[asterisk-users] conference calling

[Martin]

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 <danny at 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=t305,50000
>
> 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=xxxxxx
>
> 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:xxxxx at xxxxxx.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
>
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