[asterisk-users] Re: Fritz!Box Fon ATA
Martin Schrott - Thinking-Systems
martin.schrott at thinking-systems.eu
Fri Jul 28 04:53:50 MST 2006
Hi again,
should both be possible. With a bit of work ;-)
1. incoming.
You will have to set Rufumleitung to your choosen sip destination.
telefonie>
Rufumleitung>
set the fone, that is set up to be ringing to be forwarded to your sip
extension. As named in your extensions.conf local context.
All incoming calls should then be forwarded to your asterisk.
2. Outbound
Not as easy. Maybe you can realize that as follows:
Telefonie>
Rufumleitung>
Callthrough
(Direktdurchwahl)
You may be able to set internet calls from a given did to be presented a
callthrough option.
Set internetcalls e.g. from 12345 to callthrough via Festnetz (psd)
Then it should be possible to dial through when calling from Asterisk to
your Fritz!Box if your callerid is 12345.
(Never tested this. But with a bit of luck and time you can do it :-) )
all the best hth
Martin
----- Original Message -----
From: "Manuel Dominguez" <manuelmovil at teleline.es>
To: <asterisk-users at lists.digium.com>
Sent: Friday, July 28, 2006 12:13 PM
Subject: [asterisk-users] Re: Fritz!Box Fon ATA
Hi Martin,
Thank you for your comments. I made more or less these settings and in this
moment I can make call from de FXS port to asterisk and from asterisk to FXS
ports.
My problem it's the FXO part of this ATA. I want to redirect all the
incoming ISDN calls to a SIP phone or to an autoatendant and to make
outgoing calls from sip phones (asterisk). I'm not sure if it s possible
make this work using this ATA and the necessary settings.
Manuel
------------------------------
Message: 8
Date: Fri, 28 Jul 2006 09:59:07 +0200
From: "Martin Schrott - Thinking-Systems"
<martin.schrott at thinking-systems.eu>
Subject: Re: [asterisk-users] Fritz!Box Fon ATA
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <002101c6b21b$b49bac40$0100a8c0 at dicore.net>
Content-Type: text/plain; charset="iso-8859-1"
Hi Manuel,
I do not know the ATA, but we have a Fritz!Box fon, wich maybe is equal in
setting it up.
If you have any problems understanding the german setup, you can contact me,
so I can help you in translating the needed Words :-)
Normally you only have to do this on the Webinterface:
Telefonie>
Internettelefonie>
Internetrufnummern>
Neue Internetrufnummer>
Internetrufnummer: Your VOIP number, or if using with isdn, then the msn.
Do not use Internetrufnummer zum Anmelden verwenden!
Registrar: the ip or host of your provider or Asterisk. If you have a own
Asterisk use yur ip adress. There is a bug using hostnames.
Benutzername: Username
Passwort / Kennwort : password
Do only fill out this fields, then it should work. If you put in any proxy
or Stun Servers it may not work. (our experience)
hth,
Martin
----- Original Message -----
From: "Manuel Dominguez" <manuelmovil at teleline.es>
To: <asterisk-users at lists.digium.com>
Sent: Friday, July 28, 2006 9:25 AM
Subject: [asterisk-users] Fritz!Box Fon ATA
Hi,
I have bought a Fritz!Box Fon ATA in eBay. I'm trying to find information
about configuration this box in Asterisk.
Its possible use this box like a normal ATA (sipura 3000.) receiving and
making ISDN calls from Asterisk? Somebody has information in English about
this box? Some example settings?
Another problem is that firmware is in German. I have tried to change it but
was not possible to use a difference language. Some ideas?
Any help would be greatly appreciated
Manuel
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Message: 9
Date: Fri, 28 Jul 2006 09:04:26 +0100
From: "Steve Davies" <davies147 at gmail.com>
Subject: Re: [asterisk-users] SNOM 360
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<5caa9b870607280104w4a4d32c2hbd9b867888a594a6 at mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
On 7/28/06, Koopmann, Jan-Peter <Jan-Peter.Koopmann at seceidos.de> wrote:
> On Freitag, 28. Juli 2006 12:37 Dovid Bender wrote:
>
> > Does anyone know how to set up QoS on the SNOM 360 ? Thanks.
