On Wed, 2010-05-12 at 22:10 -0700, Steve Edwards wrote:
On Thu, 13 May 2010, Pham Quy wrote:
Hi all,
I use asterisk-1.6.2.7 and asterisk addon version 1.6.2.1.
It started ok with out cdr_addon_mysql.o. But when I put
cdr_addon_mysql.o in to modules folder, it fail at start and the
Issue solved.
Looks like all I was missing was one parameter:
fromuser=
Thanks for your time!
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Hi,
I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
One trunk is SIP and the other IAX2.
Normally, I use IAX2 but have noticed easily reproducible audio quality
problems (voice in/out is OK but there's a
Show the details on the active channels when using both methods and
check what codecs are being used.
Vieri wrote:
Hi,
I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
One trunk is SIP and the other
On Thu, May 13, 2010 at 4:17 AM, Vieri rentor...@yahoo.com wrote:
Hi,
I have an audio quality problem regarding IAX2. I have 2 Asterisk servers
interconnected via 2 LAN trunks at 1Gbps (no nat, no firewall).
One trunk is SIP and the other IAX2.
Normally, I use IAX2 but have noticed easily
--- On Thu, 5/13/10, Gareth Blades list-aster...@skycomuk.com wrote:
Show the details on the active
channels when using both methods and
check what codecs are being used.
The audio codecs are different:
Type: SIP
State: Up (6)
Rings: 0
NativeFormats: 0x4
There should be no noticeable difference between slin, ulaw and alaw so
what you have is fine. The problem must be elsewhere.
Vieri wrote:
--- On Thu, 5/13/10, Gareth Blades list-aster...@skycomuk.com wrote:
Show the details on the active
channels when using both methods and
check what
Hello,
Anyone have any insight or fix for the error below? It was the last error
in the log before Asterisk crashed. I am running Asterisk 1.6.2.6 only for
the T.38 support.
06:21:49] ERROR[1906] astobj2.c: refcount -1 on object 0xb1aab758
Google has some vague references that there is a
Hello list,
I have the following problem with MySQL-queries : it seems that the
resultid and connid are not cleared !
[macro-GetMailboxFromSIPuserID]
exten = s,1,MYSQL(Connect connid localhost xxx xxx xxx)
exten = s,n,MYSQL(Query resultid ${connid} SELECT\ extensie FROM\
tbl_SIPaccounts\
Jonas Kellens wrote:
exten = s,n,NoOp(fetchid = ${fetchid})
exten = s,n,MYSQL(Clear ${resultid})
exten = s,n,MYSQL(Disconnect ${connid})
The only different between yours and mine is that I do a disconnect
before I do the clear.
Try:
exten = s,n,MYSQL(Fetch fetchid ${resultid} extensie)
sorry, you r right i just checked it with registration so there were astdb
entries for SIP registration.
anyhow after clearing settings frm astdb i tried the same scenario you
advised but no luck.
I think i told that i am not using server as peer but want to use a user
[abc] as peer so that when
Ok, I ended up upgrading 2 of my 5 boxes to 1.6.2.7 , and using spandsp
0.0.6pre17, dahdi-linux-complete-2.3.0+2.3.0 , and enabled app_fax.
Hint: you need to install spandsp then run ./configure then make menuselect
:)
I was able to send over a 50 page fax from coast to coast with 0 issues
On Wed, May 12, 2010 at 05:29:29PM +0800, Andrew Furey wrote:
Hi all,
I have a demo machine I'm running up on Lenny - it has the packaged
Asterisk version installed (1.4.21.2+stuff).
Specifically, builds 3 different variants of app_voicemail.so as
different modules (app_voicemail.so,
All,
I am looking into open source idea's for something I play with on
the closed source side. What I am thinking is to get two Asterisk PBX's
behind a single SIP proxy to load balance calls inbound, and potentially
outbound to an external sip provider, with the potential of multiple
Hello,
Can you try trunk = no ?
How much jitter do you see on the link ?
Zoa
Gareth Blades wrote:
There should be no noticeable difference between slin, ulaw and alaw so
what you have is fine. The problem must be elsewhere.
Vieri wrote:
--- On Thu, 5/13/10, Gareth Blades
Can Skype for Asterisk process instant messages from Skype users?
I'm wondering if they can be forwarded via email or SMS.
TIA and regards to all
Enrique Mora
Context M.I.S. SL
em...@context.es
Skype: context-m.i.s.
[cid:image001.jpg@01CAF2BF.0F497C70]
inline: image001.jpg--
Is it possible to have Asterisk resend the SIP credentials in every INVITE?
