Incoming SIP calls

I successfully got Asterisk up to a point where I can make outgoing calls, but the incoming calls almost immediately hang up after barely completing a ring and the CLI log looks like below. I hope anyone can shed some light on my problems.

– Executing Goto(“SIP/my10digitnumber-ac21”, “inbound-from-BroadVoice|s|1”) in new stack
– Goto (inbound-from-BroadVoice,s,1)
– Executing Ringing(“SIP/my10digitnumber-ac21”, “”) in new stack
– Executing Answer(“SIP/my10digitnumber-ac21”, “”) in new stack
– Executing Dial(“SIP/my10digitnumber-ac21”, “(SIP/101|60||)”) in new stack
May 20 12:29:31 WARNING[14657]: channel.c:2536 ast_request: No channel type registered for '(SIP’
May 20 12:29:31 NOTICE[14657]: app_dial.c:1029 dial_exec_full: Unable to create channel of type ‘(SIP’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing VoiceMail(“SIP/my10digitnumber-ac21”, “101”) in new stack
May 20 12:29:31 WARNING[14657]: app_voicemail.c:2411 leave_voicemail: No entry in voicemail config file for ‘101’
– Executing Hangup(“SIP/my10digitnumber-ac21”, “”) in new stack
== Spawn extension (inbound-from-BroadVoice, s, 5) exited non-zero on ‘SIP/my10digitnumber-ac21’
– Executing Goto(“SIP/my10digitnumber-51b5”, “inbound-from-BroadVoice|s|1”) in new stack
– Goto (inbound-from-BroadVoice,s,1)
– Executing Ringing(“SIP/my10digitnumber-51b5”, “”) in new stack
– Executing Answer(“SIP/my10digitnumber-51b5”, “”) in new stack
– Executing Dial(“SIP/my10digitnumber-51b5”, “(SIP/101|60||)”) in new stack
May 20 12:29:32 WARNING[14661]: channel.c:2536 ast_request: No channel type registered for '(SIP’
May 20 12:29:32 NOTICE[14661]: app_dial.c:1029 dial_exec_full: Unable to create channel of type ‘(SIP’ (cause 66 - Channel not implemented)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing VoiceMail(“SIP/my10digitnumber-51b5”, “101”) in new stack
May 20 12:29:32 WARNING[14661]: app_voicemail.c:2411 leave_voicemail: No entry in voicemail config file for ‘101’
– Executing Hangup(“SIP/my10digitnumber-51b5”, “”) in new stack
== Spawn extension (inbound-from-BroadVoice, s, 5) exited non-zero on ‘SIP/my10digitnumber-51b5’

I noticed it mentions sip channel type, and cause 66 channel. How exactly do these have to be configured and what texts need to go in there if need be?

seems like a problem with your dial plan. what do you have in extensions.conf ?

I have resolved this error which was caused by a misplaced coma (on line: exten=>s,3,Dial(SIP/101|60||) ) so now when an incoming call is in progress it rings busy and displays:

– Executing Goto(“SIP/my7digitnumber-5d82”, “inbound-from-BroadVoice|s|1”) in new stack
– Goto (inbound-from-BroadVoice,s,1)
– Executing Ringing(“SIP/my7digitnumber-5d82”, “”) in new stack
– Executing Dial(“SIP/my7digitnumber-5d82”, “SIP/101|60||”) in new stack
– Called 101
– Got SIP response 486 “Busy” back from 192.168.1.5
– SIP/101-9707 is busy
== Everyone is busy/congested at this time (1:1/0/0)
== Auto fallthrough, channel ‘SIP/my7digitnumber-5d82’ status is ‘BUSY’
– Registered SIP ‘101’ at 192.168.1.5 port 5060 expires 180
– Executing Goto(“SIP/my7digitnumber-f2e5”, “inbound-from-BroadVoice|s|1”) in new stack
– Goto (inbound-from-BroadVoice,s,1)
– Executing Ringing(“SIP/my7digitnumber-f2e5”, “”) in new stack
– Executing Dial(“SIP/my7digitnumber-f2e5”, “SIP/101|60||”) in new stack
– Called 101
– Got SIP response 486 “Busy” back from 192.168.1.5
– SIP/101-f728 is busy
== Everyone is busy/congested at this time (1:1/0/0)
== Auto fallthrough, channel ‘SIP/my7digitnumber-f2e5’ status is ‘BUSY’

