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@lists.digium.com >>[mailto:asterisk-users-bounces@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@fowlmere.com> >>To: "Asterisk Users Mailing List - Non-Commercial Discussion" >><asterisk-users@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 >>> _______________________________________________ >>> --Bandwidth and Colocation provided by Easynews.com -- >>> >>> Asterisk-Users mailing list >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users >>> >> >>_______________________________________________ >>--Bandwidth and Colocation provided by Easynews.com -- >> >>Asterisk-Users mailing list >>To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users >>
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@row.com
221 => 000,Elaine's Mailbox,elaine@mere.com
223 => 000,RCF Mailbox,paul@cass.co.uk
224 => 000,RCF Mailbox,paul@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@sipgate.co.uk/3853434
register =>3858817:wwwwwwww@sipgate.co.uk/3854343
register =>3858313:wwwwwwww@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@OpSys.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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@lists.digium.com
>>>[mailto:asterisk-users-bounces@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@fowlmere.com>
>>>To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
>>><asterisk-users@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
>>>> _______________________________________________
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>_______________________________________________
>>>--Bandwidth and Colocation provided by Easynews.com --
>>>
>>>Asterisk-Users mailing list
>>>To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
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@lists.digium.com
[mailto:asterisk-users-bounces@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@row.com
221 => 000,Elaine's Mailbox,elaine@mere.com
223 => 000,RCF Mailbox,paul@cass.co.uk
224 => 000,RCF Mailbox,paul@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@sipgate.co.uk/3853434
register =>3858817:wwwwwwww@sipgate.co.uk/3854343
register =>3858313:wwwwwwww@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@OpSys.com>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<asterisk-users@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@lists.digium.com
>>>[mailto:asterisk-users-bounces@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@fowlmere.com>
>>>To: "Asterisk Users Mailing List - Non-Commercial
Discussion"
>>><asterisk-users@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
>>>> _______________________________________________
>>>> --Bandwidth and Colocation provided by Easynews.com --
>>>>
>>>> Asterisk-Users mailing list
>>>> To UNSUBSCRIBE or update options visit:
>>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>>
>>>
>>>_______________________________________________
>>>--Bandwidth and Colocation provided by Easynews.com --
>>>
>>>Asterisk-Users mailing list
>>>To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-users
>>>
> _______________________________________________
> --Bandwidth and Colocation provided by Easynews.com --
>
> Asterisk-Users mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-users
>
_______________________________________________
--Bandwidth and Colocation provided by Easynews.com --
Asterisk-Users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
________________________________________________________________________
This email has been scanned for all viruses by the Star Internet Virus
Screen.
The service is provided in partnership with MessageLabs, the email
security company.
For more information on a higher level of virus protection visit
www.star.net.uk
________________________________________________________________________
______________________________________________________________________
This email has been scanned by the MessageLabs Email Security System.
For more information please visit http://www.messagelabs.com/email
______________________________________________________________________