similar to: Problem after upgrading from 1.2.21.1 to 1.2.22

After upgrading from 1.2.x to 1.4.x call detail records are not being written to /var/log/asterisk/cdr-csv/Master.csv In cdr_manager.conf I have [general] Enabled = yes Apparently there is something else that needs to be configured for call detail records in 1.4.x. Can someone point me in the right direction? Don Pobanz

1. RE: Start recording automatically when xferring to anextension? (Don Pobanz) Message: 1 Date: Fri, 25 May 2007 11:54:33 -0500 From: "Don Pobanz" <dpobanz@hastingsutilities.com> Subject: RE: [asterisk-users] Start recording automatically when xferring to anextension? To: "Asterisk Users Mailing List - Non-Commercial Discussion"

Modem connect speeds on calls through * seem to be lower than calls made through the telephone company lines or our old Rolm PBX. All data calls have 2 wire analog modems on both ends. For my set up I have channels of a Zhone channel bank tied to 2 modems. The Zhone channel bank interfaces my * server with a T400P card. modem --- Zhone Channel bank - * via T400P card - Zhone channel bank -

We have had a few different times when a user has forwarded their phone to himself. This has overloaded the communications to our operator panel (FOP). One user should not be able to effect the whole phone system! Is there a way that the number of times that a call can be forwarded could be limited like to 10 or even 100? Then even if a user does something stupid like forwarding their calls to

I recently purchased a Sipura 2000 and connected a phone to it which is connected to my asterisk box via sip. Calls to the Sipura 2000 work fine from another sip device connected through *, from either an fxo or fxs (via adtran channel bank connected to a T400P card) port. However, when a call comes in from the phone company over a T1 with em_w trunks, the phone on the Sipura will ring but I

There has been discussions about the voicemail menus and some of us would like to see an overall plan for the voicemail menus. There are 3 primary ways of arranging the menus. First is a tree structure, second is a random access structure and the third would be a hybrid of the two. (Comedian mail is currently a hybrid.) As was pointed out by Brad Bergman, the ideal would be to have it

When you have a bunch of analog phones that you want to connect to asterisk, but those analog phones have no transfer button, what are the options to allow the phones to transfer a call? -- ------------------------------------------------------------ Erick Perez Panama Sistemas Integradores de Telefonia IP y Soluciones Para Centros de Datos Panama, Republica de Panama Cel Panama. +(507)

Hi list, Sometimes I receive a Red Alarm in my E1 trunk (E&M immediate start signaling), and just few seconds after this, all alarms are cleared. This problem ocurrs many times/day, and if are calls in progress, these calls just hang-up. Could it be an asterisk bug? Or may I contact the PSTN provider? Thanks Eduardo

The Asterisk development team has released Asterisk version 1.2.21.1 and 1.4.7.1. These releases are minor updates to the releases that were made yesterday to fix a couple of introduced issues. One issue was related to the ODBC realtime driver. Another was related to music on hold. Thank you for your support!

The Asterisk development team has released Asterisk version 1.2.21.1 and 1.4.7.1. These releases are minor updates to the releases that were made yesterday to fix a couple of introduced issues. One issue was related to the ODBC realtime driver. Another was related to music on hold. Thank you for your support!

Hi guys, I posted a somewhat similar question about a month ago and got a thoughtful resonse from Steven Critchfield, but I've got a quick follow up question to it. I'm looking to setup a 16 extension / 10-14 phone line Asterisk install for a customer who would like to have DID numbers for the extensions, since they're currently on Centrex and already have the 1-to-1

I get the following error when trying to compile zaptel on CentOS 5 kernel 2.6.18-8.1.3.el5 CC [M] /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c: In function ? /root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.c:171: error: ? has no member named ? make[3]: *** [/root/asterisk-src/zaptel-1.2.17.1/xpp/xbus-core.o] Error 1 make[2]: ***

hello all, I still experience the random off-hook on my fxo cxhannels, i am using a zplex-10b channel bank. which does not allow me to call out. The situation still persists...this is what i have in the zapata.conf [channels] context=internal context=incoming context=default usecallerid=no usecallwaiting=no signalling=fxs_ks channel=1-8 signalling=fxo_ks channel=16-24 but i still have the the

I have a linejack and a phone jack in my asterisk server working well between the SIP phones and the phonejack. what I cannot get to work is the outbound linejack Phone/phone0 trunk line? how can I get a SIP or Phone/phone1 phonejack phone to dial 9 then outside number and pickup Phone/phone0 and dial it? right now it accepts a 95551212 but busy's on the last digit 2. no outside dial.

Hi I've upgraded my server to asterisk-1.2.22 from 1.2.10 after that my DeadAGI scripts are not working properly. Like after hangup I used to do some more work now its not working. Please help. thanks arun -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070722/09df42e9/attachment.htm

Is there a way to reduce the volume of the dial tone on a zap channel? I don't want to reduce the audio volume on calls so txgain in zapata.conf will not work. I am having problems with asterisk not recognizing the first dialed digit from an analog phone about 8-15% of the time. Once the dialtone goes away, the digits are always recognized. Any other thoughts on how to solve this are also

Is it possible to configure * so that if a caller reaches voicemail for someone in Engineering, but doesn't want to leave a message they can press zero (0) and reach the Engineering Secretary or if they are calling someone in Accounting and reach voicemail, pressing '0' would reach the Accounting secretary, not the Engineering secretary? Don Pobanz

I've released Xapian 1.2.22 (including Search::Xapian 1.2.22.0), which you can download from: http://xapian.org/download Since 1.2.16, release tarballs are compressed with xz instead of gzip, as it provides much better compression, without being too slow to decompress. (For now at least, Search::Xapian is still gzipped.) We decided not to provide multiple formats for the downloads. The

I did cvs update on asterisk, zaptel, libpri as of today (November 5, 2003). I also did 'make clean' on each of them. My previous version of asterisk was cvs of September 15, 2003. No other changes have been made to my system other that these updates. when running 'make asterisk' the following error appears term.c:55: conflicting types for `term_color'

Thanks anyone, I found the problem in rhconfig.h. After the fix I successfully compiled zaptel. V. --- Ron Wellsted <ron@wellsted.org.uk> wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > Have you built your kernel on that machine? > > The errors suggest that while the kernel sources are > installed, the > kernel has not been built. > > Check on