>
> What _EXACTLY_ are you trying to accomplish? There is no simply QoS switch
on a Snom 360 that will manage things for you. AFAIK all you can do is tell
the phone (or * or whatever) what TOS bits to use (or DIFFSERV). It is the
task of the equipment managing the bottleneck (firewall, router whatever) to
use this information and manage your traffic accordingly.
>
As I understand it, you can set a QoS priority if the phone is in a
VLAN. When you configure the (Tagged) VLAN, you can specify the
priority of the packets in the VLAN.
Otherwise, newer firmware allows the setting of TOS values IIRC.
Regards,
Steve
------------------------------
Message: 10
Date: Fri, 28 Jul 2006 10:11:35 +0200
From: Olivier MONNET <olivier.monnet at altiva.fr>
Subject: [asterisk-users] PAP2T always busy on incoming calls with
zaptel
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <FCAF2743-EC44-4B32-B214-2FCBF703FC3C at altiva.fr>
Content-Type: text/plain; charset=US-ASCII; delsp=yes; format=flowed
Hi,
I'm starting to use the new PAP2T instead of the old PAP2-NA for my
new installations.
I'm having a weird problem: when a call is comming from a zaptel
channel (from a bri with bristuff driver) the PAP2T say BUSY to the
SIP channel.
I have disabled all the features like DND and call forward.
If it's the last line for this number in the dialplan I can answer
the call normally, but I can't use voicemail, because it jump to it
each time.
I have installed about 50 PAP-NA and never had this kind of problem.
If the call is coming from an other PAP2T (via asterisk with
canreinvite=no), everything is fine.
This occur with asterisk 1.0.10 and 1.2.9.1
the firmware version for the PAP2T is 3.1.9(LSc)
I am using a dialplan coming from another customer with a similar
setup, but with PAP2-NA, where it's working fine.
What can I do to fix this.
Regards,
Olivier
------------------------------
Message: 11
Date: Fri, 28 Jul 2006 10:30:50 +0200
From: Olivier <oza-4h07 at myamail.com>
Subject: Re: [asterisk-users] Sip phone settings set when user
registers
To: nik.engel at googlemail.com, "Asterisk Users Mailing List -
Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Message-ID:
<442fbb120607280130h35bf28e5lf7254a292a0f4859 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
2006/7/27, Nik Engel <nik.engel at googlemail.com>:
>
> User logs into any phone and the settings of the phone are always the
> same. Meaning individual key
> assignement is always the same.
>
> Hi,
Do you mean :
1. Without user logins, phones are unusable ? Or do you plan to offer
default services (local calls for instance) for unidentifed users ? I'm not
sure many phones offer special keys for login-logout.
2. What should happen when users change phones settings ? Shall these
changes be saved somehow (during logoffs ?) and somewhere for latter reuse ?
That implies phone config should be portable from one phone to another. That
doesn't seem easy if phones are installed in different locations.
Regards
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Message: 12
Date: Fri, 28 Jul 2006 10:52:20 +0200
From: Olivier <oza-4h07 at myamail.com>
Subject: Re: [asterisk-users] RE: alcatel ip touch 4068 ... sip?
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID:
<442fbb120607280152vbde2371j311253d5f247e0a3 at mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
2006/7/28, Leo Ann Boon <leo at datvoiz.com>:
>
> (AstATN) wrote:
>
> >Hi Cesc,
> >Alcatel does not offer any SIP Terminal, it phone's are H.323 it's handle
> >for their own features usages. ( like ADSI type )
> >
> >
> Common misconception. Their phones are not H.323 despite claims in their
> documentation. The server has to do the signaling conversion. The native
> protocol is UAIP (User Agent IP) which runs over UDP.
Hi,
I've never heard of that (UAIP) before !
Do you have anything describing this protocol ?
Would it be difficult to implement it inside Asterisk just like UNISTIM has
been ?
