autodestruct on dialog

[davesdatasystems]

I think i have a dialplan that is keeping my calls open and allowing a lot of calls to go through the dialer

__sip_autodestruct: Autodestruct on dialog

and i am being charged by the voip provider for calls that only show they was in VDAD, some of these calls are over 50 minutes long. 50.017 minutes to be exact.

I figured it was a asterisk error so i completely rebuilt the dialer 2 times, but i see the same error and charges from the voip provider. But i i am 96% sure it is the dialplan now.

I think this dial plan worked fine on 4.0 and 5.0 but maybe not so much on 6.0. But anyway, here is the dialplan and i hope someone can help

Same hardware as listed in sigutare and the vicidial version i used is;
Vicibox_v.6.0.x86_64-6.0.3.preload.iso

I seen other posts on this subject, but they dont seem to help me any because i think it is in my dialplan. The dial plan entry was sent to me by xcast themselves, it looks a bit strange to me, but it makes calls, but i am worried about the account entry.

Dial plan entry is
[Xcast]
insecure=port,invite
secret=no
qualify=yes
host=38.102.250.50
disallow=all
allow=ulaw,alaw,g729
dtmfmode=rfc2833
nat=no
canreinvite=no
context=from-internal
type=peer
context=trunkinbound
host=38.102.250.60


Account entry is;

exten => _81XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81XXXXXXXXXX,2,Dial(SIP/Xcast/${EXTEN:1},,To)
exten => _81XXXXXXXXXX,3,Hangup





I also seen this issue can be in the sip file; SO here it is

allowoverlap=no ; Disable overlap dialing support. (Default is yes)
;allowtransfer=no ; Disable all transfers (unless enabled in peers or users)
;realm=mydomain.tld ; Realm for digest authentication
bindport=5060 ; UDP Port to bind to (SIP standard port is 5060)
bindaddr=0.0.0.0 ; IP address to bind to (0.0.0.0 binds to all)
srvlookup=yes ; Enable DNS SRV lookups on outbound calls
;domain=mydomain.tld ; Set default domain for this host
;pedantic=yes ; Enable checking of tags in headers,
;tos_sip=cs3 ; Sets TOS for SIP packets.
;tos_audio=ef ; Sets TOS for RTP audio packets.
;tos_video=af41 ; Sets TOS for RTP video packets.
;maxexpiry=3600 ; Maximum allowed time of incoming registrations
;minexpiry=60 ; Minimum length of registrations/subscriptions (default 60)
;defaultexpiry=120 ; Default length of incoming/outgoing registration
;t1min=100 ; Minimum roundtrip time for messages to monitored hosts
;notifymimetype=text/plain ; Allow overriding of mime type in MWI NOTIFY
;checkmwi=10 ; Default time between mailbox checks for peers
;buggymwi=no ; Cisco SIP firmware doesn't support the MWI RFC
;vmexten=voicemail ; dialplan extension to reach mailbox sets the
disallow=all ; First disallow all codecs
allow=ulaw ; Allow codecs in order of preference
allow=gsm
mohinterpret=default
mohsuggest=default
language=en ; Default language setting for all users/peers
relaxdtmf=yes ; Relax dtmf handling
trustrpid = no ; If Remote-Party-ID should be trusted
sendrpid = yes ; If Remote-Party-ID should be sent
progressinband=no ; If we should generate in-band ringing always
;useragent=Asterisk PBX ; Allows you to change the user agent string
;promiscredir = no ; If yes, allows 302 or REDIR to non-local SIP address
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains
dtmfmode = rfc2833 ; Set default dtmfmode for sending DTMF. Default: rfc2833
;compactheaders = yes ; send compact sip headers.
videosupport=no ; Turn on support for SIP video. You need to turn this on
;maxcallbitrate=384 ; Maximum bitrate for video calls (default 384 kb/s)
callevents=yes ; generate manager events when sip ua
;alwaysauthreject = yes ; When an incoming INVITE or REGISTER is to be rejected,
;g726nonstandard = yes ; If the peer negotiates G726-32 audio, use AAL2 packing
;matchexterniplocally = yes ; Only substitute the externip or externhost setting if it matches
;regcontext=sipregistrations
rtptimeout=60 ; Terminate call if 60 seconds of no RTP or RTCP activity
;rtpholdtimeout=300 ; Terminate call if 300 seconds of no RTP or RTCP activity
;rtpkeepalive=<secs> ; Send keepalives in the RTP stream to keep NAT open
;sipdebug = yes ; Turn on SIP debugging by default, from
;recordhistory=yes ; Record SIP history by default
;dumphistory=yes ; Dump SIP history at end of SIP dialogue
;allowsubscribe=no ; Disable support for subscriptions. (Default is yes)
;subscribecontext = default ; Set a specific context for SUBSCRIBE requests
notifyringing = yes ; Notify subscriptions on RINGING state (default: no)
notifyhold = yes ; Notify subscriptions on HOLD state (default: no)
limitonpeers = yes ; Apply call limits on peers only. This will improve
;t38pt_udptl = yes ; Default false
;register => 1234:password@mysipprovider.com
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
externip = XXX.XXX.XXX.XXX ; Address that we're going to put in outbound SIP (externip is correct, it is just removed for saftey reasons here)
;externhost=test.test.com ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=192.168.0.0/255.255.0.0; All RFC 1918 addresses are local networks
localnet=10.0.0.0/255.0.0.0 ; Also RFC1918
localnet=172.16.0.0/12 ; Another RFC1918 with CIDR notation
localnet=169.254.0.0/255.255.0.0 ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=yes ; Enable the new experimental direct RTP setup. This sets up
;rtcachefriends=yes ; Cache realtime friends by adding them to the internal list
;rtsavesysname=yes ; Save systemname in realtime database at registration
;rtupdate=yes ; Send registry updates to database using realtime? (yes|no)
;rtautoclear=yes ; Auto-Expire friends created on the fly on the same schedule
;ignoreregexpire=yes ; Enabling this setting has two functions:
;domain=mydomain.tld,mydomain-incoming
;domain=1.2.3.4 ; Add IP address as local domain
;allowexternaldomains=no ; Disable INVITE and REFER to non-local domains
;autodomain=yes ; Turn this on to have Asterisk add local host
;fromdomain=mydomain.tld ; When making outbound SIP INVITEs to
jbenable = yes ; Enables the use of a jitterbuffer on the receiving side of a
jbforce = no ; Forces the use of a jitterbuffer on the receive side of a SIP
jbmaxsize = 100 ; Max length of the jitterbuffer in milliseconds.
jbresyncthreshold = 1000 ; Jump in the frame timestamps over which the jitterbuffer is
jbimpl = fixed ; Jitterbuffer implementation, used on the receiving side of a SIP
jblog = no ; Enables jitterbuffer frame logging. Defaults to "no".
qualify=yes ; By default, qualify all peers at 2000ms
limitonpeer = yes ; enable call limit on a per peer basis, different from limitonpeers

