Pulling Hair Out (now bald) ... Voicemail keeps asking Ext

Asterisk Asterisk Support
1

I have sip set up on Callmanager 4.x. When others call my ext of 2016 on ccm after a busy or no answer, asterisk voice mail answers by saying, “Mailbox … password.” I want it to put them into my mailbox so they can leave a message. Somehow I must be missing something… Please help!

I have spent 16 hours easy on trying to figure this one out.

SIP DN is 7777 on CCM
VOICEMAIL on Asterisk is 7777.

Here is my sip.conf:

[general]
context=default
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
allowexternaldomains=yes
allowexternalinvites=no
allowguest=yes
allowsubscribe=no
allowtransfer=yes
alwaysauthreject=no
autodomain=no
callevents=no
compactheaders=no
dumphistory=no
g726nonstandard=no
ignoreregexpire=no
jbenable=no
jbforce=no
jblog=no
maxcallbitrate=384
maxexpiry=3600
minexpiry=60
nat=no
notifyringing=no
pedantic=no
promiscredir=no
recordhistory=no
relaxdtmf=no
rtcachefriends=no
rtsavesysname=no
rtupdate=no
sendrpid=yes
sipdebug=no
t1min=100
t38pt_udptl=no
[authentication]

[sip]
type=friend
context=incoming
host=172.20.1.57
ipaddr=172.20.1.57
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes

Here is my 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
[other]

[general]
format=wav49|gsm|wav
serveremail=asterisk
attach=yes
skipms=3000
maxsilence=10
silencethreshold=128
maxlogins=3
emaildateformat=%A, %B %d, %Y at %r
sendvoicemail=yes
attachfmt=wav
deletevoicemail=no
envelope=no
maxgreet=60
maxmessage=120
maxmsg=100
minmessage=1
operator=yes
review=yes
saycid=no
sayduration=yes
mailcmd=/usr/sbin/sendmail -t
externotify=/var/libasterisk/scripts/vm.sh
[default]
2016=1234,Steve,steve@abc.com

Here is the relevant parts of my extensions.conf:

[macro-dialout-callmanager]
exten=s,1,ChanIsAvail(SIP/sip)
exten=s,2,Cut(AVAILCHAN=AVAILCHAN,1)
exten=s,3,Dial(${AVAILCHAN}/${ARG1})
exten=s,4,Hangup
exten=s,102,Congestion
[incoming]
exten=7777,1,GotoIf($[${RDNIS}]?2:400)
exten=7777,2,MailboxExists(${RDNIS}@default
exten=7777,3,Congestion
exten=7777,103,Voicemail(su${RDNIS})
exten=7777,104,Playback(vm-goodbye)
exten=7777,105,Hangup
exten=7777,400,VoicemailMain
[general]
static=yes
writeprotect=no
clearglobalvars=no
autofallthrough=yes
priorityjumping=no
[default]
exten=_230XXXX,1,SetCallerID(${EXTEN:3})
exten=_230XXXX,2,Dial(SIP/28888@172.20.1.57)
exten=_230XXXX,3,Answer
exten=_230XXXX,4,Wait,1
exten=_230XXXX,5,Hangup
exten=_231XXXX,1,SetCallerID(${EXTEN:3})
exten=_231XXXX,2,Dial(SIP/28889@172.20.1.57)
exten=_231XXXX,3,Answer
exten=_231XXXX,4,Wait,1
exten=_231XXXX,5,Hangup

I am using users.conf, but don’t know how that ties in or whether I even need it…???

thanks,

Steve

2

Steve what if any errors do you get in the CLI ?

3

Hi

You need to check what the value of ${RDNIS} is so add a line

[incoming]
exten=7777,1,Noop(${RDNIS})
exten=7777,2,GotoIf($[${RDNIS}]?3:400)
exten=7777,3,MailboxExists(${RDNIS}@default
exten=7777,4,Congestion
exten=7777,103,Voicemail(su${RDNIS})
exten=7777,104,Playback(vm-goodbye)
exten=7777,105,Hangup
exten=7777,400,VoicemailMain

But im guessing its empty as its returning false

Ian

4

How would I see these errors in the CLI? Just a newbie…

I will try the NOOP thing. Someone told me that the rdnis is now to be formatted as CALLERID(rdnis)? Does that change the format of the NOOP command lines?