Regards
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Message: 13
Date: Fri, 28 Jul 2006 10:55:58 +0200
From: "Joseph Dudash" <j.dudash at fonlight.com>
Subject: [asterisk-users] asterisk+ooh323.. one way audio issue
To: <asterisk-users at lists.digium.com>
Message-ID: <002901c6b223$a53521c0$2b8f7457 at computer>
Content-Type: text/plain; charset="iso-8859-1"
Hi guys,
I tried to make call from SIP channel to H323 using asterisk+ooh323. The SIP
client is x-lite.
The problem is that there is one way audio. I hear everything from h323
endpoint, and I see the messages also:
Got RTP packet from 66.135.35.xx:5002 (type 3, seq 36250, ts 74400, len 33)
Sent RTP packet to 212.183.41.xx:45956 (type 18, seq 22288, ts 70880, len
20)
But the problem, when I talk via X-lite, or send dtmf tones, no audio is
transfered, no RTP packets on asterisk console.
My ooh323.conf:
[general]
port=1720
bindaddr=0.0.0.0
gateway=no
h323id=ObjSysAsterisk
e164=100
callerid=asterisk
gatekeeper = DISABLE
disallow=all
allow=g729
allow=gsm
allow=ulaw
Note that I tried all combinations of faststart and h245tunneling, but no
luck.
Also tried with gsm and g729 codecs (that time X-pro was used) but same
oneway audio.
Asterisk version is 1.2.7.1
With full debug this is what I see in asterisk console:
Jul 28 10:40:22 DEBUG[15775]: pbx.c:1674 pbx_extension_helper: Launching
'Dial'
Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables:
Not copying variable STACK-test-381637790067-2.
Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables:
Not copying variable STACK-test-381637790067-1.
Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables:
Not copying variable SIPCALLID.
Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables:
Not copying variable SIPUSERAGENT.
Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables:
Not copying variable SIPDOMAIN.
Jul 28 10:40:22 DEBUG[15775]: channel.c:2813 ast_channel_inherit_variables:
Not copying variable SIPURI.
Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel
OOH323/66.135.33.xx-14b0 to read format gsm
Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel
SIP/666666-9cbb to write format gsm
Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel
SIP/666666-9cbb to read format gsm
Jul 28 10:40:22 DEBUG[15775]: channel.c:2348 set_format: Set channel
OOH323/66.135.33.xx-14b0 to write format gsm
Jul 28 10:40:22 DEBUG[16193]: res_config_mysql.c:636 mysql_reconnect: MySQL
RealTime: Everything is fine.
Jul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel
SIP/666666-9cbb to read format gsm
Jul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel
OOH323/66.135.33.xx-14b0 to write format gsm
Jul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel
OOH323/66.135.33.xx-14b0 to read format gsm
Jul 28 10:40:23 DEBUG[15775]: channel.c:2348 set_format: Set channel
SIP/666666-9cbb to write format gsm
Jul 28 10:40:23 DEBUG[15775]: chan_sip.c:2527 sip_answer:
sip_answer(SIP/666666-9cbb)
Jul 28 10:40:23 DEBUG[15775]: channel.c:1956 ast_read: Dropping duplicate
answer!
Jul 28 10:40:23 DEBUG[16193]: res_config_mysql.c:125 realtime_mysql: MySQL
RealTime: Retrieve SQL: SELECT * FROM sip_buddies WHERE name = '666666'
Jul 28 10:40:23 DEBUG[16193]: res_config_mysql.c:636 mysql_reconnect: MySQL
RealTime: Everything is fine.
Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:1394 __sip_ack: Stopping
retransmission on '4D979F84-6EF9-4A17-9007-4082B33E6835 at 87.116.143.xx' of
Response 6305: Match Found
Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:9442 check_pendings: Sending
pending reinvite on '4D979F84-6EF9-4A17-9007-4082B33E6835 at 87.116.143.xx'
Jul 28 10:40:23 DEBUG[15775]: rtp.c:410 ast_rtcp_read: Got RTCP report of 84
bytes
Got RTP packet from 87.116.143.xx:8000 (type 3, seq 1, ts 5920, len 33)
Jul 28 10:40:23 DEBUG[15775]: rtp.c:1341 ast_rtp_write: Ooh, format changed
from unknown to gsm
Sent RTP packet to 66.135.33.xx:5004 (type 3, seq 54271, ts 0, len 33)
Got RTP packet from 87.116.143.xx:8000 (type 3, seq 2, ts 6080, len 33)
Sent RTP packet to 66.135.33.xx:5004 (type 3, seq 54272, ts 160, len 33)
Got RTP packet from 87.116.143.xx:8000 (type 3, seq 3, ts 6240, len 33)
Sent RTP packet to 66.135.33.xx:5004 (type 3, seq 54273, ts 320, len 33)
Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:1447 __sip_semi_ack: (Provisional)
Stopping retransmission (but retaining packet) on
'4D979F84-6EF9-4A17-9007-4082B33E6835 at 87.116.143.xx' Request 102: Found
Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:1372 __sip_ack: Acked pending
invite 102
Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:1394 __sip_ack: Stopping
retransmission on '4D979F84-6EF9-4A17-9007-4082B33E6835 at 87.116.143.xx' of
Request 102: Match Found
Jul 28 10:40:23 DEBUG[16207]: chan_sip.c:6047 build_route: build_route:
Contact hop: <sip:666666 at 87.116.143.xx:5060>
Got RTP packet from 66.135.33.xx:5004 (type 3, seq 55961, ts 3520, len 33)
Jul 28 10:40:24 DEBUG[15775]: src/chan_h323.c:3045 ooh323_rtp_read: Oooh,
format changed to 2
Jul 28 10:40:24 DEBUG[15775]: channel.c:2348 set_format: Set channel
OOH323/66.135.33.xx-14b0 to read format gsm
Jul 28 10:40:24 DEBUG[15775]: channel.c:2348 set_format: Set channel
OOH323/66.135.33.xx-14b0 to write format gsm
Jul 28 10:40:24 DEBUG[15775]: rtp.c:1341 ast_rtp_write: Ooh, format changed
from unknown to gsm
Sent RTP packet to 87.116.143.xx:8000 (type 3, seq 50183, ts 0, len 33)
Got RTP packet from 66.135.33.xx:5004 (type 3, seq 55962, ts 3680, len 33)
Sent RTP packet to 87.116.143.xx:8000 (type 3, seq 50184, ts 160, len 33)
Got RTP packet from 66.135.33.xx:5004 (type 3, seq 55963, ts 3840, len 33)
Sent RTP packet to 87.116.143.xx:8000 (type 3, seq 50185, ts 320, len 33)
Got RTP packet from 66.135.33.xx:5004 (type 3, seq 55964, ts 4000, len
Got RTP packet from 66.135.33.xx:5004 (type 3, seq 56070, ts 20960, len 33)
Sent RTP packet to 87.116.143.xx:8000 (type 3, seq 50292, ts 17440, len 33)
Jul 28 10:40:26 DEBUG[15775]: rtp.c:1700 ast_rtp_bridge: Oooh, got a hangup
Jul 28 10:40:26 DEBUG[15775]: channel.c:3478 ast_channel_bridge: Returning
from native bridge, channels: SIP/666666-9cbb, OOH323/66.135.33.xx-14b0
Jul 28 10:40:26 DEBUG[15775]: channel.c:1323 ast_hangup: Hanging up channel
'OOH323/66.135.33.xx-14b0'
Jul 28 10:40:26 DEBUG[15775]: app_dial.c:1605 dial_exec_full: Exiting with
DIALSTATUS=ANSWER.
Jul 28 10:40:26 DEBUG[15775]: pbx.c:2313 __ast_pbx_run: Spawn extension
(test,381637790067,2) exited non-zero on 'SIP/666666-9cbb'
Jul 28 10:40:26 DEBUG[15775]: cdr.c:992 ast_cdr_detach: Dropping CDR !
