Joe Wood
2010-Aug-05  01:52 UTC
[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
Hello.
I have been beating my head over this problem for about 6 hours now.
I have a SIP peer, who I register to (successfully), who should be
directing all incoming calls at my [default] stanza in my
extensions.conf:
[ Context 'default' created by 'pbx_config' ]
  's' =>            1. Wait(1)                                   
[pbx_config]
                    2. Answer()                                   [pbx_config]
                    3. Background(welcome)                        [pbx_config]
                    4. Background(and)                            [pbx_config]
                    5. Background(thank-you-for-calling)          [pbx_config]
                    6. Background(conference-reservations)        [pbx_config]
                    7. Waitfor()                                  [pbx_config]
                    8. Hangup()                                   [pbx_config]
Unfortunately, no matter how I configure extensions.conf or sip.conf,
the phone call always ends up saying: "Extension is unavailable.
Please leave your message after the tone".
sip.conf:
[general]
register => NPANXXZZZZ:PASSWORD at SERVICE_PROVIDER_IP
registertimeout=29
registerattempts=0
defaultexpiry=60
[DID_NUMBER]
type=peer
context=default
host=SERVICE_PROVIDER_IP
authuser=DID_NUMBER
fromuser=DID_NUMBER
fromdomain=SERVICE_PROVIDER_REALM
remotesecret=SERVICE_PROVIDER_PASSWD
secret=SERVICE_PROVIDER_PASSWD
dtmfmode=rfc2833
disallow=all
allow=ulaw
qualify=yes
I am attempting just to get the starting point where I can direct
users through my asterisk box, but it won't direct users to the 's'
extention, only to some voicemail box. I've removed the voicemail
config.
My extensions.conf is tiny:
[globals]
[general]
[default]
exten => s,1,Wait(1)
exten => s,n,Answer()
exten => s,n,Background(welcome)
exten => s,n,Background(and)
exten => s,n,Background(thank-you-for-calling)
exten => s,n,Background(conference-reservations)
exten => s,n,Waitfor()
exten => s,n,Hangup()
What am I doing wrong here?
Thanks for any help you can give.
Joe
Warren Selby
2010-Aug-05  02:04 UTC
[asterisk-users] Incoming SIP Calls dumped to non-existent VM no matter what extensions.conf setup is used
On Wed, Aug 4, 2010 at 8:52 PM, Joe Wood <schmoe at gmail.com> wrote:> Hello. > > I have been beating my head over this problem for about 6 hours now. > > I have a SIP peer, who I register to (successfully), who should be > directing all incoming calls at my [default] stanza in my > extensions.conf: > > [ Context 'default' created by 'pbx_config' ] > 's' => 1. Wait(1) > [pbx_config] > 2. Answer() > [pbx_config] > 3. Background(welcome) > [pbx_config] > 4. Background(and) > [pbx_config] > 5. Background(thank-you-for-calling) > [pbx_config] > 6. Background(conference-reservations) > [pbx_config] > 7. Waitfor() > [pbx_config] > 8. Hangup() > [pbx_config] > > Unfortunately, no matter how I configure extensions.conf or sip.conf, > the phone call always ends up saying: "Extension is unavailable. > Please leave your message after the tone". > > sip.conf: > > [general] > register => NPANXXZZZZ:PASSWORD at SERVICE_PROVIDER_IP > registertimeout=29 > registerattempts=0 > defaultexpiry=60 > > [DID_NUMBER] > type=peer > context=default > host=SERVICE_PROVIDER_IP > authuser=DID_NUMBER > fromuser=DID_NUMBER > fromdomain=SERVICE_PROVIDER_REALM > remotesecret=SERVICE_PROVIDER_PASSWD > secret=SERVICE_PROVIDER_PASSWD > dtmfmode=rfc2833 > disallow=all > allow=ulaw > qualify=yes > > I am attempting just to get the starting point where I can direct > users through my asterisk box, but it won't direct users to the 's' > extention, only to some voicemail box. I've removed the voicemail > config. > > My extensions.conf is tiny: > > [globals] > > [general] > > [default] > exten => s,1,Wait(1) > exten => s,n,Answer() > exten => s,n,Background(welcome) > exten => s,n,Background(and) > exten => s,n,Background(thank-you-for-calling) > exten => s,n,Background(conference-reservations) > exten => s,n,Waitfor() > exten => s,n,Hangup() > > > What am I doing wrong here? > > > > Thanks for any help you can give. > > > Joe >You don't have any extensions in your default context that match the extension that your sip peer is dialing in on. 's' is not a default extension for SIP...try using _X., and see what you get. Bump up the CLI (core set verbose 10) and then repost a failed called attempt. Some SIP providers also use a + symbol in front of their inbound calls, so you may need to use _+X., instead. -- Thanks, --Warren Selby http://www.selbytech.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100804/64e40eaa/attachment.htm