--
Mike A. Leonetti
As warm as green tea
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Dear all,
I have a question about session timers. I have one of my installations
(* 1.6.2.7) where all SIP calls get stuck, like this:
cs4wall*CLI sip show channels
Peer User/ANR Call ID Format Hold
Last MessageExpiry
192.168.40.178 42
The whole idea of TIMEOUT(absolute) is to end to call after a certain
time. My advice is to explain what you are trying to achieve, there
might be a solutions but I doubt you will find it while using
TIMEOUT(absolute). If the dial plan reaches the t or T extension there
are, as far as I
Hello,
If you have canreinvite=no and a peer sends you a re-invite, what will
Asterisk reply with?
Thanks,
--
David Cunningham, Voisonics
http://voisonics.com/
US toll-free: +1 888 842 2720
UK: +44 (0) 20 3298 1642
Australia: +61 (0) 2 9037 2180
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It is possible. I do a whole lot of processing after dial and before hanging
up a call. In your case you can try using something like:
exten = h,1,Playback(blah)
exten = h,2,HangUp()
And make sure these lines are in the same context where the Dial command is.
There are other ways too to achieve
On 05/13/2010 01:41 PM, David Cunningham wrote:
If you have canreinvite=no and a peer sends you a re-invite, what will
Asterisk reply with?
It will accept it. 'canreinvite' is mis-named, and that's why in more
modern versions of Asterisk it has been renamed to 'directmedia'.
Asterisk will
Kevin,
Thank you for that reply!
We're having an issue where a peer's response to an INVITE includes
a=sendonly. Later it sends a re-invite with a=sendrecv, however
Asterisk responds to that with an OK that includes a=recvonly. The
end result is the called party can't hear the caller.
Do you
On 05/13/2010 05:16 PM, David Cunningham wrote:
We're having an issue where a peer's response to an INVITE includes
a=sendonly. Later it sends a re-invite with a=sendrecv, however
Asterisk responds to that with an OK that includes a=recvonly. The
end result is the called party can't hear the
When you press *1 in Asterisk (1.6.2.7) to start/stop call recording,
the console CLI shows:
User hit '*1' to record call. filename: wav,auto-1273791789-103-5551212,m
Is it possible to play a sound to back to the person who pressed *1 to
indicate to them that recording has actually started or
I want to make a web interface so my users can listen/erase voicemails.
Is there a way to do this from the Asterisk manager interface? Since
Asterisk and the web server do not run as the same user I cannot do a
direct manipulation of the voicemail files
in /var/spool/asterisk/voicemail.
You might be able to use local channels to do what you want.
As for the user asterisk runs as and the user the web server run as you can
maybe have both users belong to the same secondary group and gain the access
you need that way.
Partly depends on what exactly you are wanting to do.
--
Jim
On 13/05/2010, Tzafrir Cohen tzafrir.co...@xorcom.com wrote:
Specifically, builds 3 different variants of app_voicemail.so as
different modules (app_voicemail.so, app_voicemail_imap.so,
app_voicemail_odbc.so).
Correct; the other two were noload(ed) by default so I left them.
What happens
I have a delay of 0 on SpeecBackGround, but when I enter DTMF, there's an
almost-exactly five second delay before it returns. Where is this
delay controlled? How can I shorten it?
Is there a way to set the maximum number of digits to look for?
--
Dear all,
using asterisk-1.4.23.1, I encountered a problem of asterisk that cannot
release the channel.* *
We have several of asterisk server(client ,Guest). Now channels remaining
problem occurs only in the server where the number of user agent is more
than 660 and where many simultaneous
hello
i hope i am posting to the right list, i am a totally blind user, and i want
to reprogram my aastra pt 480e phone, my friend used the web configurator,
but i think he programmed thw wrong codes, a few questions, is it possible
to damage the phone by programming it wrong? also, how does
Hello Everyone,
Are these indications of attacks on this system? I specifically have port 22
disabled at all times and only port forward it to server when I access SSH
for a minute or so. Shouldn't UNKNOWN be an actual IP address?
*/var/log/secure:*
May 14 00:35:39 pbx sshd[9011]: Did not
Unplugging just turns off the phone and has no effect on the settings. You
can not damage the phone by tampering configurations but you can mess up
the settings and it might not register, send, or receive calls.
User manu for your reference:
On Fri, 14 May 2010, bruce bruce wrote:
Are these indications of attacks on this system? I specifically have
port 22 disabled at all times and only port forward it to server when I
access SSH for a minute or so. Shouldn't UNKNOWN be an actual IP
address?
/var/log/secure:
May 14
Talking about file permissions, on Linux everything is possible using
POSIX ACLs. You can set specific rights to files/directories for
certain users.
Note 1: if setting group permissions is enough, use that.
Note 2: Asterisk and web server should be on separate machines (at
least virtual machines)
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