I am not sure I understand the required routing pattern, but I'm sure
queues are your friends, as you can dynamically add and remove member and
you can have a first-level queue easily move fall-through to another queue
in case all members should be busy or none should be available. Plus by
using
On Wed, 2013-04-10 at 11:06 -0700, Carlos Alvarez wrote:
On Wed, Apr 10, 2013 at 11:02 AM, Steve Edwards
asterisk@sedwards.com wrote:
dumpcap can capture all of the SIP (and RTP) packets into a
series of files without a huge performance hit.
Hi,
I have the following setup:
Ubuntu 12.04.02 LTS (64 bit)
Asterisk 11.2.1
Sangoma 4-Port-Card (A104d) with firmware 43 (german e1-ports connected)
WANPIPE Release: 3.5.28
DAHDI Version: 2.6.1 Echo Canceller: HWEC
libpri version: 1.4.12
I call via sip into the dialplan. Then I do a
CLIchannel request hangup DAHDI/1-1
Would work.
But 'dahdi destroy channel 1' shouldn't segfault asterisk.
Alec
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of
Thorsten Göllner
Sent: Thursday, 11 April
hi,
strange behaviour while trying to use pri debugging on asterisk 11.x ...
please take a look:
bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
*pri intense debug span*no description available
Hi,
I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and
would say it is a bug...
To remotely hang up a call use
*
**hangup request channel*
where channel is the exact id of your channel as you would receive it via
*core show channels*
yves
Am 11.04.2013 10:56, schrieb
hello all
i,m newbie in asterisk and now want to sip and h323 connection.
this is my scenario:
phone(ext100)---freepbx---sip---system1---H323---system2---freepbx---phone(ext200)
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service
On Thursday 11 April 2013, s m wrote:
when i call 100 from 200, every thing is ok and phone is ringing but
when i call 200 from 100, it says service unavailable.
i debug asterisk in my system 2 and see below message:
Dropping call because extensions '200', 's' and 'i' doesn't exists
in
Thanks! I do not have experience with bug reporting. Is that neccessary
in that case? Where can I open a ticket for it (if neccessary)?
Am 11.04.2013 12:23, schrieb Yves A.:
Hi,
I can reproduce your report (11.0.1, libpri 1.4.13, dahdi 2.6.1) and
would say it is a bug...
To remotely hang up
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
[to-232]
host=192.168.0.232
type=peer
qualify=yes
and 192.168.0.232 in the ip address of my freepbx.
On 4/11/13, A J Stiles asterisk_l...@earthshod.co.uk
hi,
try
exten= _2.,1,Dial(SIP/to-232/2${EXTEN:1})
Note space before underscore.
On Thu, Apr 11, 2013 at 2:50 PM, s m sam.gh1...@gmail.com wrote:
this is my [from-trunk] extension:
[from-trunk]
exten=_2.,1,Dial(SIP/to-232/2${EXTEN:1})
and this is [to-231] in sip_additional.conf:
- Original Message -
CLIchannel request hangup DAHDI/1-1
Would work.
But 'dahdi destroy channel 1' shouldn't segfault asterisk.
The dahdi destroy channel command is *only* for use when you know
what your doing. Even then I would not recommend ever using that
command. The CLI help
hi,
you have not assign any value to CDR(userfield).
try
code = #111,self,SET(CDR(userfield)=111)
On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez cur...@telecomabmex.comwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I am trying to set the CDR(userfield) to a certain vaule
Hi,
I have a problem with forwarding a voicemail and prepending a message to it.
If a user just forwards a voicemail, everything works fine.
However, if a user prepends a message to the voicemail when forwarding, the
voicemail that is forwarded only contains the prepended message and not the
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 4/11/13 11:18 AM, Asghar Mohammad wrote:
hi, you have not assign any value to CDR(userfield). try code =
#111,self,SET(CDR(userfield)=111)
On Thu, Apr 11, 2013 at 12:53 AM, Carlos Chavez
cur...@telecomabmex.com
hi,
it is not difficult in php and mysql i have created a simple billing system
for my wholesale postpay clients without any AGI.
it report ACD ASR all calls ANSWERD calls filter by date by callerid etc.
do billing as soon as call end.
for billing i am using mysql trigger.
report live calls.
2
- Original Message -
hi,
strange behaviour while trying to use pri debugging on asterisk 11.x
...
please take a look:
bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show version
DAHDI Version: 2.6.1 Echo Canceller: HWEC
bas1104*CLI help pri
pri
i am using
exten = _XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)})
cli_name is field in mysql and it work fine.
show me cli output without AGI.
On Thu, Apr 11, 2013 at 6:41 PM, Carlos Chavez cur...@telecomabmex.comwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On 4/11/13 11:18 AM,
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
When I execute without using the AGI method I get no output on the CLI
at all.
On 4/11/13 11:54 AM, Asghar Mohammad wrote:
i am using exten =
_XXX.,n,Set(CDR(cli_name)=${CHANNEL(peerip)}) cli_name is field
in mysql and it work fine. show me
how you are executing?
show me your full context and how call enter in context.
On Thu, Apr 11, 2013 at 7:07 PM, Carlos Chavez cur...@telecomabmex.comwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
When I execute without using the AGI method I get no output on the CLI
at all.
On
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Right now it is a simple call between 2 extensions. The receiving
extension dials the code. The 3rd line of my h extension is a
Noop(${CRD(userfield)})
pbxoficina*CLI features reload
== Parsing '/etc/asterisk/features.conf': == Found
==
you should set variable in extensions.conf not in features.conf
On Thu, Apr 11, 2013 at 7:34 PM, Carlos Chavez cur...@telecomabmex.comwrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Right now it is a simple call between 2 extensions. The receiving
extension dials the code. The 3rd
thanks, that command syntax works.
yves
Am 11.04.2013 18:51, schrieb Richard Mudgett:
- Original Message -
hi,
strange behaviour while trying to use pri debugging on asterisk 11.x
...
please take a look:
bas1104*CLI pri show version
libpri version: 1.4.13
bas1104*CLI dahdi show
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