[asterisk-users] peer IP address in CDR
Hi, The subject says it all. Is it possible to put the IP address of the peer in the CDR records? Using AGI maybe? -- Kind regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbress...@signet.nl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] peer IP address in CDR
Hi, Sorry, but i forgot to notice that i am already using the 'userfield' column so that's not a possibility. Is there any way i can add the IP address to a custom MySQL field in CDR? With AGI possibly? The problem is, that the CDR entry is written in MySQL when the call is hungup, so i have no possibility to write the IP address after a call. Regards, Remco On 06/29/2010 03:53 PM, Zeeshan Zakaria wrote: Hi, There is usually an empty column in the cdr table named 'userfield'. You can also add a column of your own. Then in the dialplan use: Set(CDR(userfield)=user IP address) And this will automatically add this information into the userfield column. Do you already have script to capture user's IP address? If not, check it here how I am capturing it: http://www.ilovetovoip.com/2010/05/getting-users-ip-address-remaining-within-the-dialplan Zeeshan A Zakaria -- www.ilovetovoip.com http://www.ilovetovoip.com On 2010-06-29 8:20 AM, Faisal Hanif fai...@vopium.com mailto:fai...@vopium.com wrote: Simply set it to costume field of cdrs in dialplan and you will have it a part of native cdr Regards, *Faisal Hanif* -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Met vriendelijke groet, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbress...@signet.nl altijd online? www.signet.nl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] peer IP address in CDR
Thanks Zeeshan, but i don't use (and understand) AEL :)
Any regular examples out there? :)
regards,
Remco
On 06/29/2010 05:27 PM, Zeeshan Zakaria wrote:
Let me make it simple for you:
Add a column to your table, e.g. `my column`.
In the dialplan do the following (AEL example):
MYSQL(Connect connid localhost username password database);
MYSQL(Query resultid ${connid} INSERT INTO `cdr`
(`mycolumn`) VALUES('${SIPCHANINFO(ip)}'));
MYSQL(Disconnect ${connid});
--
Zeeshan
On Tue, Jun 29, 2010 at 10:32 AM, Gareth Blades
list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote:
Remco Bressers wrote:
Hi,
Sorry, but i forgot to notice that i am already using the 'userfield'
column so that's not a possibility. Is there any way i can add the IP
address to a custom MySQL field in CDR? With AGI possibly? The problem
is, that the CDR entry is written in MySQL when the call is
hungup, so i
have no possibility to write the IP address after a call.
Regards,
Remco
See my earlier post. You can certenly write the information after the
call is hung up by using the 'h' extension. I do this myself to write
the calculated call cost to a custom column in the mysql table.
--
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Re: [asterisk-users] peer IP address in CDR
I'll try it out tomorrow. Youre my hero of the day! Regards, Remco Op 29 jun. 2010 om 17:45 heeft Zeeshan Zakaria zisha...@gmail.com het volgende geschreven: Just put exten = _pattern,s, before the MYSQL ... Zeeshan A Zakaria -- www.ilovetovoip.com On 2010-06-29 11:41 AM, Remco Bressers rbress...@signet.nl wrote: Thanks Zeeshan, but i don't use (and understand) AEL :) Any regular examples out there? :) regards, Remco On 06/29/2010 05:27 PM, Zeeshan Zakaria wrote: Let me make it simple for you: Add a column ... list-aster...@skycomuk.com mailto:list-aster...@skycomuk.com wrote: Remco Bressers wr... -- _ -- Bandwidth and Colocation Pr... -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] G729 license key registration
On 06/25/2010 09:48 AM, Kiss András wrote: You selected 5, G.729 Codec Please enter your Key-ID: G729-10D2X----X This product key cannot be registered! Please verify you entered the correct product key. Server response: 404 - Key not found. Any suggestions? How about contacting Digium about this? -- Met vriendelijke groet, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbress...@signet.nl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] UDPTL T38 via NAT
Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. On the LAN PBX, i've got the following config : [general] t38pt_udptl=yes [202] type=friend secret=*** username=202 regexten=202 host=dynamic canreinvite=yes allow=alaw context=local qualify=yes On the WAN PBX, the config for the trunk is the following : [general] t38pt_udptl=yes [trunk] type=peer context=trunk-in host=62.180.xxx.xxx port=5070 disallow=all allow=alaw allow=ulaw qualify=yes nat=no Can anybody tell me how to change this behaviour? Fax isn't working ofcourse. -- Kind regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbress...@signet.nl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UDPTL T38 via NAT
On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. Did you try t38pt_usertpsource=yes ? Hi, Yes, i tried adding that to the SIP peer configuration for the FAX ATA. Should i put it on the PBX trunk configuration also?? Remco -- Met vriendelijke groet, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbress...@signet.nl altijd online? www.signet.nl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UDPTL T38 via NAT
On 06/22/2010 04:35 PM, Johann Steinwendtner wrote: On 2010-06-22 15:16, Remco Bressers wrote: On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. Did you try t38pt_usertpsource=yes ? Hi, Yes, i tried adding that to the SIP peer configuration for the FAX ATA. Should i put it on the PBX trunk configuration also?? Remco Yes. This results in the very same problem : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 101, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 102, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 103, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) -- Kind regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbress...@signet.nl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] UDPTL T38 via NAT
