ok thanks for your response i have created an agent in sip
sip.conf [222] type=friend context=agents host=dynamic dtmfmode=auto disallow=all allow=alaw allow=ulaw qualify=yes context=test i have add in extensions.conf the fil below but when i check in *var/spool/asterisk/monitor there is no record call* ** *could you please chek these configuration and tell me if there is any issue or wrong * *tahnks a lot * extensions.conf [test] exten => 100,1,Answer() exten => 100,2,MixMonitor(test.wav|av(0)V(0)) exten => 100,3,Dial(SIP/222) exten => 100,4,Hangup() 2011/3/1 Fellipe ... <fellipe...@hotmail.com> > Hi, > > here is an example: > > http://www.asteriskguru.com/tutorials/mixmonitor.html > > Enjoy it! > > Best regards, > > Fellipe > > ------------------------------ > Date: Tue, 1 Mar 2011 17:06:32 +0000 > From: salah.elharit...@gmail.com > To: asterisk-users@lists.digium.com > Subject: Re: [asterisk-users] records inbound and outbound calls > > > thank you so much but i don't know how can i do > > could you please give an example to record an external call or which file I > must to configure > > > > Thanks a lot > > > 2011/3/1 Danny Nicholas <da...@debsinc.com> > > ------------------------------ > > *From:* asterisk-users-boun...@lists.digium.com [mailto: > asterisk-users-boun...@lists.digium.com] *On Behalf Of *salaheddine > elharit > *Sent:* Tuesday, March 01, 2011 10:35 AM > *To:* Asterisk Users Mailing List - Non-Commercial Discussion > *Subject:* [asterisk-users] records inbound and outbound calls > > > > Hello List > > > > i have asterisk installed in our call centre i have configured the snom > phone 320 and 370 with in sip.conf and dialplan.com and extenssion.com > > > > i have just one question how can i do in order to record all the calls > automatically in our server > > > > Thanks and regards > Just put a mixmonitor command after your Answer for incoming and add a > macro to your dial command to start mixmonitor when dialing out. > > -- > _____________________________________________________________________ > -- 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 > > -- > _____________________________________________________________________ > -- 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