Dialplan is solid.. exten => *8,1,VoicemailMain(@default) exten => *8,2,Hangup exten => 88,1,VoicemailMain(@default) exten => 88,2,Hangup
Also tried "_*8" in the dialplan at the request of a fellow BOFH. Using 88 temporarily, Which works fine. Also, DAHDI dumps into the same context and has no issue. I did indeed restart the service after any features change. I always run my CLI with about 8 million v's, But still don't get any useful feedback on this issue. I understand I can easily change the voicemail number. But this customer (hotel) has the voicemail number printed on in-room cards. So I'm hoping not to cause them a re-print. Nick Olsen Network Operations (855) FLSPEED x106 ---------------------------------------- From: "Adrian Serafini" <[email protected]> Sent: Tuesday, December 31, 2013 12:51 PM To: [email protected] Subject: Re: [asterisk-users] *8 and SIP On 12/31/2013 12:41 PM, Vladimir Mikhelson wrote: > Nick, > > You may want to try *97 and *98 to access voice mail. > > Regards, > Vladimir > > > On 12/31/2013 10:23 AM, Nick Olsen wrote: >> Greetings all, First time poster, Sorry if this has been answered here >> before. >> >> We recently replaced a failed 1.4x asterisk PBX at a customer location. >> >> Voicemail access was setup when the customer dialed *8, This worked in >> 1.4. >> >> Now, Running 1.6 (I know it's old I had to load it quickly, And that's >> what I got working first. It'll get upgraded to 1.8 soon). >> >> The strange part is *8 no longer works. >> The only CLI feedback I get is "== Using SIP RTP CoS mark 5" >> >> In features.conf, Callpickup *8 is commented out, But just incase I >> also changed it to *7 (We don't use that feature). >> >> It appears to be something completely SIP based, As if the call >> originates from DAHDI, It works fine.. Maybe it's a context issue. Check the dialplan context for the *8 logic. Crank up the verbosity of the CLI and make a test call. You might have to reboot after the features.conf change. Adrian -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