Jul 28 10:40:26 DEBUG[15775]: channel.c:1323 ast_hangup: Hanging up channel
'SIP/666666-9cbb'
Jul 28 10:40:26 DEBUG[15775]: chan_sip.c:2411 sip_hangup: Hangup call
SIP/666666-9cbb, SIP callid
4D979F84-6EF9-4A17-9007-4082B33E6835 at 87.116.143.xx)
Jul 28 10:40:26 DEBUG[15775]: chan_sip.c:2419 sip_hangup:
update_call_counter(666666) - decrement call limit counter
And this is from /var/log/asterisk/h323_log:
07:20:01:413 Processing MakeCall command ooh323c_o_3
07:20:01:413 Created a new call (outgoing, ooh323c_o_3)
07:20:01:413 Enabled RFC2833 DTMF capability for (outgoing, ooh323c_o_3)
07:20:01:413 Parsing destination 216.12.173.48
07:20:01:413 Trying to connect to remote endpoint(216.12.173.48:1720) to
setup H2250 channel (outgoing, ooh323c_o_3)
07:20:01:501 H2250 transmiter channel creation - succesful (outgoing,
ooh323c_o_3)
07:20:01:502 Sent Message - Setup (outgoing, ooh323c_o_3)
07:20:01:671 H.225 Call Proceeding message received (outgoing, ooh323c_o_3)
07:20:01:671 Tunneling is disabled for call as H245 address is provided in
callProceeding message (outgoing, ooh323c_o_3)
07:20:01:759 H.225 Progress message received (outgoing, ooh323c_o_3)
07:20:13:412 H.225 Alerting message received (outgoing, ooh323c_o_3)
07:20:19:307 H.225 Connect message received (outgoing, ooh323c_o_3)
07:20:19:307 Remote endpoint has rejected fastStart. (outgoing,
ooh323c_o_3)
07:20:19:307 Clearing all logical channels (outgoing, ooh323c_o_3)
07:20:19:307 Creating H245 Connection
07:20:19:307 Trying to connect to remote endpoint to setup H245 connection
216.12.173.49:54659(outgoing, ooh323c_o_3)
07:20:19:396 H245 connection creation succesful (outgoing, ooh323c_o_3)
07:20:19:397 H.225 Notify message Received (outgoing, ooh323c_o_3)
07:20:19:397 Sent Message - TerminalCapabilitySet (outgoing, ooh323c_o_3)
07:20:19:397 Sent Message - MasterSlaveDetermination (outgoing,
ooh323c_o_3)
07:20:19:486 Reducing framesPerPkt for transmission of GSM capability from
4 to 1 to match receive capability of remote endpoint.(outgoing,
ooh323c_o_3)
07:20:19:486 Reducing framesPerPkt for transmission of Simple capability
from 6 to 2 to match receive capability of remote endpoint.(outgoing,
ooh323c_o_3)
07:20:19:486 Reducing framesPerPkt for transmission of Simple capability
from 6 to 2 to match receive capability of remote endpoint.(outgoing,
ooh323c_o_3)
07:20:19:486 Master Slave Determination received (outgoing, ooh323c_o_3)
07:20:19:486 MasterSlaveDetermination done - Slave(outgoing, ooh323c_o_3)
07:20:19:486 Sent Message - TerminalCapabilitySetAck (outgoing,
ooh323c_o_3)
07:20:19:486 Opening logical channels (outgoing, ooh323c_o_3)
07:20:19:486 Looking for matching capabilities. (outgoing, ooh323c_o_3)
07:20:19:486 Created new logical channel entry (outgoing, ooh323c_o_3)
07:20:19:486 Sent Message - MasterSlaveDeterminationAck (outgoing,
ooh323c_o_3)
07:20:19:486 Sent Message - OpenLogicalChannel(1001). (outgoing,
ooh323c_o_3)
07:20:19:664 Created new logical channel entry (outgoing, ooh323c_o_3)
07:20:19:664 Receive channel of type OO_G729 started at
213.203.222.xxx:12222(outgoing, ooh323c_o_3)
07:20:19:664 TransmitLogical Channel of type OO_GSMFULLRATE started
(outgoing, ooh323c_o_3)
07:20:19:664 Sent Message - OpenLogicalChannelAck(1) (outgoing,
ooh323c_o_3)
07:20:19:664 Received close logical Channel - 1 (outgoing, ooh323c_o_3)
07:20:19:664 Closing logical channel number 1 (outgoing, ooh323c_o_3)
07:20:19:664 Stopped Receive channel 1 (outgoing, ooh323c_o_3)
