Hello all,
I found a problem regarding SIP presence in asterisk (1.6.2). The scenario is
not working properly for all users.
Our SIP client sends SIP:SUBSCRIBE to all the configured extensions in asterisk
during registration process. Asterisk replies with 200 OK for all SUBSCRIBE.
But if I run
2011/7/5 Nikhil 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 call ,A and B Call should connect back.
IMHO, blind tranfer
Hi all,
Trying to find where i got wrong in my config
Is the realm parameter in sip.conf only used for possible
autentication?
The thing is, i got my box more-or-less working as i wanted,
but i can only reach internal functions (like echo-test and so on) and
other sip-clients if i dial
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
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
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
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
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:
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
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 as
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
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
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
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
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
callid 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
On 07/05/2011 01:54 AM, Olivier wrote:
2011/7/5 Nikhil d.nik...@cem-solutions.net
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.
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
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
--
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
Thank you I'll watch. Support for Asterisk-Mysql is a bit minimal ... :-(
2011/7/1 Mickael MONSIEUR mickael.monsi...@gmail.com
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
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
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
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
On Wed, Jul 6, 2011 at 12:00 AM, Daniel - Asterisk earohua...@gmail.comwrote:
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
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
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
2011/7/6 Nikhil d.nik...@cem-solutions.net
**
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
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 -
On Wed, Jul 6, 2011 at 1:49 AM, Faisal Hanif fai...@vopium.com 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
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