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
Martin Schrott - Thinking-Systems
2006-Jul-28 00:59 UTC
[asterisk-users] Fritz!Box Fon ATA
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@teleline.es> To: <asterisk-users@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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
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@thinking-systems.eu> Subject: Re: [asterisk-users] Fritz!Box Fon ATA To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <002101c6b21b$b49bac40$0100a8c0@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@teleline.es> To: <asterisk-users@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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users ------------------------------ Message: 9 Date: Fri, 28 Jul 2006 09:04:26 +0100 From: "Steve Davies" <davies147@gmail.com> Subject: Re: [asterisk-users] SNOM 360 To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <5caa9b870607280104w4a4d32c2hbd9b867888a594a6@mail.gmail.com> Content-Type: text/plain; charset=ISO-8859-1; format=flowed On 7/28/06, Koopmann, Jan-Peter <Jan-Peter.Koopmann@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 switchon 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@altiva.fr> Subject: [asterisk-users] PAP2T always busy on incoming calls with zaptel To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <FCAF2743-EC44-4B32-B214-2FCBF703FC3C@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@myamail.com> Subject: Re: [asterisk-users] Sip phone settings set when user registers To: nik.engel@googlemail.com, "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <442fbb120607280130h35bf28e5lf7254a292a0f4859@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 2006/7/27, Nik Engel <nik.engel@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/b8a1c9 87/attachment-0001.htm ------------------------------ Message: 12 Date: Fri, 28 Jul 2006 10:52:20 +0200 From: Olivier <oza-4h07@myamail.com> Subject: Re: [asterisk-users] RE: alcatel ip touch 4068 ... sip? To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <442fbb120607280152vbde2371j311253d5f247e0a3@mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" 2006/7/28, Leo Ann Boon <leo@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/77bddf da/attachment-0001.htm ------------------------------ Message: 13 Date: Fri, 28 Jul 2006 10:55:58 +0200 From: "Joseph Dudash" <j.dudash@fonlight.com> Subject: [asterisk-users] asterisk+ooh323.. one way audio issue To: <asterisk-users@lists.digium.com> Message-ID: <002901c6b223$a53521c0$2b8f7457@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@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@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@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@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@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@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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/a72a6d 2f/attachment-0001.htm ------------------------------ Message: 14 Date: Fri, 28 Jul 2006 09:59:16 +0100 From: Kenny Millington <kenny@3ait.co.uk> Subject: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <44C9D1E4.4060805@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@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@inknbits.co.uk> Subject: [asterisk-users] Sending email after voicemail To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <LCEKKIAPJJIPDKOCHPODIECDCJAA.dean.bath@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@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@ontp.net> Subject: Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <44C9D58B.5050308@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@3ait.co.uk> Subject: Re: [asterisk-users] Asterisk 1.2.10 - Continually Restarting Logger To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <44C9D830.9000502@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@3ait.co.uk W: http://www.3ait.co.uk ------------------------------ Message: 18 Date: Fri, 28 Jul 2006 12:34:05 +0300 From: "Khaled Chehab" <kchehab@xplorium.com> Subject: [asterisk-users] CDR IP Authorization To: "'Asterisk Users Mailing List - Non-Commercial Discussion'" <asterisk-users@lists.digium.com> Cc: <asterisk-users-bounces@lists.digium.com> Message-ID: <013601c6b228$f9af77a0$6e05a8c0@ck> Content-Type: text/plain; charset="us-ascii" Dear This function retrieves the ip address of the caller ,I want to import the value of (recvip) in the mysql cdr ,how can I do that exten => s,1,NoOp(${SIPCHANINFO(recvip)}) Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of Xplorium. 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URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060728/106fb7 04/attachment-0001.htm ------------------------------ Message: 19 Date: Fri, 28 Jul 2006 02:50:52 -0700 (PDT) From: richard Coco <coco_richard@yahoo.com> Subject: Re: [asterisk-users] SIP client with video??? To: Asterisk Users Mailing List - Non-Commercial Discussion <asterisk-users@lists.digium.com> Message-ID: <20060728095052.47075.qmail@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@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 > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com > -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > >http://lists.digium.com/mailman/listinfo/asterisk-users>__________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com ------------------------------ _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users End of asterisk-users Digest, Vol 24, Issue 165 ***********************************************