07:20:19:664 Created new logical channel entry (outgoing, ooh323c_o_3)
07:20:19:664 Receive channel of type OO_GSMFULLRATE started at
213.203.222.xxx:12222(outgoing, ooh323c_o_3)
07:20:19:664 Sent Message - CloseLogicalChannelAck (outgoing, ooh323c_o_3)
07:20:19:664 Sent Message - OpenLogicalChannelAck(2) (outgoing,
ooh323c_o_3)
07:20:34:690 H.225 Release Complete message received (outgoing,
ooh323c_o_3)
07:20:34:690 Closing H.245 connection (outgoing, ooh323c_o_3)
07:20:34:690 Cleaning Call (outgoing, ooh323c_o_3)-
reason:OO_REASON_REMOTE_CLEARED
07:20:34:690 Clearing all logical channels (outgoing, ooh323c_o_3)
07:20:34:690 Stopped Transmit channel 1001 (outgoing, ooh323c_o_3)
07:20:34:690 Stopped Receive channel 2 (outgoing, ooh323c_o_3)
07:20:34:690 Removed call (outgoing, ooh323c_o_3) from list
Any idea?:)
Thanks,
Joseph
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Message: 14
Date: Fri, 28 Jul 2006 09:59:16 +0100
From: Kenny Millington <kenny at 3ait.co.uk>
Subject: [asterisk-users] Asterisk 1.2.10 - Continually Restarting
Logger
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <44C9D1E4.4060805 at 3ait.co.uk>
Content-Type: text/plain; charset=ISO-8859-1
Hi,
We're seeing a problem on Asterisk 1.2.10 where when we get in in the
morning it's continually rotating the logs over and over again,
generating 100's of thousands of log rotated 0 byte files:-
/var/logs/asterisk # find . -type f -maxdepth 1 | wc -l
176930
/var/log/asterisk # find . -type f -maxdepth 1 -size 0 -exec mv {} nulls/ \;
/var/log/asterisk # find . -type f -maxdepth 1 | wc -l
69169
A segment of the relevant log is:-
Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[9635] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[9638] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Queue Logger restarted
Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
Jul 25 06:33:42 VERBOSE[9641] logger.c: -- Remote UNIX connection
disconnected
Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Event Logger restarted
Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Queue Logger restarted
etc...
Has anyone else seen this or have any ideas what the problem may be?
Thanks,
--
Kenny Millington
Systems Developer
3aIT Limited
T: 0870 881 5097
F: 01403 248 105
E: kenny.millington at 3ait.co.uk
W: http://www.3ait.co.uk
------------------------------
Message: 15
Date: Fri, 28 Jul 2006 09:59:23 +0100
From: "Dean @ INKnBITs" <dean.bath at inknbits.co.uk>
Subject: [asterisk-users] Sending email after voicemail
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users at lists.digium.com>
Message-ID: <LCEKKIAPJJIPDKOCHPODIECDCJAA.dean.bath at inknbits.co.uk>
Content-Type: text/plain; charset="iso-8859-1"
Hi,
I'm having trouble getting asterisk to send a voicemail message via email. I
can do a mail xxx at email.com from the linux command line and I receive the
email fine, and if I look in the exim4 logs it looks ok, has from user, to
user and completed but no email is received.
Any thoughts?
Thanks,
Dean.
------------------------------
Message: 16
Date: Fri, 28 Jul 2006 11:14:51 +0200
From: Filip Dr?gowski <f.dragowski at ontp.net>
Subject: Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting
Logger
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <44C9D58B.5050308 at ontp.net>
Content-Type: text/plain; charset=ISO-8859-2; format=flowed
check your cron jobs.
mayby there is asterisk -rx "logger rotate" executing too often ?
