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 at 10.65.6.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868 To: <sip:*8 at 10.65.6.10> Call-ID: 695101044 at 172.16.10.101 CSeq: 1 INVITE Contact: <sip:nicktest at 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 at 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 at 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 at 10.65.6.10>;tag=1470823868 To: <sip:*8 at 10.65.6.10> Call-ID: 695101044 at 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 at 10.65.6.10> Content-Length: 0 <------------> Scheduling destruction of SIP dialog '695101044 at 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 at 10.65.6.10>;tag=1470823868 To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be Call-ID: 695101044 at 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 at 10.65.6.10:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868 To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be Call-ID: 695101044 at 172.16.10.101 CSeq: 1 ACK Content-Length: 0 <-------------> Nick Olsen Network Operations (855) FLSPEED x106 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131231/2b7aa8d8/attachment.html>
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 at 10.65.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 > From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10> > Call-ID: 695101044 at 172.16.10.101 > CSeq: 1 INVITE > Contact: <sip:nicktest at 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 at 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 at 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 at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10> > Call-ID: 695101044 at 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 at 10.65.6.10> > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog '695101044 at 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 at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be > Call-ID: 695101044 at 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 at 10.65.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 > From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be > Call-ID: 695101044 at 172.16.10.101 > CSeq: 1 ACK > Content-Length: 0 > > > <-------------> > > Nick Olsen > Network Operations > (855) FLSPEED x106 > > >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20131231/b443df2f/attachment.html>
On 12/31/13, 11:23 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.I suggest trying command 'features show' to pinpoint the conflict. # asterisk -rx 'features show' Builtin Feature Default Current --------------- ------- ------- Pickup *8 Blind Transfer # # Attended Transfer One Touch Monitor Disconnect Call * * Park Call One Touch MixMonitor Dynamic Feature Default Current --------------- ------- ------- (none) Feature Groups: --------------- (none) Call parking (Parking lot: default) ------------ Parking extension : 700 Parking context : parkedcalls Parked call extensions: 701-750 Parkingtime : 45000 ms MusicOnHold class : default Enabled : Yes> > 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 at 10.65.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 > From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10> > Call-ID: 695101044 at 172.16.10.101 > CSeq: 1 INVITE > Contact: <sip:nicktest at 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 at 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 at 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 at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10> > Call-ID: 695101044 at 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 at 10.65.6.10> > Content-Length: 0 > > > <------------> > Scheduling destruction of SIP dialog '695101044 at 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 at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be > Call-ID: 695101044 at 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 at 10.65.6.10:5060 SIP/2.0 > Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 > From: "nicktest" <sip:nicktest at 10.65.6.10>;tag=1470823868 > To: <sip:*8 at 10.65.6.10>;tag=as65ceb9be > Call-ID: 695101044 at 172.16.10.101 > CSeq: 1 ACK > Content-Length: 0 > > > <-------------> > > Nick Olsen > Network Operations > (855) FLSPEED x106 > > >-- Technical Support http://www.cellroute.net -------------- next part -------------- An HTML attachment was scrubbed... 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