Steve

ps. We are running version 1.4

Here is what the .conf’s look like now (pertinent parts). It still will not let me leave voicemail, still wants extension…

sip.conf

[callman01]
type=peer
context=incoming
host=172.20.1.57
ipaddr=172.20.1.57
disallow=all
allow=ulaw
allow=alaw
nat=no
canreinvite=yes
qualify=yes
usereqphone=yes

users.conf (I don’t see where this gets referred to though)
[2016]
callwaiting=no
cid_number=2016
email=steve@korehicom.com
fullname=Steve
hasagent=no
hasdirectory=no
hasiax=no
hasmanager=no
hassip=no
hasvoicemail=yes
deletevoicemail=no
host=dynamic
mailbox=2016
secret=1234
threewaycalling=no
vmsecret=1234
registeriax=no
registersip=no
autoprov=no
canreinvite=no
nat=no
dtmfmode=rfc2833
disallow=all
allow=all
signalling=fxo_ks

extensions.conf (7777 is voicemailmain; default context exists here and in voicemail.conf)

[globals]
CONSOLE=Console/dsp
IAXINFO=guest
TRUNK=SIP/172.20.1.57
TRUNKMSD=1

[demo]
exten=s,1,Wait(1)
exten=s,n,Answer
exten=s,n,Set(TIMEOUT(digit)=5)
exten=s,n,Set(TIMEOUT(response)=10)
exten=s,n(restart),BackGround(demo-congrats)
exten=s,n(instruct),BackGround(demo-instruct)
exten=s,n,WaitExten
exten=2,1,BackGround(demo-moreinfo)
exten=2,n,Goto(s,instruct)
exten=3,1,Set(LANGUAGE()=fr)
exten=3,n,Goto(s,restart)
exten=1000,1,Goto(default,s,1)
exten=1234,1,Playback(transfer,skip)
exten=1234,n,Macro(stdexten,1234,${GLOBAL(CONSOLE)})
exten=1235,1,Voicemail(1234,u)
exten=1236,1,Dial(Console/dsp)
exten=1236,n,Voicemail(1234,b)
exten=#,1,Playback(demo-thanks)
exten=#,n,Hangup
exten=t,1,Goto(#,1)
exten=i,1,Playback(invalid)
exten=500,1,Playback(demo-abouttotry)
exten=500,n,Dial(IAX2/guest@misery.digium.com/s@default)
exten=500,n,Playback(demo-nogo)
exten=500,n,Goto(s,6)
exten=600,1,Playback(demo-echotest)
exten=600,n,Echo
exten=600,n,Playback(demo-echodone)
exten=600,n,Goto(s,6)
exten=76245,1,Macro(page,SIP/Grandstream1)
exten=_7XXX,1,Macro(page,SIP/${EXTEN})
exten=7999,1,Set(TIMEOUT(absolute)=60)
exten=7999,2,Page(Local/Grandstream1@page&Local/Xlite1@page&Local/1234@page/n|d)
exten=7777,1,VoicemailMain
exten=7777,n,Goto(s,6)

[general]
static=yes
writeprotect=no
clearglobalvars=no
autofallthrough=yes
priorityjumping=no
[incoming]
exten=7777,1,Noop(${RDNIS})
exten=7777,2,GotoIf($[${RDNIS}]?3:400)
exten=7777,3,MailboxExists(${RDNIS}@default
exten=7777,4,Congestion
exten=7777,103,Voicemail(su${RDNIS})
exten=7777,104,Playback(vm-goodbye)
exten=7777,105,Hangup
exten=7777,400,VoicemailMain

[default] (seems to be the same context as in voicemail.conf?)
exten=7777,1,VoiceMailMain
exten=_230XXXX,1,SetCallerID(${EXTEN:3})
exten=_230XXXX,2,Dial(SIP/28888@ciscocm)
exten=_230XXXX,3,Answer
exten=_230XXXX,4,Wait,1
exten=_230XXXX,5,Hangup
exten=_231XXXX,1,SetCallerID(${EXTEN:3})
exten=_231XXXX,2,Dial(SIP/28889@ciscocm)
exten=_231XXXX,3,Answer
exten=_231XXXX,4,Wait,1
exten=_231XXXX,5,Hangup

5

Connected to Asterisk 1.4.17~dfsg-2ubuntu1 currently running on ubuntu-n160 (pid = 17930)
Reliably Transmitting (no NAT) to 172.20.1.57:5060:
OPTIONS sip:172.20.1.57 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK5af4603a;rport
From: “asterisk” sip:asterisk@192.168.1.69;tag=as56901708
To: sip:172.20.1.57
Contact: sip:asterisk@192.168.1.69
Call-ID: 43d57ee17005a6cb3de1ee1537e541ff@192.168.1.69
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 08 May 2008 16:31:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

