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:*[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 From: "nicktest" <sip:[email protected]>;tag=1470823868 To: <sip:*[email protected]> Call-ID: [email protected] CSeq: 1 INVITE Contact: <sip:[email protected]: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 - [email protected] 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:[email protected]: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:[email protected]>;tag=1470823868 To: <sip:*[email protected]> Call-ID: [email protected] 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:*[email protected]> Content-Length: 0 <------------> Scheduling destruction of SIP dialog '[email protected]' 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:[email protected]>;tag=1470823868 To: <sip:*[email protected]>;tag=as65ceb9be Call-ID: [email protected] 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:*[email protected]:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.10.101:5063;branch=z9hG4bK908225576 From: "nicktest" <sip:[email protected]>;tag=1470823868 To: <sip:*[email protected]>;tag=as65ceb9be Call-ID: [email protected] CSeq: 1 ACK Content-Length: 0 <-------------> Nick Olsen Network Operations (855) FLSPEED x106
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