[asterisk-users] Cut offs on outgoing SIP calls

[Asghar Mohammad]

sip set debug peer 90102 and check in log why call drop or upload log
somewhere. configuration seems ok.


On Wed, May 15, 2013 at 10:09 PM, Daniel - Asterisk <earohuanca at gmail.com>wrote:

> Current configuration follows:
>
> [general]
> context=default
> allowguest=no
> alwaysauthreject=yes
> allowoverlap=yes
> allowtransfer=yes
> tcpenable=no
> tlsenable=no
> srvlookup=yes
> vmexten=vm
> rtcachefriends=yes
> nat=no
> directmedia=nonat
> directrtpsetup=no
> videosupport=yes
> maxcallbitrate=384
> disallow=all
> allow=ulaw
> allow=alaw
> allow=g729
> allow=gsm
> allow=ilbc
> allow=speex
> allow=g726
> allow=g723
> mohinterpret=default
> mohsuggest=default
> dtmfmode=rfc2833
> timer1b=60000
> transport=udp
>
> [carrier-1]
> host=a.b.c.d
> type=friend
> context=from-pstn
> disallow=all
> allow=ulaw,alaw
> qualify=yes
> trunk=yes
>
> [90102]
> secret=xxxxxx
> mailbox=90102 at default
> cid_number=NXXXXXXXXX
> accountcode=401
> type=friend
> host=dynamic
> port=5060
> qualify=yes
> nat=yes
> transport=udp
> context=users
> disallow=all
> allow=ulaw,alaw,g729,gsm,speex,ilbc,h264,h263p,h263
> directmedia=no
> canreinvite=no
> videosupport=no
>
>
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