ubuntu-n160*CLI>
<— SIP read from 172.20.1.57:5060 —>
SIP/2.0 400 Bad Request - 'Malformed/Missing URL’
Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK5af4603a;rport
From: “asterisk” sip:asterisk@192.168.1.69;tag=as56901708
To: sip:172.20.1.57
Call-ID: 43d57ee17005a6cb3de1ee1537e541ff@192.168.1.69
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘43d57ee17005a6cb3de1ee1537e541ff@192.168.1.69’ Method: OPTIONS
Reliably Transmitting (no NAT) to 172.20.1.57:5060:
OPTIONS sip:172.20.1.57 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK561c440f;rport
From: “asterisk” sip:asterisk@192.168.1.69;tag=as71b3a43a
To: sip:172.20.1.57
Contact: sip:asterisk@192.168.1.69
Call-ID: 2ab0117f29de20f43668c7ce65fda945@192.168.1.69
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 08 May 2008 16:32:53 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

ubuntu-n160*CLI>
<— SIP read from 172.20.1.57:5060 —>
SIP/2.0 400 Bad Request - 'Malformed/Missing URL’
Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK561c440f;rport
From: “asterisk” sip:asterisk@192.168.1.69;tag=as71b3a43a
To: sip:172.20.1.57
Call-ID: 2ab0117f29de20f43668c7ce65fda945@192.168.1.69
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘2ab0117f29de20f43668c7ce65fda945@192.168.1.69’ Method: OPTIONS
[May 8 12:33:20] NOTICE[18500]: cdr.c:1371 do_reload: CDR simple logging enabled.
[May 8 12:33:20] NOTICE[18500]: res_odbc.c:565 reload: Adding ENV var: INFORMIXSERVER=my_special_database
[May 8 12:33:20] NOTICE[18500]: res_odbc.c:565 reload: Adding ENV var: INFORMIXDIR=/opt/informix
[May 8 12:33:20] WARNING[18500]: res_smdi.c:746 reload: No SMDI interfaces were specified to listen on, not starting SDMI listener.
[May 8 12:33:20] NOTICE[18500]: indications.c:505 ast_unregister_indication_country: Removed default indication country ‘us’
[May 8 12:33:20] NOTICE[18500]: app_playback.c:455 reload: Reloading say.conf
[May 8 12:33:20] NOTICE[18500]: pbx_ael.c:4094 pbx_load_module: Starting AEL load process.
[May 8 12:33:20] NOTICE[18500]: pbx_ael.c:4101 pbx_load_module: AEL load process: calculated config file name ‘/etc/asterisk/extensions.ael’.
[May 8 12:33:20] NOTICE[18500]: pbx_ael.c:4109 pbx_load_module: AEL load process: parsed config file name ‘/etc/asterisk/extensions.ael’.
[May 8 12:33:20] NOTICE[18500]: pbx_ael.c:4112 pbx_load_module: AEL load process: checked config file name ‘/etc/asterisk/extensions.ael’.
[May 8 12:33:20] NOTICE[18500]: pbx_ael.c:4114 pbx_load_module: AEL load process: compiled config file name ‘/etc/asterisk/extensions.ael’.
[May 8 12:33:20] NOTICE[18500]: pbx_ael.c:4117 pbx_load_module: AEL load process: merged config file name ‘/etc/asterisk/extensions.ael’.
[May 8 12:33:20] NOTICE[18500]: pbx_ael.c:4120 pbx_load_module: AEL load process: verified config file name ‘/etc/asterisk/extensions.ael’.
[May 8 12:33:20] WARNING[18500]: pbx_config.c:2292 pbx_load_config: No closing parenthesis found? ‘MailboxExists(${RDNIS}@default’
[May 8 12:33:20] WARNING[18500]: chan_zap.c:13004 process_zap: Ignoring switchtype
[May 8 12:33:20] WARNING[18500]: chan_zap.c:13004 process_zap: Ignoring signalling
[May 8 12:33:20] WARNING[18500]: chan_zap.c:13004 process_zap: Ignoring rxwink
Reliably Transmitting (no NAT) to 172.20.1.57:5060:
OPTIONS sip:172.20.1.57 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK2dddbb3e;rport
From: “asterisk” sip:asterisk@192.168.1.69;tag=as6d974566
To: sip:172.20.1.57
Contact: sip:asterisk@192.168.1.69
Call-ID: 4a67a03c5cf410614d7bd9093ed2101d@192.168.1.69
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 08 May 2008 16:33:21 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0

ubuntu-n160*CLI>
<— SIP read from 172.20.1.57:5060 —>
SIP/2.0 400 Bad Request - 'Malformed/Missing URL’
Via: SIP/2.0/UDP 192.168.1.69:5060;branch=z9hG4bK2dddbb3e;rport
From: “asterisk” sip:asterisk@192.168.1.69;tag=as6d974566
To: sip:172.20.1.57
Call-ID: 4a67a03c5cf410614d7bd9093ed2101d@192.168.1.69
CSeq: 102 OPTIONS
Content-Length: 0

