[Asterisk-Users] Problem - Voicemail resets phone

Peter Braidwood Peter at braidwood.co.uk
Mon Apr 10 12:12:06 MST 2006


I have this for my voicemail, has worked perfectly for years

; Dial 300 from any phone to go to the voicemail system
;
exten => 300,1,Wait,1
exten => 300,2,VoicemailMain,s${CALLERIDNUM}
exten => 300,3,Hangup

Peter


-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Paul A
Brown
Sent: 10 April 2006 19:52
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Problem - Voicemail resets phone

Hi

Details below.... If you see any other oddities with my configs please
let 
me know......

Thanks

Cisco 7960 running SIP config P003-07-3-00

VOICEMAIL.CONF
[general]
format=wav49|gsm|wav
attach=yes
maxmessage=180
maxgreet=60
skipms=3000
silencethreshold=128

[default]
; Define maximum number of messages per folder for partcular context.
;maxmsg=50
220 => 000,220 Mailbox,paul at row.com
221 => 000,Elaine's Mailbox,elaine at mere.com
223 => 000,RCF Mailbox,paul at cass.co.uk
224 => 000,RCF Mailbox,paul at cass.co.uk


EXTENSIONS.CONF
[general]
static=yes
writeprotect=no

[globals]
 PHONES1=SIP/220
 PHONES1VM=220
 PHONES2=SIP/221
 PHONES2VM=221
 PHONES3=SIP/222
 PHONES3VM=222
 PHONES4=SIP/223
 PHONES4VM=223
 PHONES5=SIP/224
PHONES5VM=224
 PHONES5=SIP/225
 PHONES5VM=225

[sipdiscount-outbound]
exten => _4.,1,Dial(SIP/${EXTEN}@sipdiscount)

 [home]

 ; Line 1

 exten => 220,1,Dial(${PHONES1},20,Ttr)
 exten => 220,2,VoiceMail,u220
 exten => 220,3,Hangup
 exten => 220,102,VoiceMail,b220

 ; Line 2

 exten => 221,1,Dial(${PHONES2},20,Ttm)
 exten => 221,2,VoiceMail,u221
 exten => 221,3,Hangup
 exten => 221,102,VoiceMail,b221

; Mailbox

 exten => 666,1,Ringing
exten => 666,2,Wait(2)
 exten => 666,3,VoicemailMain

 ; MeetMe 100

 exten => 1000,1,Meetme,100

 include => sipdiscount-outbound
 include => parkedcalls

SIP.CONF
[general]
context=default
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
;defaultexpiry=3600
videosupport=yes
recordhistory=yes
;sipdebug = yes
externip = 84.5.5.115
externhost=wall.co.uk
localnet=192.192.192.0/255.255.255.0
allow=alaw
allow=alaw
allow=ulaw
allow=g729
allow=gsm
allow=slinear
srvlookup=yes
register =>3858294:wwwwwwww at sipgate.co.uk/3853434
register =>3858817:wwwwwwww at sipgate.co.uk/3854343
register =>3858313:wwwwwwww at sipgate.co.uk/3858343



[220]
type=friend
context=home
callerid=Paul<220>
nat=yes
host=dynamic
defaultip=192.192.192.220
username=220
secret=secret
mailbox=220
dtmfmode=rfc2833

[221]
type=friend
context=home
callerid=Ellie<221>
nat=yes
host=dynamic
defaultip=192.192.192.221
username=221
secret=secret
mailbox=221
dtmfmode=rfc2833

SIP<mac>

# Line 1 appearance
line1_name: "220"
line1_shortname: "220"
line1_authname: "220"
line1_password: "secret"

# Line 2 appearance
line2_name:
line2_shortname:
line2_authname:
line2_password:

# Line 3 Camb Para
line3_name: "3853434"
line3_authname: "3853434"
line3_password: secret
line3_shortname: "CambPara"
line3_displayname: "CambPara"
preferred_codec: g729a

#Line 4 London
line4_name: "1993434"
line4_authname: "1993434" (SIP ID)
line4_password: secret
line4_shortname: "020 7099 3434"
line4_displayname: "020 7099 3434"
preferred_codec: g729a

####Line5 Sipgate####

line5_name : "3854343"
line5_authname: "3854343"
line5_password:  secret
line5_shortname: "01223 854343"
line5_displayname: "01223 85434"
preferred_codec: g729a

####Line6 SipDiscount####

line6_name: "Sipdiscount"
line6_authname: "343434" (SIP ID)
line6_password: secret
line6_shortname: "Free Calls" (London in this sample with SIP ID)
line6_displayname: "Free Calls" (London in this sample with SIP ID)
preferred_codec: g711ulaw

####### New Parameters added in Release 2.0 #######

# All user_parameters have been removed

# Phone Label (Text desired to be displayed in upper right corner)
phone_label: "Paul "    ; Has no effect on SIP messaging

# Line 1 Display Name (Display name to use for SIP messaging)
#line1_displayname: "223434"
line1_displayname: "Paul"

# Line 2 Display Name (Display name to use for SIP messaging)
#line2_displayname: " "


####### New Parameters added in Release 3.0 ######

# Phone Prompt (The prompt that will be displayed on console and telnet)
phone_prompt:   "PB Phone"      ; Limited to 15 characters (Default -
SIP 
Phone
)

# Phone Password (Password to be used for console or telnet login)
phone_password: "secret" ; Limited to 31 characters (Default - cisco)

# User classifcation used when Registering [ none(default), phone, ip ]
user_info: 220


----- Original Message ----- 
From: "Alexander Lopez" <Alex.Lopez at OpSys.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion" 
<asterisk-users at lists.digium.com>
Sent: Monday, April 10, 2006 7:19 PM
Subject: RE: [Asterisk-Users] Problem - Voicemail resets phone


> Can you also post information such as:
>
> Type of phone (model Number would be idela)
> How is it conencted, SIP, ZAP, IAX, Channel Bank.
>
> Corresponding config files would also help.
>
> Help us help you.
>
>
>>>-----Original Message-----
>>>From: asterisk-users-bounces at lists.digium.com
>>>[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of
>>>Paul A Brown
>>>Sent: Monday, April 10, 2006 2:12 PM
>>>To: Asterisk Users Mailing List - Non-Commercial Discussion
>>>Subject: Re: [Asterisk-Users] Problem - Voicemail resets phone
>>>
>>>Just reposting as I know a lot of fresh faces are online :-)
>>>
>>>Any help appreciated
>>>
>>>Thanks
>>>
>>>Paul
>>>----- Original Message -----
>>>From: "Paul A Brown" <paul at fowlmere.com>
>>>To: "Asterisk Users Mailing List - Non-Commercial Discussion"
>>><asterisk-users at lists.digium.com>
>>>Sent: Sunday, April 09, 2006 8:44 PM
>>>Subject: [Asterisk-Users] Problem - Voicemail resets phone
>>>
>>>
>>>> Hi Everyone,
>>>>
>>>> Things seem to work fine (except my phone audio issue in a
>>>previous mail)
>>>>
>>>> I can leave a vmail message and it emails it out fine.
>>>However when I dial
>>>> the vmail server from any phone it usually resets the phone
>>>half way
>>>> through. There is no single point where it starts to do
>>>this, it can vary
>>>> but it happens sometimes after I connect to the vmail server.
>>>>
>>>> Has anyone seen this? What other details can I post?
>>>>
>>>> Thanks
>>>>
>>>> Paul
>>>> _______________________________________________
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>>>
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>>>
> _______________________________________________
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>   http://lists.digium.com/mailman/listinfo/asterisk-users
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