My extensions.conf is as below

;
; Static extension configuration file, used by
; the pbx_config module. This is where you configure all your
; inbound and outbound calls in Asterisk.
;
; This configuration file is reloaded
; - With the “extensions reload” command in the CLI
; - With the “reload” command (that reloads everything) in the CLI

;
; The “General” category is for certain variables.
;
[general]
;
; If static is set to no, or omitted, then the pbx_config will rewrite
; this file when extensions are modified. Remember that all comments
; made in the file will be lost when that happens.
;
; XXX Not yet implemented XXX
;
static=yes
;
; if static=yes and writeprotect=no, you can save dialplan by
; CLI command ‘save dialplan’ too
;
writeprotect=no
;
; If autofallthrough is set, then if an extension runs out of
; things to do, it will terminate the call with BUSY, CONGESTION
; or HANGUP depending on Asterisk’s best guess (strongly recommended).
;
; If autofallthrough is not set, then if an extension runs out of
; things to do, asterisk will wait for a new extension to be dialed
; (this is the original behavior of Asterisk 1.0 and earlier).
;
autofallthrough=yes
;
; If clearglobalvars is set, global variables will be cleared
; and reparsed on an extensions reload, or Asterisk reload.
;
; If clearglobalvars is not set, then global variables will persist
; through reloads, and even if deleted from the extensions.conf or
; one if its included files, will remain set to the previous value.
;
clearglobalvars=no
;
; If priorityjumping is set to ‘yes’, then applications that support
; ‘jumping’ to a different priority based on the result of their operations
; will do so (this is backwards compatible behavior with pre-1.2 releases
; of Asterisk). Individual applications can also be requested to do this
; by passing a ‘j’ option in their arguments.
;
priorityjumping=no
;
; You can include other config files, use the #include command
; (without the ‘;’). Note that this is different from the “include” command
; that includes contexts within other contexts. The #include command works
; in all asterisk configuration files.
;#include “filename.conf”

; The “Globals” category contains global variables that can be referenced
; in the dialplan with ${VARIABLE} or ${ENV(VARIABLE)} for Environmental
; variables,
; ${${VARIABLE}} or ${text${VARIABLE}} or any hybrid
;
[globals]

include=>inbound
include=>outbound
OUTBOUND=>SIP/101 ;This agent is able to place outgoing calls

;CONSOLE=Console/dsp ; Console interface for demo
;CONSOLE=Zap/1
;CONSOLE=Phone/phone0
;IAXINFO=guest ; IAXtel username/password
;IAXINFO=myuser:mypass
;TRUNK=Zap/g2 ; Trunk interface
;
; Note the ‘g2’ in the TRUNK variable above. It specifies which group (defined
; in zapata.conf) to dial, i.e. group 2, and how to choose a channel to use in
; the specified group. The four possible options are:
;
; g: select the lowest-numbered non-busy Zap channel
; (aka. ascending sequential hunt group).
; G: select the highest-numbered non-busy Zap channel
; (aka. descending sequential hunt group).
; r: use a round-robin search, starting at the next highest channel than last
; time (aka. ascending rotary hunt group).
; R: use a round-robin search, starting at the next lowest channel than last
; time (aka. descending rotary hunt group).
;
TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0)
;TRUNK=IAX2/user:pass@provider

