I dial the extension for voicemail access via Voicemailmain. I then enter a configured mailbox and the appropriate password. Asterisk then tells me that the password is incorrect, even though it is. What am I doing wrong?
if you voicemail aren’t in the default context, add this :
exten => x,x,Voicemailmain(@context)
replace the context by your voicemail’s context
I do have the context specified in the Voicemailmain call. It is Voicemailmain(@sip-in-local)
can You post part of extensions.conf where access to voicemail is defined,
voicemail.conf voicemailbox definitions and
Asterisk CLI> info about call to voicemail ?
extensions.conf
;
;
[general]
;
autofallthrough=yes
;
[macro-stdexten];
;
; Standard extension macro:
; ${ARG1} - Extension (we could have used ${MACRO_EXTEN} here as well
;
exten => s,1,Dial(SIP/${ARG1},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}@sip-in-local) ; If unavailable, send to voicemail w/ unavail announce
exten => s-NOANSWER,n,Goto(default,s,1) ; If they press #, return to start
exten => s-BUSY,1,Voicemail(b${ARG1}@sip-in-local) ; If busy, send to voicemail w/ busy announce
exten => s-BUSY,n,Goto(default,s,1) ; If they press #, return to start
exten => _s-.,1,Goto(s-NOANSWER,1) ; Treat anything else as no answer
exten => a,1,VoicemailMain(${ARG1}@sip-in-local) ; If they press *, send the user into VoicemailMain
exten => a,n,Playback(vm-goodbye)
exten => a,n,Hangup()
;
;
[default]
;
exten => s,1,Answer()
exten => s,n,Set(TIMEOUT(digit)=2)
exten => s,n,Set(TIMEOUT(response)=5)
exten => s,n,Background(enter-ext-of-person)
exten => s,n,WaitExten(10)
exten => 300,1,VoicemailMain(@sip-in-local)
exten => 300,n,Playback(vm-goodbye)
exten => 300,n,Hangup()
exten => _3XX,1,Answer()
exten => _3XX,n,Macro(stdexten,${EXTEN})
exten => i,1,playback(pbx-invalid)
exten => i,2,Goto(s,2)
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
;
;
[sip-in-local]
ignorepat = 9
exten => 300,1,Answer()
exten => 300,n,VoiceMailMain(@sip-in-local)
exten => 300,n,Hangup()
exten => _3XX,1,Answer()
exten => _3XX,n,Macro(stdexten,${EXTEN})
exten => _91800NXXXXXX,1,Dial(SIP/ipgwout/${EXTEN:1})
exten => _91888NXXXXXX,1,Dial(SIP/ipgwout/${EXTEN:1})
exten => _91877NXXXXXX,1,Dial(SIP/ipgwout/${EXTEN:1})
exten => _91866NXXXXXX,1,Dial(SIP/ipgwout/${EXTEN:1})
exten => _91NXXNXXXXXX,1,Dial(SIP/ipgwout/${EXTEN:1})
exten => _9011,1,Dial(SIP/ipgwout/${EXTEN}:1})
exten => i,1,playback(pbx-invalid)
exten => i,2,Hangup()
exten => t,1,Playback(vm-goodbye)
exten => t,2,Hangup()
;
;
[from-world]
exten => s,1,Goto(default,s,1)
voicemail.conf
;
; Voicemail Configuration
;
;
; NOTE: Asterisk has to edit this file to change a user’s password. This does
; note currently work with the "#include " directive for Asterisk
; configuration files. Do not use it with this configuration file.
;
[general]
; Default formats for writing Voicemail
;format=g723sf|wav49|wav
format=wav49|gsm|wav
;
; WARNING:
; If you change the list of formats that you record voicemail in
; when you have mailboxes that contain messages, you MUST absolutely
; manually go through those mailboxes and convert/delete/add the
; the message files so that they appear to have been stored using
; your new format list. If you don’t do this, very unpleasant
; things may happen to your users while they are retrieving and
; manipulating their voicemail.
;
; In other words: don’t change the format list on a production system
; unless you are VERY sure that you know what you are doing and are
; prepared for the consequences.
;
; Who the e-mail notification should appear to come from
serveremail=asterisk
;serveremail=asterisk@linux-support.net
; Should the email contain the voicemail as an attachment
attach=yes
; Maximum number of messages per folder. If not specified, a default value
; (100) is used. Maximum value for this option is 9999.
;maxmsg=100
; Maximum length of a voicemail message in seconds
;maxmessage=180
; Minimum length of a voicemail message in seconds for the message to be kept
; The default is no minimum.
