0.
I observed that MDCX sent to aaln/1 contains its own SDP. Some I
observed with aaln/2.
So voice path is not established b/w aaln/1 and aaln/2.
My Configurations:
mgcp.cong:
Do you need to know to get it in dialplan? If I not, from shell (not
Asterisk CLI) I usually use:
netstata -a | grep asterisk
By default Asterisk settings it should be something between 10k-20k
-Bruce
On Fri, Oct 21, 2011 at 3:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How
Hello,
is every development on chan_skype out of the question after Skypcrosoft pulled
the plug, or can we hope for an Asterisk 10 Version that supports the new,
shiny messaging-api in asterisk 10?
Andreas...
--
On 11-10-21 11:45 AM, Andreas Anderson wrote:
Hello,
is every development on chan_skype out of the question after Skypcrosoft pulled
the plug, or can we hope for an Asterisk 10 Version that supports the new,
shiny messaging-api in asterisk 10?
Andreas...
On 10/20/2011 05:59 AM, JR Richardson wrote:
Hello Everyone,
The documentation suggests using unixodbc for asterisk realtime. Is
there any way
we can just use native database clients such as libmysqlclient from
MySQL? The native
clients tend to be more up-to-date.
Thanks in Advance,
On Fri, Oct 21, 2011 at 4:50 AM, salaheddine elharit
salah.elharit...@gmail.com wrote:
-- Accepting call from '522343535' to '560' on channel 1/14, span 1
-- Executing [560@default:1] Answer(Zap/14-1, ) in new stack
-- Executing [560@default:2] Queue(Zap/14-1, hotline) in new stack
On Fri, Oct 21, 2011 at 6:54 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
** **
Is there a way to read in the dialplan a macro output parameter?
For instance, in the following macro I would like to know the pid of the
Linux process for killing it when hanging up.
I
Hi Paul,
is every development on chan_skype out of the question after
Skypcrosoft pulled the plug, or can we hope for an Asterisk 10 Version
that supports the new, shiny messaging-api in asterisk 10?
Nope, nobody submitted any patches for it. So anything now
On Fri, Oct 21, 2011 at 2:46 AM, ISABEL ORDAS ARNAL i...@tid.es wrote:
Hi all,
How can I get the RTP port one SIP client is using for sending/receiving
RTP flow? Can I obtain it in from SIP_HEADER of something like that in the
dialplan?
Thank you!
** **
I don't think you
yes if i chang it from queues or meetme to dial there is no issue it'works
withou issue
i call the same numbers 560 and i can reponse this call in sip 1000 without
issue
exten = 560,1,Dial(SIP/1000, 30)
the asterisk version is
Asterisk 1.4-r110474M
zaptel-1.4.12.1
i want to know also why for
Since when can someone submit a patch for chan_skype?? Did i miss an
announcement that it has been opensourced? I'm under the impression
that digium is the only party who *can* extend chan_skype...
Paul was a little confused and thought something would have to be added to
Asterisk. But, with
On Fri, Oct 21, 2011 at 12:24 PM, salaheddine elharit
salah.elharit...@gmail.com wrote:
yes if i chang it from queues or meetme to dial there is no issue it'works
withou issue
Please do the call again, this time please show us the output also with a
sip debug and a zap debug.
the
On 11-10-21 01:25 PM, Terry Wilson wrote:
Since when can someone submit a patch for chan_skype?? Did i miss an
announcement that it has been opensourced? I'm under the impression
that digium is the only party who *can* extend chan_skype...
Paul was a little confused and thought something would
Moved to Asterisk 1.8.7, most of the watnings/errors are gone. I have a
new error though:
[Oct 21 13:40:40] ERROR[15709] ais/clm.c: Could not initialize cluster
membership service: Try Again
And I get a warning that no music on hold classes are configured.
Never mind. The chan_dahdi warnings
The Asterisk Development Team is pleased to announce the release of
DAHDI-Linux 2.5.0.2 and DAHDI-Tools 2.5.0.2.
2.5.0.2 is a bug fix release. It is recommended that current users of v2.5 to
upgrade.
DAHDI-Linux 2.5.0.2, DAHDI-Tools 2.5.0.2, and DAHDI-Linux-Complete
2.5.0.1+2.5.0.1 are available
20 matches
Mail list logo
