[asterisk-users] *8 and SIP

[Nick Olsen]

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

 
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