;minmessage=3
; Maximum length of greetings in seconds
;maxgreet=60
; How many miliseconds to skip forward/back when rew/ff in message playback
skipms=3000
; How many seconds of silence before we end the recording
maxsilence=10
; Silence threshold (what we consider silence, the lower, the more sensitive)
silencethreshold=128
; Max number of failed login attempts
maxlogins=3
; If you need to have an external program, i.e. /usr/bin/myapp called when a
; voicemail is left, delivered, or your voicemailbox is checked, uncomment
; this:
;externnotify=/usr/bin/myapp
; If you need to have an external program, i.e. /usr/bin/myapp called when a
; voicemail password is changed, uncomment this:
;externpass=/usr/bin/myapp
; For the directory, you can override the intro file if you want
;directoryintro=dir-intro
; The character set for voicemail messages can be specified here
;charset=ISO-8859-1
; The ADSI feature descriptor number to download to
;adsifdn=0000000F
; The ADSI security lock code
;adsisec=9BDBF7AC
; The ADSI voicemail application version number.
;adsiver=1
; Skip the “[PBX]:” string from the message title
;pbxskip=yes
; Change the From: string
;fromstring=The Asterisk PBX
; Permit finding entries for forward/compose from the directory
;usedirectory=yes
;
; Change the from, body and/or subject, variables:
; VM_NAME, VM_DUR, VM_MSGNUM, VM_MAILBOX, VM_CALLERID, VM_CIDNUM,
; VM_CIDNAME, VM_DATE
;
; Note: The emailbody config row can only be up to 512 characters due to a
; limitation in the Asterisk configuration subsystem.
;emailsubject=[PBX]: New message ${VM_MSGNUM} in mailbox ${VM_MAILBOX}
; The following definition is very close to the default, but the default shows
; just the CIDNAME, if it is not null, otherise just the CIDNUM, or “an unknown
; caller”, if they are both null.
;emailbody=Dear ${VM_NAME}:\n\n\tjust wanted to let you know you were just left a ${VM_DUR} long message (number ${VM_MSGNUM})\nin mailbox ${VM_MAILBOX} from ${VM_CALLERID}, on ${VM_DATE}, so you might\nwant to check it when you get a chance. Thanks!\n\n\t\t\t\t–Asterisk\n
;
; You can also change the Pager From: string, the pager body and/or subject.
; The above defined variables also can be used here
;pagerfromstring=The Asterisk PBX
;pagersubject=New VM
;pagerbody=New ${VM_DUR} long msg in box ${VM_MAILBOX}\nfrom ${VM_CALLERID}, on ${VM_DATE}
;
; Set the date format on outgoing mails. Valid arguments can be found on the
; strftime(3) man page
;
; Default
emaildateformat=%A, %B %d, %Y at %r
; 24h date format
;emaildateformat=%A, %d %B %Y at %H:%M:%S
;
; You can override the default program to send e-mail if you wish, too
;
;mailcmd=/usr/sbin/sendmail -t
;
; Users may be located in different timezones, or may have different
; message announcements for their introductory message when they enter
; the voicemail system. Set the message and the timezone each user
; hears here. Set the user into one of these zones with the tz= attribute
; in the options field of the mailbox. Of course, language substitution
; still applies here so you may have several directory trees that have
; alternate language choices.
;
; Look in /usr/share/zoneinfo/ for names of timezones.
; Look at the manual page for strftime for a quick tutorial on how the
; variable substitution is done on the values below.
;
; Supported values:
; ‘filename’ filename of a soundfile (single ticks around the filename
; required)
; ${VAR} variable substitution
; A or a Day of week (Saturday, Sunday, …)
; B or b or h Month name (January, February, …)
; d or e numeric day of month (first, second, …, thirty-first)
; Y Year
; I or l Hour, 12 hour clock
; H Hour, 24 hour clock (single digit hours preceded by “oh”)
; k Hour, 24 hour clock (single digit hours NOT preceded by “oh”)
; M Minute, with 00 pronounced as “o’clock”
; N Minute, with 00 pronounced as “hundred” (US military time)
; P or p AM or PM
; Q “today”, “yesterday” or ABdY
; (*note: not standard strftime value)
; q “” (for today), “yesterday”, weekday, or ABdY
; (*note: not standard strftime value)
; R 24 hour time, including minute
;
;
;
; Each mailbox is listed in the form =,,,<pager_email>,
; if the e-mail is specified, a message will be sent when a message is
; received, to the given mailbox. If pager is specified, a message will be
; sent there as well. If the password is prefixed by ‘-’, then it is
; considered to be unchangable.