Hi Martin, you say only a bit of work? ;-) 1. Incoming Yes, works like you suggest me!! The problem is that using this method, it's not possible to use the FXS ports in the Fritz!Box like normal extensions from Asterisk. We only use it to forward calls to a SIP extension. 2. Outbound I don?t understand exactly your comments but I think is working. I go to the Rufumleitung -> Durchwahl (Call Through) aktiv -> definierte Durchwahl. In the combo box "Durchwahl f?r Anrufe auf der Rufnummer" I select my connection to Asterisk. I write a PIN and in the combo box "Anrufe weiterverbinden ?ber die Rufnummer" I select the Festnetz.>From a SIP phone, I make a call to the extension selected in "Durchwahl f?rAnrufe auf der Rufnummer". In that moment another tone appears, I enter the PIN and I can make an external call from the SIP phone. Thanks for you help & greeting from Spain Manuel -----Mensaje original----- De: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] En nombre de asterisk-users-request@lists.digium.com Enviado el: viernes, 28 de julio de 2006 13:50 Message: 16 Date: Fri, 28 Jul 2006 13:53:50 +0200 From: "Martin Schrott - Thinking-Systems" <martin.schrott@thinking-systems.eu> Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <000401c6b23c$7ef67220$0100a8c0@dicore.net> Content-Type: text/plain; charset="iso-8859-1" Hi again, should both be possible. With a bit of work ;-) 1. incoming. You will have to set Rufumleitung to your choosen sip destination. telefonie> Rufumleitung> set the fone, that is set up to be ringing to be forwarded to your sip extension. As named in your extensions.conf local context. All incoming calls should then be forwarded to your asterisk. 2. Outbound Not as easy. Maybe you can realize that as follows: Telefonie> Rufumleitung> Callthrough (Direktdurchwahl) You may be able to set internet calls from a given did to be presented a callthrough option. Set internetcalls e.g. from 12345 to callthrough via Festnetz (psd) Then it should be possible to dial through when calling from Asterisk to your Fritz!Box if your callerid is 12345. (Never tested this. But with a bit of luck and time you can do it :-) ) all the best hth Martin ----- Original Message ----- From: "Manuel Dominguez" <manuelmovil@teleline.es> To: <asterisk-users@lists.digium.com> Sent: Friday, July 28, 2006 12:13 PM Subject: [asterisk-users] Re: Fritz!Box Fon ATA Hi Martin, Thank you for your comments. I made more or less these settings and in this moment I can make call from de FXS port to asterisk and from asterisk to FXS ports. My problem it's the FXO part of this ATA. I want to redirect all the incoming ISDN calls to a SIP phone or to an autoatendant and to make outgoing calls from sip phones (asterisk). I'm not sure if it s possible make this work using this ATA and the necessary settings. Manuel ------------------------------ Message: 8 Date: Fri, 28 Jul 2006 09:59:07 +0200 From: "Martin Schrott - Thinking-Systems" <martin.schrott@thinking-systems.eu> Subject: Re: [asterisk-users] Fritz!Box Fon ATA To: "Asterisk Users Mailing List - Non-Commercial Discussion" <asterisk-users@lists.digium.com> Message-ID: <002101c6b21b$b49bac40$0100a8c0@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@teleline.es> To: <asterisk-users@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 _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Hi Martin,
No exactly. The Fritz!Box is connected to Asterisk using SIP. Not a direct
connection between FXS ports and Asterisk.
I would like to use this box like a Sipura 3000. This Sipura has 1 FXO port
and 1 FXS port. You can use and register these ports in Asterisk
independently. You register de FXS port like a normal extension in SIP.conf
and you can use the FXO port for outbound calls from any extension (SIP or
analog phones using FXS ports).
With Fritz!Box to redirect all the calls from ISDN to Asterisk the only
possibility we found is in the Rufumleitung menu. But in this menu you can't
select the FXO port to redirect to Asterisk. You must select the FXS port
(FON 1 or 2). This is ok but you can't use these ports to add other
extensions.
I find much information people making new firmware, changing settings inside
Linux, using in asterisk... but always in German. I try to translate with
Google but it is really complicated and my English is also terrible.
Thanks,
Manuel
Message: 3
Date: Fri, 28 Jul 2006 23:08:00 +0200
From: "Martin Schrott - Thinking-Systems"
<martin.schrott@thinking-systems.eu>
Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <000601c6b289$e8cdc160$0100a8c0@dicore.net>
Content-Type: text/plain; charset="iso-8859-1"
Hi Manuel,
:-)
If I understood you correctly, Your Fritz!Box and Asterisk are also
connected via the fxs Ports?