> Hi,
>
> We're seeing a problem on Asterisk 1.2.10 where when we get in in the
> morning it's continually rotating the logs over and over again,
> generating 100's of thousands of log rotated 0 byte files:-
>
> /var/logs/asterisk # find . -type f -maxdepth 1 | wc -l
> 176930
>
> /var/log/asterisk # find . -type f -maxdepth 1 -size 0 -exec mv {} nulls/
\;
>
> /var/log/asterisk # find . -type f -maxdepth 1 | wc -l
> 69169
>
> A segment of the relevant log is:-
>
> Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Event Logger restarted
> Jul 25 06:33:42 VERBOSE[9638] logger.c: Asterisk Queue Logger restarted
> Jul 25 06:33:42 VERBOSE[9635] logger.c: -- Remote UNIX connection
> disconnected
> Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
> Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Event Logger restarted
> Jul 25 06:33:42 VERBOSE[9641] logger.c: Asterisk Queue Logger restarted
> Jul 25 06:33:42 VERBOSE[9638] logger.c: -- Remote UNIX connection
> disconnected
> Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
> Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Event Logger restarted
> Jul 25 06:33:42 VERBOSE[9644] logger.c: Asterisk Queue Logger restarted
> Jul 25 06:33:42 VERBOSE[18276] logger.c: -- Remote UNIX connection
> Jul 25 06:33:42 VERBOSE[9641] logger.c: -- Remote UNIX connection
> disconnected
> Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Event Logger restarted
> Jul 25 06:33:42 VERBOSE[9647] logger.c: Asterisk Queue Logger restarted
>
> etc...
>
> Has anyone else seen this or have any ideas what the problem may be?
>
> Thanks,
>
------------------------------
Message: 17
Date: Fri, 28 Jul 2006 10:26:08 +0100
From: Kenny Millington <kenny at 3ait.co.uk>
Subject: Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting
Logger
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <44C9D830.9000502 at 3ait.co.uk>
Content-Type: text/plain; charset=ISO-8859-2
Filip Dr±gowski wrote:
> check your cron jobs.
> mayby there is asterisk -rx "logger rotate" executing too often ?
Nope - nothing in crontab.
>> Hi,
>>
>> We're seeing a problem on Asterisk 1.2.10 where when we get in in the
>> morning it's continually rotating the logs over and over again,
>> generating 100's of thousands of log rotated 0 byte files:-
<snip>
--
Kenny Millington
Systems Developer
3aIT Limited
T: 0870 881 5097
F: 01403 248 105
E: kenny.millington at 3ait.co.uk
W: http://www.3ait.co.uk
------------------------------
Message: 18
Date: Fri, 28 Jul 2006 12:34:05 +0300
From: "Khaled Chehab" <kchehab at xplorium.com>
Subject: [asterisk-users] CDR IP Authorization
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<asterisk-users at lists.digium.com>
Cc: <asterisk-users-bounces at lists.digium.com>
Message-ID: <013601c6b228$f9af77a0$6e05a8c0 at ck>
Content-Type: text/plain; charset="us-ascii"
Dear
This function retrieves the ip address of the caller ,I want to import the
value of (recvip) in the mysql cdr ,how can I do that
exten => s,1,NoOp(${SIPCHANINFO(recvip)})
Regards
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Message: 19
Date: Fri, 28 Jul 2006 02:50:52 -0700 (PDT)
From: richard Coco <coco_richard at yahoo.com>
Subject: Re: [asterisk-users] SIP client with video???
To: Asterisk Users Mailing List - Non-Commercial Discussion
<asterisk-users at lists.digium.com>
Message-ID: <20060728095052.47075.qmail at web57015.mail.re3.yahoo.com>
Content-Type: text/plain; charset=iso-8859-1
Hi,
i have xlite too and it works without any problems.
ps: what about ekiga? (www.ekiga.org)
rich
--- Joao Pereira <joao.pereira at fccn.pt> wrote:
> Hello to all
> can someone recommend me a nice SIP client with
> video for windows??
>
> I tried X-Lite 3.0 but it's a lousy piece of
> software.....
>
> Does someone knows about a better software?
> Regards
> Joao Pereira
>
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