;
; Any category other than “General” and “Globals” represent
; extension contexts, which are collections of extensions.
;
; Extension names may be numbers, letters, or combinations
; thereof. If an extension name is prefixed by a ‘_’
; character, it is interpreted as a pattern rather than a
; literal. In patterns, some characters have special meanings:
;
; X - any digit from 0-9
; Z - any digit from 1-9
; N - any digit from 2-9
; [1235-9] - any digit in the brackets (in this example, 1,2,3,5,6,7,8,9)
; . - wildcard, matches anything remaining (e.g. _9011. matches
; anything starting with 9011 excluding 9011 itself)
; ! - wildcard, causes the matching process to complete as soon as
; it can unambiguously determine that no other matches are possible
;
; For example the extension _NXXXXXX would match normal 7 digit dialings,
; while _1NXXNXXXXXX would represent an area code plus phone number
; preceeded by a one.
;
; Each step of an extension is ordered by priority, which must
; always start with 1 to be considered a valid extension. The priority
; “next” or “n” means the previous priority plus one, regardless of whether
; the previous priority was associated with the current extension or not.
; The priority “same” or “s” means the same as the previously specified
; priority, again regardless of whether the previous entry was for the
; same extension. Priorities may be immediately followed by a plus sign
; and another integer to add that amount (most useful with ‘s’ or ‘n’).
; Priorities may then also have an alias, or label, in
; parenthesis after their name which can be used in goto situations
;
; Contexts contain several lines, one for each step of each
; extension, which can take one of two forms as listed below,
; with the first form being preferred. One may include another
; context in the current one as well, optionally with a
; date and time. Included contexts are included in the order
; they are listed.
;
;[context]
;exten => someexten,priority[+offset][(alias)],application(arg1,arg2,…)
;exten => someexten,priority[+offset][(alias)],application,arg1|arg2…
;
; Timing list for includes is
;
; |||
;
;include => daytime|9:00-17:00|mon-fri||
;
; ignorepat can be used to instruct drivers to not cancel dialtone upon
; receipt of a particular pattern. The most commonly used example is
; of course ‘9’ like this:
;
;ignorepat => 9
;
; so that dialtone remains even after dialing a 9.
;

;
; Sample entries for extensions.conf
;
;
[dundi-e164-canonical]
;
; List canonical entries here
;
;exten => 12564286000,1,Macro(std-exten,6000,IAX2/foo)
;exten => _125642860XX,1,Dial(IAX2/otherbox/${EXTEN:7})

[dundi-e164-customers]
;
; If you are an ITSP or Reseller, list your customers here.
;
;exten => _12564286000,1,Dial(SIP/customer1)
;exten => _12564286001,1,Dial(IAX2/customer2)

[dundi-e164-via-pstn]
;
; If you are freely delivering calls to the PSTN, list them here
;
;exten => _1256428XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Expose all of 256-428
;exten => _1256325XXXX,1,Dial(Zap/g2/${EXTEN:7}) ; Ditto for 256-325

[dundi-e164-local]
;
; Context to put your dundi IAX2 or SIP user in for
; full access
;
include => dundi-e164-canonical
include => dundi-e164-customers
include => dundi-e164-via-pstn

[dundi-e164-switch]
;
; Just a wrapper for the switch
;
switch => DUNDi/e164

[dundi-e164-lookup]
;
; Locally to lookup, try looking for a local E.164 solution
; then try DUNDi if we don’t have one.
;
include => dundi-e164-local
include => dundi-e164-switch
;
; DUNDi can also be implemented as a Macro instead of using
; the Local channel driver.
;
[macro-dundi-e164]
;
; ARG1 is the extension to Dial
;
exten => s,1,Goto(${ARG1},1)
include => dundi-e164-lookup

;
; Here are the entries you need to participate in the IAXTEL
; call routing system. Most IAXTEL numbers begin with 1-700, but
; there are exceptions. For more information, and to sign
; up, please go to www.iaxtel.com
;
[iaxtel700]
exten => _91700XXXXXXX,1,Dial(IAX2/${IAXINFO}@iaxtel.com/${EXTEN:1}@iaxtel)