;
; Advanced options example is extension 4069
; NOTE: All options can be expressed globally in the general section, and
; overriden in the per-mailbox settings, unless listed otherwise.
;
; tz=central ; Timezone from zonemessages above. Irrelevant if envelope=no.
; attach=yes ; Attach the voicemail to the notification email NOT the pager email
; saycid=yes ; Say the caller id information before the message. If not described,
; or set to no, it will be in the envelope
; cidinternalcontexts=intern ; Internal Context for Name Playback instead of extension digits when saying caller id.
; sayduration=no ; Turn on/off the duration information before the message. [ON by default]
; saydurationm=2 ; Specify the minimum duration to say. Default is 2 minutes
; dialout=fromvm ; Context to dial out from [option 4 from the advanced menu]
; if not listed, dialing out will not be permitted
sendvoicemail=yes ; Context to Send voicemail from [option 5 from the advanced menu]
; if not listed, sending messages from inside voicemail will not be
; permitted
; searchcontexts=yes ; Current default behavior is to search only the default context
; if one is not specified. The older behavior was to search all contexts.
; This option restores the old behavior [DEFAULT=no]
; callback=fromvm ; Context to call back from
; if not listed, calling the sender back will not be permitted
; review=yes ; Allow sender to review/rerecord their message before saving it [OFF by default
; operator=yes ; Allow sender to hit 0 before/after/during leaving a voicemail to
; reach an operator [OFF by default]
; envelope=no ; Turn on/off envelope playback before message playback. [ON by default]
; This does NOT affect option 3,3 from the advanced options menu
; delete=yes ; After notification, the voicemail is deleted from the server. [per-mailbox only]
; This is intended for use with users who wish to receive their voicemail ONLY by email.
; Note: deletevoicemail is provided as an equivalent option for Realtime configuration.
; nextaftercmd=yes ; Skips to the next message after hitting 7 or 9 to delete/save current message.
; [global option only at this time]
; forcename=yes ; Forces a new user to record their name. A new user is
; determined by the password being the same as
; the mailbox number. The default is “no”.
; forcegreetings=no ; This is the same as forcename, except for recording
; greetings. The default is “no”.
; hidefromdir=yes ; Hide this mailbox from the directory produced by app_directory
; The default is “no”.
[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’
[default]
; Define maximum number of messages per folder for partcular 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
[sip-in-local]
;
301 => 301,“Engineering Lab”,tz=eastern|operator=yes|cidinternalcontexts=sip-in-local|forcename=yes|forcegreetings=yes
302 => 123456,“Engineering Lab”,tz=eastern|operator=yes|cidinternalcontexts=sip-in-local|forcename=yes|forcegreetings=yes
;
;
; 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
CLI
CLI> show voicemail users
Context Mbox User Zone NewMsg
sip-in-local 301 “Engineering Lab” eastern 0
sip-in-local 302 “Engineering Lab” eastern 0
other 1234 Company2 User 0
*CLI> show voicemail zones
Zone Timezone Message Format
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’
*CLI> sip debug
SIP Debugging enabled
*CLI>
<-- SIP read from 10.50.1.94:5064:
INVITE sip:300@obo.globalsat.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK28ca1b44348b5c28
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone
Contact: sip:301@10.50.1.94:5064;user=phone
Supported: replaces, timer
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58529 INVITE
User-Agent: Grandstream GXP2000 1.1.0.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 247
v=0
o=301 8000 8000 IN IP4 10.50.1.94
s=SIP Call
c=IN IP4 10.50.1.94
t=0 0
m=audio 5004 RTP/AVP 0 3 4 8 18
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:20
— (13 headers 13 lines)—
Using INVITE request as basis request - bf4b6c2cb2817883@10.50.1.94
Sending to 10.50.1.94 : 5064 (non-NAT)
Reliably Transmitting (no NAT) to 10.50.1.94:5064:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK28ca1b44348b5c28;received=10.50.1.94
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone;tag=as422fb188
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58529 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:300@10.50.1.21
Proxy-Authenticate: Digest realm=“obo.globalsat.net”, nonce="533ec03b"
Content-Length: 0
Scheduling destruction of call ‘bf4b6c2cb2817883@10.50.1.94’ in 15000 ms
Found user ‘301’
<-- SIP read from 10.50.1.94:5064:
ACK sip:300@obo.globalsat.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK28ca1b44348b5c28
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone;tag=as422fb188
Contact: sip:301@10.50.1.94:5064;user=phone
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58529 ACK
