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