;
; The SWITCH statement permits a server to share the dialplain with
; another server. Use with care: Reciprocal switch statements are not
; allowed (e.g. both A -> B and B -> A), and the switched server needs
; to be on-line or else dialing can be severly delayed.
;
[iaxprovider]
;switch => IAX2/user:[key]@myserver/mycontext

[trunkint]
;
; International long distance through trunk
;
exten => _9011.,1,Macro(dundi-e164,${EXTEN:4})
exten => _9011.,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunkld]
;
; Long distance context accessed through trunk
;
exten => _91NXXNXXXXXX,1,Macro(dundi-e164,${EXTEN:1})
exten => _91NXXNXXXXXX,n,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunklocal]
;
; Local seven-digit dialing accessed through trunk interface
;
;exten => _9NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[trunktollfree]
;
; Long distance context accessed through trunk interface
;
;exten => _91800NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;exten => _91888NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;exten => _91877NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})
;exten => _91866NXXXXXX,1,Dial(${TRUNK}/${EXTEN:${TRUNKMSD}})

[international]
;
; Master context for international long distance
;
ignorepat => 9
include => longdistance
include => trunkint

[longdistance]
;
; Master context for long distance
;out-bv
ignorepat => 9
include => local
include => trunkld

[local]
;
; Master context for local, toll-free, and iaxtel calls only
;
;ignorepat => 9
;include => default
;include => parkedcalls
include => trunklocal
include => iaxtel700
include => trunktollfree
include => iaxprovider
;
; You can use an alternative switch type as well, to resolve
; extensions that are not known here, for example with remote
; IAX switching you transparently get access to the remote
; Asterisk PBX
;
; switch => IAX2/user:password@bigserver/local
;
; An “lswitch” is like a switch but is literal, in that
; variable substitution is not performed at load time
; but is passed to the switch directly (presumably to
; be substituted in the switch routine itself)
;
; lswitch => Loopback/12${EXTEN}@othercontext
;
; An “eswitch” is like a switch but the evaluation of
; variable substitution is performed at runtime before
; being passed to the switch routine.
;
; eswitch => IAX2/context@${CURSERVER}

[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
;
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start

exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start

exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

[macro-stdPrivacyexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
; ${ARG2} - Device(s) to ring
; ${ARG3} - Optional DONTCALL context name to jump to (assumes the s,1 extension-priority)
; ${ARG4} - Optional TORTURE context name to jump to (assumes the s,1 extension-priority)`
;
exten => s,1,Dial(${ARG2},20|p) ; Ring the interface, 20 seconds maximum, call screening option (or use P for databased call screening)
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)

;exten => s-NOANSWER,1,Voicemail(u${ARG1}) ; If unavailable, send to voicemail w/ ;unavail announce
;exten => s-NOANSWER,2,Goto(default,s,1) ; If they press #, return to start

;exten => s-BUSY,1,Voicemail(b${ARG1}) ; If busy, send to voicemail w/ busy announce
;exten => s-BUSY,2,Goto(default,s,1) ; If they press #, return to start

;exten => s-DONTCALL,1,Goto(${ARG3},s,1) ; Callee chose to send this call to a polite “Don’t call again” script.

exten => s-TORTURE,1,Goto(${ARG4},s,1) ; Callee chose to send this call to a telemarketer torture script.

;exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer

;exten => a,1,VoicemailMain(${ARG1}) ; If they press *, send the user into VoicemailMain

[demo]
;
; We start with what to do when a call first comes in.
;
;exten => s,1,Wait,1 ; Wait a second, just for fun
;exten => s,n,Answer ; Answer the line
;exten => s,n,Set(TIMEOUT(digit)=5) ; Set Digit Timeout to 5 seconds
;exten => s,n,Set(TIMEOUT(response)=10) ; Set Response Timeout to 10 seconds
;exten => s,n(restart),BackGround(demo-congrats) ; Play a congratulatory message
;exten => s,n(instruct),BackGround(demo-instruct) ; Play some instructions
;exten => s,n,WaitExten ; Wait for an extension to be dialed.