<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘4a67a03c5cf410614d7bd9093ed2101d@192.168.1.69’ Method: OPTIONS
ubuntu-n160CLI> sip set debug
SIP Debugging re-enabled
ubuntu-n160CLI>
<— SIP read from 172.20.1.57:5060 —>
INVITE sip:7777@192.168.1.69:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.1.57:5060;branch=z9hG4bK1aa0354
From: “callman01” sip:7777@172.20.1.57;tag=16787476
To: sip:7777@192.168.1.69
Date: Thu, 08 May 2008 16:33:58 GMT
Call-ID: 8d662480-1de13b1f-2373-390114ac@172.20.1.57
Supported: timer
Min-SE: 1800
User-Agent: Cisco-CCM4.1
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK
CSeq: 101 INVITE
Max-Forwards: 70
Remote-Party-ID: “callman01” sip:7777@172.20.1.57;party=calling;screen=no;privacy=off
Contact: sip:7777@172.20.1.57:5060
Diversion: sip:2016@172.20.1.57;reason=no-answer
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 225

v=0
o=CiscoSystemsCCM-SIP 2000 1000 IN IP4 172.20.1.57
s=SIP Call
c=IN IP4 172.20.1.57
t=0 0
m=audio 25238 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 PCMU/8000
a=ptime:20
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

<------------->
— (19 headers 11 lines) —
Sending to 172.20.1.57 : 5060 (no NAT)
Using INVITE request as basis request - 8d662480-1de13b1f-2373-390114ac@172.20.1.57
Found peer 'callman01’
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 172.20.1.57:25238
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 172.20.1.57:25238
Looking for 7777 in incoming (domain 192.168.1.69)
RDNIS is 2016
list_route: hop: sip:7777@172.20.1.57:5060

<— Transmitting (no NAT) to 172.20.1.57:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.20.1.57:5060;branch=z9hG4bK1aa0354;received=172.20.1.57
From: “callman01” sip:7777@172.20.1.57;tag=16787476
To: sip:7777@192.168.1.69
Call-ID: 8d662480-1de13b1f-2373-390114ac@172.20.1.57
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7777@192.168.1.69
Content-Length: 0

<------------>
Audio is at 192.168.1.69 port 17270
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 172.20.1.57:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.1.57:5060;branch=z9hG4bK1aa0354;received=172.20.1.57
From: “callman01” sip:7777@172.20.1.57;tag=16787476
To: sip:7777@192.168.1.69;tag=as7007e039
Call-ID: 8d662480-1de13b1f-2373-390114ac@172.20.1.57
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7777@192.168.1.69
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 17930 17930 IN IP4 192.168.1.69
s=session
c=IN IP4 192.168.1.69
t=0 0
m=audio 17270 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16>
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

<------------>
ubuntu-n160*CLI>
<— SIP read from 172.20.1.57:5060 —>
ACK sip:7777@192.168.1.69:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.1.57:5060;branch=z9hG4bK1f343e68
From: “callman01” sip:7777@172.20.1.57;tag=16787476
To: sip:7777@192.168.1.69;tag=as7007e039
Date: Thu, 08 May 2008 16:33:58 GMT
Call-ID: 8d662480-1de13b1f-2373-390114ac@172.20.1.57
Max-Forwards: 70
CSeq: 101 ACK
Content-Length: 0

<------------->
— (9 headers 0 lines) —
ubuntu-n160*CLI>
<— SIP read from 172.20.1.57:5060 —>
BYE sip:7777@192.168.1.69:5060 SIP/2.0
Via: SIP/2.0/UDP 172.20.1.57:5060;branch=z9hG4bK4e909d2
From: “callman01” sip:7777@172.20.1.57;tag=16787476
To: sip:7777@192.168.1.69;tag=as7007e039
Date: Thu, 08 May 2008 16:33:58 GMT
Call-ID: 8d662480-1de13b1f-2373-390114ac@172.20.1.57
User-Agent: Cisco-CCM4.1
Max-Forwards: 70
CSeq: 102 BYE
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Sending to 172.20.1.57 : 5060 (no NAT)

<— Transmitting (no NAT) to 172.20.1.57:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 172.20.1.57:5060;branch=z9hG4bK4e909d2;received=172.20.1.57
From: “callman01” sip:7777@172.20.1.57;tag=16787476
To: sip:7777@192.168.1.69;tag=as7007e039
Call-ID: 8d662480-1de13b1f-2373-390114ac@172.20.1.57
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:7777@192.168.1.69
Content-Length: 0

<------------>
[May 8 12:34:05] WARNING[18504]: app_voicemail.c:6294 vm_authenticate: Couldn’t read username
Really destroying SIP dialog ‘8d662480-1de13b1f-2373-390114ac@172.20.1.57’ Method: BYE
ubuntu-n160CLI> sip set debug off
SIP Debugging Disabled
ubuntu-n160CLI>