Then you should also be able to send incoming calls to this ports.
Search for settings of
Nebenstellen, eingehende Anrufe or ankommende Gespr?che...
But I do not see, where the sence would be, when you also can send directly
to a Sip extension?!
When you connect Asterisk via the fxs Ports, then you could directly dial
out, without a Direktruf/Calltrough and pin.
But Fritz!box is not really very userfriendly and not at least flexible. You
can hardly do special configurations. :-(
I am happy, that the things work as i supposed them to do.
Best greetings from Austria
Martin
----- Original Message -----
From: "Manuel Dominguez" <manuelmovil@teleline.es>
To: <asterisk-users@lists.digium.com>
Sent: Friday, July 28, 2006 9:39 PM
Subject: [asterisk-users] Re: Fritz!Box Fon ATA
Hi Martin, you say only a bit of work? ;-)
1. Incoming
Yes, works like you suggest me!! The problem is that using this method, it's
not possible to use the FXS ports in the Fritz!Box like normal extensions
from Asterisk. We only use it to forward calls to a SIP extension.
2. Outbound
I don't understand exactly your comments but I think is working. I go to the
Rufumleitung -> Durchwahl (Call Through) aktiv -> definierte Durchwahl. In
the combo box "Durchwahl f?r Anrufe auf der Rufnummer" I select my
connection to Asterisk. I write a PIN and in the combo box "Anrufe
weiterverbinden ?ber die Rufnummer" I select the
Festnetz.>From a SIP phone, I make a call to the extension selected in "Durchwahl
f?r
Anrufe auf der Rufnummer". In that moment another tone appears, I enter the
PIN and I can make an external call from the SIP phone.
Thanks for you help & greeting from Spain
Manuel
-----Mensaje original-----
De: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] En nombre de
asterisk-users-request@lists.digium.com
Enviado el: viernes, 28 de julio de 2006 13:50
Message: 16
Date: Fri, 28 Jul 2006 13:53:50 +0200
From: "Martin Schrott - Thinking-Systems"
<martin.schrott@thinking-systems.eu>
Subject: Re: [asterisk-users] Re: Fritz!Box Fon ATA
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <000401c6b23c$7ef67220$0100a8c0@dicore.net>
Content-Type: text/plain; charset="iso-8859-1"
Hi again,
should both be possible. With a bit of work ;-)
1. incoming.
You will have to set Rufumleitung to your choosen sip destination.
telefonie>
Rufumleitung>
set the fone, that is set up to be ringing to be forwarded to your sip
extension. As named in your extensions.conf local context.
All incoming calls should then be forwarded to your asterisk.
2. Outbound
Not as easy. Maybe you can realize that as follows:
Telefonie>
Rufumleitung>
Callthrough
(Direktdurchwahl)
You may be able to set internet calls from a given did to be presented a
callthrough option.
Set internetcalls e.g. from 12345 to callthrough via Festnetz (psd)
Then it should be possible to dial through when calling from Asterisk to
your Fritz!Box if your callerid is 12345.
(Never tested this. But with a bit of luck and time you can do it :-) )
all the best hth
Martin
----- Original Message -----
From: "Manuel Dominguez" <manuelmovil@teleline.es>
To: <asterisk-users@lists.digium.com>
Sent: Friday, July 28, 2006 12:13 PM
Subject: [asterisk-users] Re: Fritz!Box Fon ATA
Hi Martin,
Thank you for your comments. I made more or less these settings and in this
moment I can make call from de FXS port to asterisk and from asterisk to FXS
ports.
My problem it's the FXO part of this ATA. I want to redirect all the
incoming ISDN calls to a SIP phone or to an autoatendant and to make
outgoing calls from sip phones (asterisk). I'm not sure if it s possible
make this work using this ATA and the necessary settings.
Manuel
------------------------------
Message: 8
Date: Fri, 28 Jul 2006 09:59:07 +0200
From: "Martin Schrott - Thinking-Systems"
<martin.schrott@thinking-systems.eu>
Subject: Re: [asterisk-users] Fritz!Box Fon ATA
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@lists.digium.com>
Message-ID: <002101c6b21b$b49bac40$0100a8c0@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@teleline.es>
To: <asterisk-users@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
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users