[asterisk-users] *8 and SIP

Nick Olsen
Tue, 31 Dec 2013 16:23:45 -0000

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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