On 06/22/2010 04:38 PM, marek cervenka wrote: On 06/22/2010 02:51 PM, Johann Steinwendtner wrote: On 2010-06-22 12:36, Remco Bressers wrote: Dear list, I've got the following setup : [FAX-ATA]--[PBX LAN]--[Firewall]--[PBX WAN]-[upstream SIP] On the PBX's we run Asterisk 1.4.33 with t38pt_udptl=yes in [general]. The FAX ATA is a Teles VoIPBox with T.38 support (that works). On the PBX WAN, i see the following in udptl debug : Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 184, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 185, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 32) Sent UDPTL packet to 172.16.0.156:4460 (type 0, seq 186, len 32) Got UDPTL packet from 62.180.xx.xx:36170 (type 0, seq 0, len 29) This means my outgoing udptl traffic is correctly translated, but somehow i'm sending 172.16.0.156 instead of my public IP address on the firewall. try asterisk 1.6.2.9 What would be the reason to do that? Is there any change on this in 1.6.2.9? -- Regards, Signet bv Remco Bressers T 040 - 707 4 907 F 040 - 707 4 909 E rbress...@signet.nl -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing storm-prevention behaviour in logger.conf
According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from ' sip:x...@xxx.xxx.xxx.xxx' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 times This IS a good thing to do, but i want to disable this behaviour. We are using fail2ban to ban scripts and people from the Asterisk system. On version 1.4.23 this worked fine, but now this mechanism is in place, i cannot use fail2ban anymore. Is there any option to disable this behaviour, or even better, add it to logger.conf so anybody can decide what to do? I just want all logging and it seems impossible now. Maybe a patch on the source? If you use a newer version of rsyslogd to do your logging, there is a global configuration directive: $RepeatedMsgReduction off that will do what you are asking. The issue #14905 patch you mention is not in 1.6.2.x. Hi, Well, this sounds fair, but this happened after an upgrade to 1.4.29 from 1.4.23. Nothing else changed in my setup after that. My logger.conf : [general] dateformat=%F %T [logfiles] console = notice,warning,error messages = notice,warning,error This tells me i'm not using the syslog feature at all and /var/log/asterisk/messages is generated by Asterisk and not by syslogd Second, I just downloaded 1.4.29. The patch that does the message repeated stuff is just not there, as Tilghman said. Is it possible that someone applied that patch to your source? Have you tried downloading the 1.4.29 tarball again and recompiling? If you installed asterisk as a package from somebody's repo, I can't really say, but it seems highly unlikely that the patch would be present. I hope this helps a little bit. Alright, i figured it out. I use the FreeBSD port of Asterisk and there's a file named patch-suppress_log_dups.diff in the files directory which patches logger.c. I guess i should convince the port maintainer that this is a nice patch, but not how Asterisk is supposed to work. It only raises eyebrows if they add extra features. Thanks a million! Remco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Changing storm-prevention behaviour in logger.conf
On Apr 18, 2010, at 12:40 AM, Barry Miller wrote: On Sat, Apr 17, 2010 at 11:14:23PM +0200, Remco Bressers wrote: Dear List, According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from ' sip:x...@xxx.xxx.xxx.xxx' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 times This IS a good thing to do, but i want to disable this behaviour. We are using fail2ban to ban scripts and people from the Asterisk system. On version 1.4.23 this worked fine, but now this mechanism is in place, i cannot use fail2ban anymore. Is there any option to disable this behaviour, or even better, add it to logger.conf so anybody can decide what to do? I just want all logging and it seems impossible now. Maybe a patch on the source? If you use a newer version of rsyslogd to do your logging, there is a global configuration directive: $RepeatedMsgReduction off that will do what you are asking. The issue #14905 patch you mention is not in 1.6.2.x. Hi, Well, this sounds fair, but this happened after an upgrade to 1.4.29 from 1.4.23. Nothing else changed in my setup after that. My logger.conf : [general] dateformat=%F %T [logfiles] console = notice,warning,error messages = notice,warning,error This tells me i'm not using the syslog feature at all and /var/log/asterisk/messages is generated by Asterisk and not by syslogd Please help. Regards, Remco -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Changing storm-prevention behaviour in logger.conf
Dear List, According to https://issues.asterisk.org/view.php?id=14905 there is a storm prevention mechanism in newer Asterisks. If i look in my logfile, i see : [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Registration from ' sip:x...@xxx.xxx.xxx.xxx' failed for 'xx.xx.xx.xx' - Wrong password [2010-04-17 15:12:01] NOTICE[1190] chan_sip.c: Last message repeated 3 times This IS a good thing to do, but i want to disable this behaviour. We are using fail2ban to ban scripts and people from the Asterisk system. On version 1.4.23 this worked fine, but now this mechanism is in place, i cannot use fail2ban anymore. Is there any option to disable this behaviour, or even better, add it to logger.conf so anybody can decide what to do? I just want all logging and it seems impossible now. Maybe a patch on the source? Regards, Remco Bressers Signet B.V. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