#include sip-vicidial.conf

; register SIP account on remote machine if using SIP trunks
; register => testSIPtrunk:test@10.10.10.16:5060
;
; setup account for SIP trunking:
; [SIPtrunk]
; disallow=all
; allow=ulaw
; allow=alaw
; type=friend
; username=testSIPtrunk
; secret=test
; host=10.10.10.16
; dtmfmode=inband
; qualify=1000

[williamconley]

Oddly enough, we had a notation of this happening on a client server, but the solution was merely killing the asterisk process and restarting it.

The technician believes that it was actually backup and possibly other scripts running in the system that caused a backlog of some sort. The system was still obnoxious for another 15 minutes before it suddenly began working again. He did not think (at the time) to check and see if the optimization or other processes were running on the client server. And of course the client did not want to pay any more once it was "up and happy" so he logged out and never looked back.

He also noticed that there were a lot of "REGISTER" entries stuck in the "sip show channels" output.

There were a lot of these in the asterisk cli, generating new ones regularly even without calls in the system:

Code: Select all

[Aug 14 01:32:40] WARNING[3180]: chan_sip.c:4058 __sip_autodestruct: Autodestruct on dialog '6b64253331365a8a31e7f365508f5a8a@xx.xx.xx.xx:5060' with owner SIP/xxxxxx-00000000 in place (Method: BYE). Rescheduling destruction for 10000 ms


And a dozen of these in "sip show channels":

Code: Select all

xx.xx.xx.xx    (None)           27e67bfd-af5026  0x0 (nothing)    No       Rx: REGISTER               <guest>

All the IPs crossed to valid call center IPs for the server in question. Firewall UP.

[davesdatasystems]

I just wanted to do a post back and tell you guys that i was able to resolve this issue by putting () after hangup on the dial plan

Old dial plan
exten => _81XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81XXXXXXXXXX,2,Dial(SIP/Xcast/${EXTEN:1},,To)
exten => _81XXXXXXXXXX,3,Hangup


new dial plan
exten => _81XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _81XXXXXXXXXX,2,Dial(SIP/Xcast/${EXTEN:1},,To)
exten => _81XXXXXXXXXX,3,Hangup()

This seems to be a issue i see a lot of people having, was not a issue in asterisk 1.4 but in asterisk 1.8 it seems to need () to end the calls.

[williamconley]

That's a darn good postback.

[davesdatasystems]

well it slowed the issue down but did not stop it, so i added another post with more information to see i can get some help on this.

viewtopic.php?f=4&t=35354 comment on this post about any help you may be able to give.