similar to: Capture Media IP in CDR (CDR)

I am not proxying the media, but never the less I am forced to store the source media IP in my CDR, for regulatory reasons. Asterisk gets that information when the reinvite comes, but how do I store it? If I don't figure this out my next email will be from Federal Prison. Kindly help me stay away from those guys. Eventually we all need to save that information or we shall not be able to stay

Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR

I'm running 1.6.2.13 and need to record a small number of custom values use cdr_odbc and cdr_adaptive_odbc, and only the custom fields. The good news is that the custom records are being stored in the database as desired. The bad news is that I get three sets of warnings/notice about 'SQL Exec Direct failed' and dropping then reconnecting the database handle. I traced the SQL calls

Hi, We have a Kamailio and Asterisk cluster, both machines being on a real 103.x IP address and also on a 172.x OpenVPN address. The problem is that when Kamailo receives a call from the VPN and forwards it to the Asterisk server on it's 103.x address, Asterisk never sees the call. If Kamailio receives a call from the VPN and forwards the call to the Asterisk server on it's 172.x

How can we set the CDR Master file to rollover at say 30 Meg and create a new one -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100827/2e98385f/attachment.htm

Hi all, I want to upgared from asterisk-1.6.2.6 version to asterisk-1.8.0 version. When i execute "make" command for compilation i have seen below errors. In file included from /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/cdr.h:31 /usr/src/asterisk-1.8/asterisk-1.8.0/include/asterisk/data.h:233: error: field ?AST_DATA_IPADDR? has incomplete type

Hi, While compiling 1.8.8.1, I met the following error: [AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o

I want to capture all SIP messages. I have about 30 hosts in about 6 colos. My first thought was dumpcap, but the output file name format bugs me. What do you use for long term SIP capture? -- Thanks in advance, ------------------------------------------------------------------------- Steve Edwards sedwards at sedwards.com Voice: +1-760-468-3867 PST

Digium is 100% lost in the map. If they would come up with a Paid version of Asterisk, one that would use the .NET framework in Windows, something simple to install, they could go public on the product. Linux has a very steep learning curve. A Windows application that would do exactly the same would be a home run. Note: I am a Linux expert user, but it took me years to get here. And still, moving

I beg to differ. Digium is hiding from the real world and somebody is going take the software and run with it. My customers lost in excess of $50.000 and cut my pay in half, because of hackers. The hackers figured out how to scan every asterisk for weak passwords or open ports, and bang them real good. We need two things: a) disable in sip.conf the reply for INVITES that have wrong user

I'm playing around in a lab, and I was wondering if its possible to have Asterisk act similar to that of a Avaya PBX, where we have media gateways do the heavy lifting. This is what I was thinking of trying. 1. One asterisk server will contain the logic of the phone system (ex: queues, extensions...etc). 2. The mains server will not handle RTP traffic, it will send the RTP traffic to

Asterisk 11.1 Is it possible to catch the fact that an IP phone has died in the middle of a call and do something with it in the dialplan? Background: we run a small call center. Our agents sit in two groups, with their IP phones running from 2 different switches. Every once in a while the power on one side of the room will go out, or one of the switches will die, or one of the agents will

2013/3/29 Julian Lyndon-Smith <asterisk at dotr.com> > check out the endbeforehexten option in cdr.conf > > this needs to set to "yes" > > Julian > Unfortunately, this doesn't help. Let's drop the hangup handler at the moment, and focus on the "saving to file" part. Then my issue is I can't update CDR value is hangup exten. Here is a

hi,all for a long time, i cant understand the difference between ${CDR(calldate)} and ${CDR(start)} , ${CDR(answer)} i know ${CDR(start)} mean when a call is start. and ${CDR(answer)} means when a call was pick up. but what's ${CDR(calldate)} mean? Could you help me ? Thansk a lot! -- Thanks for your supporting, have a nice day. Sucan

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.7. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.7 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

To be removed from this mailing list, simply hit reply and type REMOVE in the subject line. Thank you, T3 Communications. ********************************************************* Blank CD-Recordable media special. 650Mb, 74 min. CD-R blanks in jewel case. ONLY $2.79 Each with minimum order of 50 pieces. Shipping and handling not included. Major brands in stock. For more information on

Hi, The Asterisk Wiki (page: http://www.voip-info.org/wiki/view/Asterisk+func+cdr) mentions I can set any custom CDR field I want. Here is the example it gives: ; Update our accountcode field and then save some random music facts too exten => s,1,Set(CDR(accountcode)=8675309) exten => s,2,Set(CDR(MyFavoriteBand)=Foo Fighters) exten => s,3,Set(CDR(MyFavoriteSong)=Hero) I am

Hello, I'm using Asterisk with Realtime extensions and ODBC CDR. And I'm have some trouble with the CDR userfield that is not changed when using the SET command in the realtime dialplan. In my dialplan (extensions.conf, the file) I'm setting the userfield like this : exten => s,n,Set(CDR(userfield)="X") Later, my dialplan switches to the realtime part and this is an

Hi, I have the following code that operates when a channel is hung-up: [record-hangupcause]exten => 1,n,Set(CDR(hangupcause)=${HANGUPCAUSE})exten => s,n,Return() Before the dial a hangup handler is registered: Set(CHANNEL(hangup_handler_push)=record-hangupcause,s,1) The routine is called and the variables are being set, however not on the channel's CDR which made the call. I believe this