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.. > > If anyone has any ideas, Please let me know. Thanks! > > SIP Trace Below > > <--- SIP read from UDP:208.65.55.170:5063 ---> > INVITE sip:*8@10.65.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 > From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868 > To: <sip:*8@10.65.6.10> > Call-ID: 695101044@172.16.10.101 > CSeq: 1 INVITE > Contact: <sip:nicktest@172.16.10.101:5063> > Content-Type: application/sdp > Allow: INVITE, INFO, PRACK, ACK, BYE, CANCEL, OPTIONS, NOTIFY, > REGISTER, SUBSCRIBE, REFER, PUBLISH, UPDATE, MESSAGE > Max-Forwards: 70 > User-Agent: Yealink SIP-T46G 28.71.0.180 > Supported: replaces > Allow-Events: talk,hold,conference,refer,check-sync > Content-Length: 308 > > v=0 > o=- 20402 20402 IN IP4 172.16.10.101 > s=SDP data > c=IN IP4 172.16.10.101 > t=0 0 > m=audio 11792 RTP/AVP 0 8 18 9 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:9 G722/8000 > a=fmtp:101 0-15 > a=rtpmap:101 telephone-event/8000 > a=ptime:20 > a=sendrecv > > <-------------> > --- (14 headers 15 lines) --- > == Using SIP RTP CoS mark 5 > Using INVITE request as basis request - 695101044@172.16.10.101 > Found peer 'nicktest' for 'nicktest' from 208.65.55.170:5063 > Found RTP audio format 0 > Found RTP audio format 8 > Found RTP audio format 18 > Found RTP audio format 9 > Found RTP audio format 101 > Found audio description format PCMU for ID 0 > Found audio description format PCMA for ID 8 > Found audio description format G729 for ID 18 > Found audio description format G722 for ID 9 > Found audio description format telephone-event for ID 101 > Capabilities: us - 0x4 (ulaw), peer - audio=0x110c > (ulaw|alaw|g729|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined > - 0x4 (ulaw) > Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 > (telephone-event), combined - 0x1 (telephone-event) > Peer audio RTP is at port 172.16.10.101:11792 > Looking for *8 in trunk_office (domain 10.65.6.10) > list_route: hop: <sip:nicktest@172.16.10.101:5063> > > <--- Transmitting (NAT) to 208.65.55.170:5063 ---> > SIP/2.0 100 Trying > Via: SIP/2.0/UDP > 172.16.10.101:5063;branch=z9hG4bK908225576;received=208.65.55.170 > From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868 > To: <sip:*8@10.65.6.10> > Call-ID: 695101044@172.16.10.101 > CSeq: 1 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Contact: <sip:*8@10.65.6.10> > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog '695101044@172.16.10.101' in 6400 > ms (Method: INVITE) > > <--- Reliably Transmitting (NAT) to 208.65.55.170:5063 ---> > SIP/2.0 403 Forbidden > Via: SIP/2.0/UDP > 172.16.10.101:5063;branch=z9hG4bK908225576;received=208.65.55.170 > From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868 > To: <sip:*8@10.65.6.10>;tag=as65ceb9be > Call-ID: 695101044@172.16.10.101 > CSeq: 1 INVITE > Server: Asterisk PBX 1.6.2.20 > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO > Supported: replaces, timer > Content-Length: 0 > > > <------------> > > <--- SIP read from UDP:208.65.55.170:5063 ---> > ACK sip:*8@10.65.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 > From: "nicktest" <sip:nicktest@10.65.6.10>;tag=1470823868 > To: <sip:*8@10.65.6.10>;tag=as65ceb9be > Call-ID: 695101044@172.16.10.101 > CSeq: 1 ACK > Content-Length: 0 > > > <-------------> > > Nick Olsen > Network Operations > (855) FLSPEED x106 > > >
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