On Wed, Jul 6, 2011 at 1:49 AM, Faisal Hanif wrote:
> You have to provide channel ID to command like “channel request hangup
> SIP/12316156-sad4d46a5”.
>
> **
>
Thanks, but "all" is also a valid keyword according to the documentation. I
think there are some bugs associated with hung channels
You have to provide channel ID to command like channel request hangup
SIP/12316156-sad4d46a5.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of A E [Gmail]
Sent: Wednesday, July 06, 2011 9:50 AM
To: Asterisk Users Mailing List - Non-Co
2011/7/6 Nikhil
> **
> Hi
> Below is the comment that written in chan_sip.c(handle_request_refer)
> file of asterisk .In RFC also mentioned that if blind transfer failed call
> should connect back, some of phones support this(If received refer) like
> cisco,polycom and etc.
>
> \par Blind tr
If the problem always related to some specific module then try clean
recompiling asterisk if it is with random modules then check you system RAM.
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Agustina
Berretta
Sent: Wednesday, July 0
Hi
Below is the comment that written in
chan_sip.c(handle_request_refer) file of asterisk .In RFC also mentioned
that if blind transfer failed call should connect back, some of phones
support this(If received refer) like cisco,polycom and etc.
\par Blind transfers
The transferor p
On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk wrote:
> On the CLI write: sip show channels
>
> If there are lots of bye channels you have the same problem than me.
> I've tried waiting with the call generator -sipp- and channels
> finished when there are a few. But they're not ending faster e
On the CLI write: sip show channels
If there are lots of bye channels you have the same problem than me.
I've tried waiting with the call generator -sipp- and channels
finished when there are a few. But they're not ending faster enough
when I send lots of concurrent calls.
Elder
2011/7/5, A E [G
hello people,
I am running v1.8.4.2 on debian squeeze on a sparc platform...and for some
reason I have noticed that only after a few test calls, the asterisk process
is running between 95% - 99.9% CPU when there's absolutely nothing on the
system. This is a clean Asterisk system in an internal net
Hi folks!
I´m having the following problem:
I get the following messages, asterisk get automatically reloaded and agents
log out once or twice a day, randomly.
[Jul 4 11:36:25] VERBOSE[30004] app_queue.c: -- Couldn't call Agent/2002
[Jul 4 11:36:29] VERBOSE[30320] logger.c: Asterisk Event Logger
Thank you I'll watch. Support for Asterisk-Mysql is a bit minimal ... :-(
2011/7/1 Mickael MONSIEUR
> Hello,
> I just implement the SIP Peers with MySQL.
>
> In the structure mySQL missing the following fields:
>
> nat = yes
> notransfer = yes
> dtmfmode = rfc2833
> call-limit = 2
> canreinvite
Hi,
Using Polycom's Master configuration file, I could not find any convenient
way to store 2 different versions of the same localization file on the same
TFTP server.
Did I miss something ?
What I would like is to have both files under TFTP root
versionA/SoundPointIPLocalization/French_France/So
On Tue, 2011-07-05 at 12:33 +0500, Faisal Hanif wrote:
> The problem you are reporting is not related to realm but can be context or
> domain.
>
Tnx,
It was indeed a domain issue.
In some cases static definitions in /etc/hosts is not a good replacement
for DNS...
hw
--
__
The argument to chanspy is a pattern and not an exact match.
--
Jim Dickenson
mailto:dickenson at cfmc.com
CfMC
http://www.cfmc.com/
On Jul 2, 2011, at 3:48 PM, steve casto wrote:
> asterisk 1.4.32 have zapata.conf soft link to chan_dahdi.conf to use
> flash operator panel < 2.0
>
> (from ext
On 07/05/2011 01:54 AM, Olivier wrote:
2011/7/5 Nikhil mailto:d.nik...@cem-solutions.net>>
Hi all
In asterisk if blind transfer failed ,call is not connecting back .
For Eg:
A make call to B through asterisk,then B transfer the call to C.
If C did not answer the ca
Hi all, can someone explain what siphistory is supposed to do as it
appears to record nothing to my log files. When I sip show history
it's fine but it's not logging anything. My logger.conf has
debug => debug and the debug file grows. Is my understanding correct in
that at the end of the call
Hi,
we have 4 asterisk, versions are 1.4.35 1.4.36 1.6.2.18 and 1.4.42 One
GrandStream GXV3000 is used for the tests. He is registered to asterisk
1.6.2.18 asa well as 1.4.35. Calling echo test is OK on both servers,
get audio and video. Calling echo test from asterisk 1.4.36 bye a SIP
trunk
On 11-07-05 10:07 AM, Tobias Steen wrote:
It seems that the full distro package from FreePBX with Asterisk 1.8.1.4
someway hides (deletes?) the source directory for asterisk after
installation.
I cant find the directory under /usr/src/
I am trying to compile and install the conference module "a
It seems that the full distro package from FreePBX with Asterisk 1.8.1.4
someway hides (deletes?) the source directory for asterisk after
installation.
I cant find the directory under /usr/src/
I am trying to compile and install the conference module "app_konference"
and need to point a var
Hello is use realtime sip-peers in 1.8, and have the problem, that when a peer
is loaded from database, the devstate is "AST_DEVICE_UNAVAILABLE" and
the the peers
can not be called from the queue. because the app_queue only calls
agens in state
AST_DEVICE_NOT_INUSE or AST_DEVICE_UNKNOWN.
My questi
Hi,
One of my college "Gohar Ahmed" suggested an intelligent solution to your
problem. I am coping his words below,
Create SIP trunks and create a queue [distributor] and register trunks in it
as static agents with strategy "rrmemory" ,
To keep track of number of calls served per trunk as well a
Hi All,
Following message I got in console for an extension,
[Jul 5 12:10:56] VERBOSE[3729] chan_sip.c:
<--- SIP read from UDP:132.186.230.70:7510 --->
SUBSCRIBE sip:18...@sip1.test.in SIP/2.0^M
Via: SIP/2.0/UDP
132.186.230.70:7510;branch=z9hG4bK-d8754z-2b3b65532b3b6553-1---d8754z-;rport^M
Max-
Everything was fine but i have a hangup in an other shorter route where
the extenension was 1234, but i forgot the '4' so everytime it goes to
the defined hangup in the other route.
Thx for your quick answer and sorry for my mistake
On 05.07.2011 11:34, Thorsten Göllner wrote:
Executin
Executing the query in MySQL-CLI is fine?
Am 05.07.2011 11:25, schrieb Ulrich Meckel:
Hi List
I tried to use SQL Query in my diaplan. If i only use one or two there
is no Problem but if i try to start the third one after the other it
hangup after the 2nd clear
exten => _123.,1,MYSQL(Connect
Hi List
I tried to use SQL Query in my diaplan. If i only use one or two there
is no Problem but if i try to start the third one after the other it
hangup after the 2nd clear
exten => _123.,1,MYSQL(Connect connid host user pw db_name)
exten => _123.,n,MYSQL(Query resultid ${connid} SELECT v
The problem you are reporting is not related to realm but can be context or
domain.
-Original Message-
From: asterisk-users-boun...@lists.digium.com
[mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Hans Witvliet
Sent: Tuesday, July 05, 2011 11:59 AM
To: Asterisk Users Mailing
Yes, i have done this already. Though there is no possibility of sending unique
id or just recording answered calls with the oreka GPL version. This is where
the xorcom asterisk patch comes in handy, because you can set it to start
sending the trp data when a call gets into the queue.
/ Marcus
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