User-Agent: Grandstream GXP2000 1.1.0.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
— (11 headers 0 lines)—
<-- SIP read from 10.50.1.94:5064:
INVITE sip:300@obo.globalsat.net;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK6e50b2b519349a38
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone
Contact: sip:301@10.50.1.94:5064;user=phone
Supported: replaces, timer
Proxy-Authorization: Digest username=“301”, realm=“obo.globalsat.net”, algorithm=MD5, uri="sip:300@obo.globalsat.net;user=phone", nonce=“533ec03b”, response="54497f8d85672645c91890e6335d28e6"
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58530 INVITE
User-Agent: Grandstream GXP2000 1.1.0.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Type: application/sdp
Content-Length: 247
v=0
o=301 8000 8001 IN IP4 10.50.1.94
s=SIP Call
c=IN IP4 10.50.1.94
t=0 0
m=audio 5004 RTP/AVP 0 3 4 8 18
a=sendrecv
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=ptime:20
— (14 headers 13 lines)—
Using INVITE request as basis request - bf4b6c2cb2817883@10.50.1.94
Sending to 10.50.1.94 : 5064 (non-NAT)
Found user '301’
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Peer audio RTP is at port 10.50.1.94:5004
Found description format PCMU
Found description format GSM
Found description format G723
Found description format PCMA
Found description format G729
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10f (g723|gsm|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Looking for 300 in sip-in-local (domain obo.globalsat.net)
list_route: hop: sip:301@10.50.1.94:5064;user=phone
Transmitting (no NAT) to 10.50.1.94:5064:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK6e50b2b519349a38;received=10.50.1.94
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58530 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:300@10.50.1.21
Content-Length: 0
-- Executing Answer("SIP/301-384e", "") in new stack
We’re at 10.50.1.21 port 10654
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Reliably Transmitting (no NAT) to 10.50.1.94:5064:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK6e50b2b519349a38;received=10.50.1.94
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone;tag=as180d015a
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58530 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:300@10.50.1.21
Content-Type: application/sdp
Content-Length: 203
v=0
o=root 20396 20396 IN IP4 10.50.1.21
s=session
c=IN IP4 10.50.1.21
t=0 0
m=audio 10654 RTP/AVP 3 0 8
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=silenceSupp:off - - - -
-- Executing VoiceMailMain("SIP/301-384e", "@sip-in-local") in new stack
<-- SIP read from 10.50.1.94:5064:
ACK sip:300@10.50.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK79112c002d1e8219
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone;tag=as180d015a
Contact: sip:301@10.50.1.94:5064;user=phone
Proxy-Authorization: Digest username=“301”, realm=“obo.globalsat.net”, algorithm=MD5, uri="sip:300@10.50.1.21", nonce=“533ec03b”, response="ce8d8313981cbfbd6d91a6968bc25fa6"
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58530 ACK
User-Agent: Grandstream GXP2000 1.1.0.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
— (12 headers 0 lines)—
– Playing ‘vm-login’ (language ‘en’)
– Playing ‘vm-password’ (language ‘en’)
– Incorrect password ‘’ for user ‘301’ (context = sip-in-local)
– Playing ‘vm-incorrect-mailbox’ (language ‘en’)
– Playing ‘vm-password’ (language ‘en’)
– Incorrect password ‘’ for user ‘301’ (context = sip-in-local)
– Playing ‘vm-incorrect-mailbox’ (language ‘en’)
<-- SIP read from 10.50.1.94:5064:
BYE sip:300@10.50.1.21 SIP/2.0
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK9f4986ec1174ed75
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone;tag=as180d015a
Proxy-Authorization: Digest username=“301”, realm=“obo.globalsat.net”, algorithm=MD5, uri="sip:300@10.50.1.21", nonce=“533ec03b”, response="e31829ce742201525f796b168a5da8a3"
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58531 BYE
User-Agent: Grandstream GXP2000 1.1.0.11
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK
Content-Length: 0
— (11 headers 0 lines)—
Sending to 10.50.1.94 : 5064 (non-NAT)
Transmitting (no NAT) to 10.50.1.94:5064:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.50.1.94:5064;branch=z9hG4bK9f4986ec1174ed75;received=10.50.1.94
From: “Engineering Lab” sip:301@obo.globalsat.net;user=phone;tag=db207b8d18b65a3f
To: sip:300@obo.globalsat.net;user=phone;tag=as180d015a
Call-ID: bf4b6c2cb2817883@10.50.1.94
CSeq: 58531 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:300@10.50.1.21
Content-Length: 0
Jun 21 09:28:35 WARNING[20419]: app_voicemail.c:4988 vm_authenticate: Couldn’t read username
Destroying call ‘bf4b6c2cb2817883@10.50.1.94’
i looked at Your configs for a while…
looks quite ok for me.
i tested similar configuration on my Asterisk
and it is ok.
my sugestion is to check dtmf mode of the phone
i have dtmf=rfc2833
i don’t have another ideas right now.