;exten => 2,1,BackGround(demo-moreinfo) ; Give some more information.
;exten => 2,n,Goto(s,instruct)

;exten => 3,1,Set(LANGUAGE()=fr) ; Set language to french
;exten => 3,n,Goto(s,restart) ; Start with the congratulations

;exten => 1000,1,Goto(default,s,1)
;
; We also create an example user, 1234, who is on the console and has
; voicemail, etc.
;
;exten => 1234,1,Playback(transfer,skip) ; “Please hold while…”
; (but skip if channel is not up)
;exten => 1234,n,Macro(stdexten,1234,${CONSOLE})

;exten => 1235,1,Voicemail(u1234) ; Right to voicemail

;exten => 1236,1,Dial(Console/dsp) ; Ring forever
;exten => 1236,n,Voicemail(u1234) ; Unless busy

;
; # for when they’re done with the demo
;
exten => #,1,Playback(demo-thanks) ; "Thanks for trying the demo"
exten => #,n,Hangup ; Hang them up.

;
; A timeout and “invalid extension rule”
;
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; “That’s not valid, try again”

;
; Create an extension, 500, for dialing the
; Asterisk demo.
;
exten => 500,1,Playback(demo-abouttotry); Let them know what’s going on
exten => 500,n,Dial(IAX2/guest@misery.digium.com/s@default) ; Call the Asterisk demo
exten => 500,n,Playback(demo-nogo) ; Couldn’t connect to the demo site
exten => 500,n,Goto(s,6) ; Return to the start over message.

;
; Create an extension, 600, for evaulating echo latency.
;
exten => 600,1,Playback(demo-echotest) ; Let them know what’s going on
exten => 600,n,Echo ; Do the echo test
exten => 600,n,Playback(demo-echodone) ; Let them know it’s over
exten => 600,n,Goto(s,6) ; Start over

;
; Give voicemail at extension 8500
;
exten => 8500,1,VoicemailMain
exten => 8500,n,Goto(s,6)
;
; Here’s what a phone entry would look like (IXJ for example)
;
;exten => 1265,1,Dial(Phone/phone0,15)
;exten => 1265,n,Goto(s,5)

;[mainmenu]
;
; Example “main menu” context with submenu
;
;exten => s,1,Answer
;exten => s,n,Background(thanks) ; “Thanks for calling press 1 for sales, 2 for support, …”
;exten => s,n,WaitExten
;exten => 1,1,Goto(submenu,s,1)
;exten => 2,1,Hangup
;include => default
;
;[submenu]
;exten => s,1,Ringing ; Make them comfortable with 2 seconds of ringback
;exten => s,n,Wait,2
;exten => s,n,Background(submenuopts) ; “Thanks for calling the sales department. Press 1 for steve, 2 for…”
;exten => s,n,WaitExten
;exten => 1,1,Goto(default,steve,1)
;exten => 2,1,Goto(default,mark,2)

[default]
;
; By default we include the demo. In a production system, you
; probably don’t want to have the demo there.
;
include => demo

;
; Extensions like the two below can be used for FWD, Nikotel, sipgate etc.
; Note that you must have a [sipprovider] section in sip.conf whereas
; the otherprovider.net example does not require such a peer definition
;
;exten => _41X.,1,Dial(SIP/${EXTEN:2}@sipprovider,r)
;exten => _42X.,1,Dial(SIP/user:passwd@${EXTEN:2}@otherprovider.net,30,rT)

; Real extensions would go here. Generally you want real extensions to be
; 4 or 5 digits long (although there is no such requirement) and start with a
; single digit that is fairly large (like 6 or 7) so that you have plenty of
; room to overlap extensions and menu options without conflict. You can alias
; them with names, too, and use global variables

