Cisco 7950 Can receive calls but cannot call out

Hi All,

My buddy was in town and setup asterisk for me. I can call the cisco phone and answer it, however if I try to dial another extension from it, it says “Proceeding (in 100)” then after about 10 seconds of silence I get the voicemail for the extension I was calling and the other phone never rings. I can immediately call 6003 which is voicemail though without any problems. Any help would be greatly appreciated.

Here are my config files.

Sip.conf

srvlookup=yes ; Enable DNS SRV lookups on outbound calls
; Note: Asterisk only uses the first host
; in SRV records
; Disabling DNS SRV lookups disables the
; ability to place SIP calls based on domain
; names to some other SIP users on the Internet
pedantic=no

register => *:@sip.broadvoice.com

[authentication]

[xlite1]
type=friend
regexten=6000 ; When they register, create extension 1234
callerid=“Ray” <6000>
host=dynamic ; This device needs to register
canreinvite=yes ; Typically set to NO if behind NAT
disallow=all
allow=gsm ; GSM consumes far less bandwidth than ulaw
allow=ulaw
allow=alaw
mailbox=6000@default

[analog]
type=friend
regexten=6001
callerid=“Analog” <6001>
host=dynamic
canreinvite=yes
disallow=all
allow=gsm
allow=ulaw
allow=alaw
mailbox=6001@default

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=*********
secret=************
username=*******
insecure=very
authname=*********
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no

[office]
type=friend
username=office
secret=test
host=dynamic
dtmfmode=rfc2833
context=default
canreinvite=yes
callerid="Ray <6004>"
mailbox=6004@default


Extensions.conf

; extensions.conf - the Asterisk dial plan
;
; 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 “dialplan 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 “dialplan save” 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. This is the default.
;
; 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=no
;
; 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 of its included files, will remain set to the previous value.
;
; NOTE: A complication sets in, if you put your global variables into
; the AEL file, instead of the extensions.conf file. With clearglobalvars
; set, a “reload” will often leave the globals vars cleared, because it
; is not unusual to have extensions.conf (which will have no globals)
; load after the extensions.ael file (where the global vars are stored).
; So, with “reload” in this particular situation, first the AEL file will
; clear and then set all the global vars, then, later, when the extensions.conf
; file is loaded, the global vars are all cleared, and then not set, because
; they are not stored in the extensions.conf file.
;
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=yes
;
; User context is where entries from users.conf are registered. The
; default value is ‘default’
;
;userscontext=default
;
; 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 the GLOBAL dialplan function:
; ${GLOBAL(VARIABLE)}
; ${${GLOBAL(VARIABLE)}} or ${text${GLOBAL(VARIABLE)}} or any hybrid
; Unix/Linux environmental variables can be reached with the ENV dialplan
; function: ${ENV(VARIABLE)}
;
[globals]
CONSOLE=Console/dsp ; Console interface for demo

[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(${ARG1},u) ; 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(${ARG1},b) ; 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(${ARG1},u) ; 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(${ARG1},b) ; 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

[testing]
;
; Create an extension, 6002, for evaluating echo latency.
;
exten => 6002,1,Playback(demo-echotest) ; Let them know what’s going on
exten => 6002,n,Echo ; Do the echo test
exten => 6002,n,Playback(demo-echodone) ; Let them know it’s over
exten => 6002,n,Goto(s,6) ; Start over

[default]
include => testing

; incoming calls from broadvoice
exten => *****,1,Dial(SIP/analog&SIP/xlite1,20)
exten => ******,2,Voicemail(6001,u)
exten => ******,n,Hangup

; outgoing calls to broadvoice
exten => _91NXXNXXXXXX,1,dial(SIP/${EXTEN:1}@sip.broadvoice.com,30)
exten => _91NXXNXXXXXX,2,congestion()
exten => _91NXXNXXXXXX,102,busy()
exten => _9NXXXXXX,1,dial(SIP/1812${EXTEN:1}@sip.broadvoice.com,30)
exten => _9NXXXXXX,2,congestion()
exten => _9NXXXXXX,102,busy()

; local extensions
exten => 6000,1,Macro(stdexten,6000,SIP/xlite1)
exten => 6001,1,Macro(stdexten,6001,SIP/analog)
exten => 6003,1,VoicemailMain
exten => 6003,n,Hangup
exten => 6004,1,Macro(stdexten,6004,SIP/office)


Voicemail.conf

[zonemessages]
eastern=America/New_York|‘vm-received’ Q ‘digits/at’ IMp
central=America/Chicago|‘vm-received’ Q ‘digits/at’ IMp
central24=America/Chicago|‘vm-received’ q ‘digits/at’ H N 'hours’
military=Zulu|‘vm-received’ q ‘digits/at’ H N ‘hours’ 'phonetic/z_p’
european=Europe/Copenhagen|‘vm-received’ a d b ‘digits/at’ HM

[default]
; Define maximum number of messages per folder for a particular context.
;maxmsg=50

1234 => 4242,Example Mailbox,root@localhost
;4200 => 9855,Mark Spencer,markster@linux-support.net,mypager@digium.com,attach=no|serveremail=myaddy@digium.com|tz=central|maxmsg=10
;4300 => 3456,Ben Rigas,ben@american-computer.net
;4310 => -5432,Sales,sales@marko.net
;4069 => 6522,Matt Brooks,matt@marko.net,|tz=central|attach=yes|saycid=yes|dialout=fromvm|callback=fromvm|review=yes|operator=yes|envelope=yes|sayduration=yes|saydurationm=1
;4073 => 1099,Bianca Paige,bianca@biancapaige.com,delete=1
;4110 => 3443,Rob Flynn,rflynn@blueridge.net
;4235 => 1234,Jim Holmes,jim@astricon.ips,Tz=european
6000 => 1234,Ray
6001 => 1234,Analog,r***@*****.com
6004 => 1234,office

;
; Mailboxes may be organized into multiple contexts for
; voicemail virtualhosting
;

[other]
;The intro can be customized on a per-context basis
;directoryintro=dir-company2
1234 => 5678,Company2 User,root@localhost

Could you post the output from the CLI when you try to make a call? That’s the fastest way to figure out what is wrong.