Problemwith Outbound calls

When I make outgoing calls, the following message appears:

[Jun 13 14:16:43] WARNING[13159]: app_dial.c:2345 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[Jun 13 14:16:43] == Everyone is busy/congested at this time (1:0/0/1)
[Jun 13 14:16:43] – Executing [15143235708@default:3] Hangup(“SIP/8002-00000048”, “”) in new stack
[Jun 13 14:16:43] == Spawn extension (default, 15143235708, 3) exited non-zero on ‘SIP/8002-00000048’
[Jun 13 14:16:43] – Executing [h@default:1] AGI(“SIP/8002-00000048”, “agi://–HVcauses–PRI-----NODEBUG-----20-----CHANUNAVAIL----------”) in new stack
[Jun 13 14:16:43] – <SIP/8002-00000048>AGI Script agi://–HVcauses–PRI-----NODEBUG-----20-----CHANUNAVAIL---------- completed, returning 0

The destination is a host=dynamic device that isn’t registered, or any device that is failing to respond to the OPTIONs request from the qualify option.


Thank you for your answer, now I have no problem with incoming calls, but as I said, I can not make outgoing calls.
I share with you the files sip.conf and sip-vicidial.conf.
Please let me know if I have configuration problems.

context=default ; Default context for incoming calls
allowguest=yes ; Allow or reject guest calls (default is yes)
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= ; IP address to bind to ( 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
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
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= ; 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 =>
;registertimeout=20 ; retry registration calls every 20 seconds (default)
;registerattempts=10 ; Number of registration attempts before we give up
;externip = ; Address that we’re going to put in outbound SIP
; ; Alternatively you can specify a domain
;externrefresh=10 ; How often to refresh externhost if
localnet=; All RFC 1918 addresses are local networks
localnet= ; Also RFC1918
localnet= ; Another RFC1918 with CIDR notation
localnet= ;Zero conf local network
nat=yes ; Global NAT settings (Affects all peers and users)
canreinvite=no ; Asterisk by default tries to redirect the
;directrtpsetup=no ; 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= ; 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

You haven’t defined any devices!!!

What are you dialling. What is the very first relevant message?

(I wonder if you are trying to dial directly by URI, but have a punctuation error in the URI, so it is being treated as a device name (but you have no defined devices).

You also have an number of obsolete or deprecated options. Please read and understand the documentation and correct these.

The only allow that has any real effect is allow all, as it cancels out the disallow and includes the specific allows.

Using a jitter buffer is not normal for SIP; that is the job of the phone or circuit switched network.

allowguest is sometimes needed, but otherwise inadvisable. Obviously, with out it, you need to define devices.

chan_sip is not recommended for new designs.

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Where do I have to define the devices?

In sip.conf, or a file included from it.

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Thank you ,

I just added the devices
But still the call does not pass and still displays waiting for ring


[Jun 16 16:05:28] – <Local/8600052@default-0000000f;2>AGI Script agi://127. 0.0.1:4577/call_log–HVcauses–PRI-----NODEBUG-----0--------------- completed, r eturning 0
[Jun 16 16:05:29] == Manager ‘sendcron’ logged on from
[Jun 16 16:05:29] – Executing [58600052@default:1] MeetMe(“Local/58600052@d efault-00000010;2”, “8600052,Fmq”) in new stack
[Jun 16 16:05:29] > Channel Local/58600052@default-00000010;1 was answere d.
[Jun 16 16:05:29] – Executing [8309@default:1] Answer(“Local/58600052@defau lt-00000010;1”, “”) in new stack
[Jun 16 16:05:29] – Executing [8309@default:2] Monitor(“Local/58600052@defa ult-00000010;1”, “wav,20170616-160527_5147488747_12322642_agent001”) in new stac k
[Jun 16 16:05:29] == Manager ‘sendcron’ logged off from
[Jun 16 16:05:29] – Executing [8309@default:3] Wait(“Local/58600052@default -00000010;1”, “3600”) in new stack

The call you have logged looks nothing like a typical outbound call and will end up sitting there doing nothing for an hour.

Idid it from manual_dial in goautodial interface, and when i user the directly the device, i have this in the Cli:

[Jun 16 16:41:20] == Using SIP RTP CoS mark 5
[Jun 16 16:41:20] – Executing [15147488747@default:1] AGI(“SIP/8001-0000000f”, “agi://”) in new stack
[Jun 16 16:41:20] – <SIP/8001-0000000f>AGI Script agi:// completed, returning 0
[Jun 16 16:41:20] – Executing [15147488747@default:2] Dial(“SIP/8001-0000000f”, “SIP/15147488747@MAZAGANIN,tTo”) in new stack
[Jun 16 16:41:20] WARNING[1677]: app_dial.c:2345 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Subscriber absent)
[Jun 16 16:41:20] == Everyone is busy/congested at this time (1:0/0/1)
[Jun 16 16:41:20] – Executing [15147488747@default:3] Hangup(“SIP/8001-0000000f”, “”) in new stack
[Jun 16 16:41:20] == Spawn extension (default, 15147488747, 3) exited non-zero on ‘SIP/8001-0000000f’
[Jun 16 16:41:20] – Executing [h@default:1] AGI(“SIP/8001-0000000f”, “agi://–HVcauses–PRI-----NODEBUG-----20-----CHANUNAVAIL----------”) in new stack

We are not experts on go_autodial. You either need to present the problem in pure Asterisk terms, or get support from the go_autodial people.

Your latest log shows that MAZAGANIN has not registered or has failed a qualify test.

Ok thank you so mutch.