[asterisk-users] Conference Calling
Hey All, I want to implement a conference calling scenario. Conference Call Procedure:User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room!How I can implement this scenario. What are generic steps to do so! Thanks=Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference Calling
Here is where to get you start with this. http://www.voip-info.org/wiki/view/Asterisk+n-way+call+HOWTO -Tri From: Faheem faheem_...@yahoo.com To: asterisk-users@lists.digium.com Sent: Sat, February 27, 2010 12:08:24 PM Subject: [asterisk-users] Conference Calling Hey All, I want to implement a conference calling scenario. Conference Call Procedure: User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room! How I can implement this scenario. What are generic steps to do so! Thanks = Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Conference Calling
Muhammad It is not really your scenario but the scenario to setup a conference call with three numbers could be to generate two call files that points to a local channel/a context/extension that route the leg into the conference room and have your own leg routed into the conference room after the input is done This not the solution but one of the many possible. enter the numbers for setting up the conference call like number1*number2 (check Read() cmd for storing input into a variable) split the input in seperated numbers See http://www.voip-info.org/wiki/index.php?page=Asterisk+variables generate the call files for setting up the connection. Point to a context, extension, priority to route the lef into a conference room. See http://www.voip-info.org/tiki-index.php?page=Asterisk+auto-dial+out move the call files to /var/spool/asterisk/outgoing (check System() cmd ) have your own leg routed into the conference room (check Goto() cmd ) Have a nice chat with the three of you ;-) Erik On 27 feb 2010, at 21:08, Faheem wrote: Hey All, I want to implement a conference calling scenario. Conference Call Procedure: User1 dial the User2. When call is connected put the current call on Hold and dial User3. When the call is connected between User1 and User3 join the User2 in a conference room! How I can implement this scenario. What are generic steps to do so! Thanks = Muhammad Faheem -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] conference calling
Greetings listers. I'm running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a 2-4 second delay before the callee can hear me. 3. When I call an external conference and connect, the others cannot hear me. Zapata.conf [trunkgroups] [channels] ;context=from-zaptel ;context=line1 busydetect=yes callprogress=yes busycount=4 hanguponpolarityswitch=yes answeronpolarityswitch=yes usecallingpres=yes priindication=outofband pritimer=t305,5 signalling=fxs_ks wink=50 useincomingcalleridonzaptransfer=yes echocancel=yes echocancelwhenbridged=yes faxdetect=yes rxgain=1.0 txgain=21.0 callgroup=1 group=1 usecallerid=yes callerid=asreceived cidstart=ring hidecallerid=no immediate=no pickupgroup=1 ;context=incoming channel = 1-4 Sip.conf [general] srvlookup=yes ;allows DNS lookups of server names naxexpirey=180 defaultexpirey=160 context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) tos_sip=cs3 tos_audio=ef ; bindport is the local UDP port that Asterisk will ; listen on bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls limitonpeers=yes notifyringing=yes rtupdate=yes[authentication] [104] type=peer context=phones host=dynamic fromuser=104 secret=xx canreinvite=update directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=90 session-minse=120 session-refresher=uac register = 104:xx...@xx.com/104 defaultip=192.168.xx.xxx mailbox=104 disallow=all allow=ulaw,alaw artcachefriends=yes notifyhold=yes incominglimit=1 call-limit=3 Other information will be provided as asked for. Thanks in advance for any help you can provide. Danny Nicholas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] conference calling
Turn off callprogres=yes or have it configured properly. It should fix your problem. regards Martin On Fri, Apr 3, 2009 at 2:42 PM, Danny Nicholas da...@debsinc.com wrote: Greetings listers. I’m running asterisk 1.4.21.2 on SUSE 11.0 using Polycom 501 phones. My outgoing connections are Zapata using a TDM401P. For the most part I can make and receive calls fine except for these 3 issues: 1. When I call an external conference, the call never bridges and hangs up after 60-90 seconds. 2. When I call another number there is a 2-4 second delay before the callee can hear me. 3. When I call an external conference and connect, the others cannot hear me. Zapata.conf [trunkgroups] [channels] ;context=from-zaptel ;context=line1 busydetect=yes callprogress=yes busycount=4 hanguponpolarityswitch=yes answeronpolarityswitch=yes usecallingpres=yes priindication=outofband pritimer=t305,5 signalling=fxs_ks wink=50 useincomingcalleridonzaptransfer=yes echocancel=yes echocancelwhenbridged=yes faxdetect=yes rxgain=1.0 txgain=21.0 callgroup=1 group=1 usecallerid=yes callerid=asreceived cidstart=ring hidecallerid=no immediate=no pickupgroup=1 ;context=incoming channel = 1-4 Sip.conf [general] srvlookup=yes ;allows DNS lookups of server names naxexpirey=180 defaultexpirey=160 context=default ; Default context for incoming calls allowoverlap=no ; Disable overlap dialing support. (Default is yes) bindport=5060 ; UDP Port to bind to (SIP standard port is 5060) tos_sip=cs3 tos_audio=ef ; bindport is the local UDP port that Asterisk will ; listen on bindaddr=192.168.xx.xx ; IP address to bind to (0.0.0.0 binds to all) srvlookup=yes ; Enable DNS SRV lookups on outbound calls limitonpeers=yes notifyringing=yes rtupdate=yes[authentication] [104] type=peer context=phones host=dynamic fromuser=104 secret=xx canreinvite=update directrtpsetup=no call-limit=3 nat=yes qualify=yes register=no session-timers=accept session-expires=90 session-minse=120 session-refresher=uac register = 104:xx...@xx.com/104 defaultip=192.168.xx.xxx mailbox=104 disallow=all allow=ulaw,alaw artcachefriends=yes notifyhold=yes incominglimit=1 call-limit=3 Other information will be provided as asked for. Thanks in advance for any help you can provide. Danny Nicholas ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