I have dtmfmode=rfc2833 for both channels 301 and 302.
Do you know if the DTMF works on your phones?? I had the same problem as you when I first started with asterisk. Depending on the phones, they sometimes also have DTMF settings you need to set in the web config. Try and test to see if the DTMF’s work. Create an IVR or something that requires DTMF.
A similar problem happened to me once. Everything was good, but when I tried to enter the password for the voicemail, I would always get a wrong password even thought it was the right one in my voicemail.conf
My problem was the formatting of my file. I did the configuration in windows in a text file and then copied it to my linux box. For some odd reasons, it didn’t work. I retype the exact same thing with VI and it worked.
I don’t know, might be the same problem…
DonFoucker
Had same problem here. Also something that messed me around for awhile was extracting the Asterisk.tar.gz file’s in windows then copying it to a Linux box. Beleive me, they dont compile. Come’s up with some weird errors when trying to compile.
I created the conf files either fresh or edited the samples on the linux (RHEL 4) system.
I found an error in the dtfmmode of the phone. All is now working.
Thanks for the help.
I’m having a similar problem. I’ve checked the dtmfmode=rfc2833 flag, and have gone through all the configs for the eyeBeam v1.5.5.1 (it’s the phone I’m testing on)
My config is similar to that posted previously, the only thing I’ve found, though I don’t know if it’s supposed to be that way, is when I view the Asterisk debug I get the following messages:-
Executing Answer(“SIP/xxxxxxxxxx-94b4”, “”) in new stack
Executing VoiceMailMain(“SIP/xxxxxxxxxx- 94b4”, “sip:+44xxxxxxxxxx@siptestdomaincom”) in new stack
Playing ‘vm-login’ (language ‘en’)
WARNING[21676]: app_voicemail.c:4988 vm_authenticate: Couldn’t read username
The only thing that I can see which may be out of place is on the 1st & 2nd lines where an additional “-” + 4 alphanumeric digits have been added (in bold). As I said earlier, I don’t know if this is normal behaviour.
Any help on this one?
Gareth
g_man,
Turn on sip debug at the asterisk console and place a call from eyebeam to voicemail.
Look at the SDP of the INVITE of the phone. One of the rtpmap lines should look like this.
a=rtpmap:101 telephone-event/8000
This the event for RFC2833 DTMF handling
Notice in Stuart’s invite the line was missing and when asterisk was checking the events supported by the phone it returned:
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
When I call asterisk from an eyebeam, in the sip debug log, mine shows:
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
If yours is the other way where us is 0x0 and peer is 0x1, then RFC2833 is not configured for that user on asterisk.By looking at that line in your sip debug, you should be able to see which side is the problem.
The combined needs to be 0x1 for rfc2833 DTMF to work.
[quote=“SuperB”]
a=rtpmap:101 telephone-event/8000
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x0 (nothing), combined - 0x0 (nothing)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
The combined needs to be 0x1 for rfc2833 DTMF to work.[/quote]
SuperB,
Just checked the asterisk debug, and yes I get:-
“a=rtpmap:101 telephone-event/8000”
And get:-
“Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)”
So does that mean DTMF is working OK? Any other ideas? I take it the additional alphanumeric digits in my previous post are supposed to be there?
Just to let you know as I didn’t include in before I’m using Asterisk RealTime with mySQL.
Gareth.
Just tried dialing into voicemail using a Sipura SPA2002 and had similar results to the eyeBeam, so it looks like it’s the asterisk server then?
Gareth
g-man,
Based on your posting, DTMF looks ok and the problem is elsewhere within asterisk. I do not use “real time” with mysql, but based on the asterisk error message, could asterisk be having difficult getting the user name from the database when voicemail is accessed
Thanks for your help SuperB.
Anyone else have any ideas what could be the problem? Anyone had similar problems listening to VM using RealTime?
Gareth.