;exten => 6245,hint,SIP/Grandstream1&SIP/Xlite1,Joe Schmoe ; Channel hints for presence
;exten => 6245,1,Dial(SIP/Grandstream1,20,rt) ; permit transfer
;exten => 6245,n(dial),Dial(${HINT},20,rtT) ; Use hint as listed
;exten => 6245,n,Voicemail(u6245) ; Voicemail (unavailable)
;exten => 6245,s+1,Hangup ; s+1, same as n
;exten => 6245,dial+101,Voicemail(b6245) ; Voicemail (busy)
;exten => 6361,1,Dial(IAX2/JaneDoe,rm) ; ring without time limit
;exten => 6389,1,Dial(MGCP/aaln/1@192.168.0.14)
;exten => 6394,1,Dial(Local/6275/n) ; this will dial ${MARK}

;exten => 6275,1,Macro(stdexten,6275,${MARK}) ; assuming ${MARK} is something like Zap/2
;exten => mark,1,Goto(6275|1) ; alias mark to 6275
;exten => 6536,1,Macro(stdexten,6236,${WIL}) ; Ditto for wil
;exten => wil,1,Goto(6236|1)
;
; Some other handy things are an extension for checking voicemail via
; voicemailmain
;
;exten => 8500,1,VoicemailMain
;exten => 8500,n,Hangup
;
; Or a conference room (you’ll need to edit meetme.conf to enable this room)
;
;exten => 8600,1,Meetme(1234)
;
; Or playing an announcement to the called party, as soon it answers
;
;exten = 8700,1,Dial(${MARK},30,A(/path/to/my/announcemsg))
;
; For more information on applications, just type “show applications” at your
; friendly Asterisk CLI prompt.
;
; 'show application ’ will show details of how you
; use that particular application in this file, the dial plan.
;

;Then we have to tell Asterisk what to do with the call when it comes in by adding the following

[sip.broadvoice.com] ;This is going to be your inbound trunk, telling all incoming calls ;where ;to go

exten=>my10digitnumber,1,Goto(inbound-from-BroadVoice|s|1)
exten=>my7digitnumber,1,Goto(inbound-from-BroadVoice|s|1)

[inbound-from-BroadVoice]
exten=>s,1,Ringing
exten=>s,2,Answer
exten=>s,3,Dial(SIP/101|60||) ;Rings your phone for 60 seconds then goes to voicemail.
exten=>s,4,Voicemail,101
exten=>s,5,Hangup

[global]

exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@out-bv,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()
exten=>*86,1,VoicemailMain,101 ;To check your voicemail by dialing ;*86

context=sip.broadvoice.com
;For incoming calls
;This extension line will ring SIP
;extension 01130 for 60 seconds then hang up. Modify as necessary to fit your dialplan
exten => s,1,Answer
;exten=> s,2,Dial(SIP/01130,60,tr)
exten => s,3,hangup()

; This extended Dial Plan will enable International Dialing on The Unlimited World PLUS Plan
; This dial plan enables World Plus countries
; there are no built in ways to prevent calls to cell phone users (except in germany where Cell phone prefix’s are
; carried by 1 and has been accounted for)

exten=_01130.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01131.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01132.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01133.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01134.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011351.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011352.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011353.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011378.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01139.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01141.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011420.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01143.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01144.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01145.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01146.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01147.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01148.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01149[2-9].,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01154.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01155.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01156.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01160.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01161.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01164.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01165.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01181.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01182.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011852.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_01186.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011886.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011972.,1,dial(SIP/${EXTEN}@out-bv,30)
exten=_011.,2,congestion() ; No answer, nothing
exten=_011.,102,busy() ; Busy

#define AST_CAUSE_CHAN_NOT_IMPLEMENTED 66

May the problem from Digital line signals. Please ask your provider.