[asterisk-users] Voicemail from sip phones

Sun Jan 28 19:55:56 MST 2007

I don't see anything apparent in for the SIP phones that would indicate that *8 is 
anything but VoiceMailMain().  I looked in extensions.conf for it, as well as sip.conf 
(the config that I pasted in a previous e-mail.  It does indeed appear that a literal 
*8 is being passed to asterisk.

Here's the relevant context of extensions.conf:

[internal]
;downstairs office
exten => 7191,1,Dial(Zap/1)
;cordless
exten => 7192,1,Dial(Zap/2)
;second cordless
exten => 7193,1,Dial(SIP/ht3861)
;call a couple phones
exten => 7194,1,Dial(SIP/gxp2&SIP/user)
;goto voicemail
exten => *8,1,VoiceMailMain()
include => local
include => ld

The sip.conf bits are contained in my initial post.

Here's the debug output from the console, it's somewhat long.  Could the key line be 
(towards the bottom) this?

[Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite: Nothing to
pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.

Here's the full output:

<--- SIP read from 192.168.1.165:5806 --->
INVITE sip:*8 at 192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:user at 192.168.1.165:5806>
To: "*8"<sip:*8 at 192.168.1.2>
From: "SIP User"<sip:user at 192.168.1.2>;tag=2a71c861
Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Content-Length: 328

v=0
o=- 6 2 IN IP4 192.168.1.165
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.165
t=0 0
m=audio 35172 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : UGnBtE65 8rk0z5iz 192.168.1.165 35172
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (12 headers 13 lines) ---
Sending to 192.168.1.165 : 5806 (NAT)
Using INVITE request as basis request - OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
Found user 'user' for 'user'
ord*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.165:5806 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;received=192.168.1.165;rport=5806
From: "SIP User"<sip:user at 192.168.1.2>;tag=2a71c861
To: "*8"<sip:*8 at 192.168.1.2>;tag=as31c6aff1
Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="69d017e6"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.' in 
32000 ms (Method: INVITE)
ord*CLI>
<--- SIP read from 192.168.1.165:5806 --->
ACK sip:*8 at 192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.165:5806;branch=z9hG4bK-d87543-42396923770b6e49-1--d87543-;rport
To: "*8"<sip:*8 at 192.168.1.2>;tag=as31c6aff1
From: "SIP User"<sip:user at 192.168.1.2>;tag=2a71c861
Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
ord*CLI>
<--- SIP read from 192.168.1.165:5806 --->
INVITE sip:*8 at 192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;rport
Max-Forwards: 70
Contact: <sip:user at 192.168.1.165:5806>
To: "*8"<sip:*8 at 192.168.1.2>
From: "SIP User"<sip:user at 192.168.1.2>;tag=2a71c861
Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
Content-Type: application/sdp
User-Agent: X-Lite release 1006e stamp 34025
Authorization: Digest 
username="user",realm="asterisk",nonce="69d017e6",uri="sip:*8 at 192.168.1.2",response="b82d4700cc1f72ef2711df0b597e7184",algorithm=MD5
Content-Length: 328

v=0
o=- 6 2 IN IP4 192.168.1.165
s=CounterPath X-Lite 3.0
c=IN IP4 192.168.1.165
t=0 0
m=audio 35172 RTP/AVP 107 119 0 98 8 3 101
a=alt:1 1 : UGnBtE65 8rk0z5iz 192.168.1.165 35172
a=fmtp:101 0-15
a=rtpmap:107 BV32/16000
a=rtpmap:119 BV32-FEC/16000
a=rtpmap:98 iLBC/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv

<------------->
--- (13 headers 13 lines) ---
Sending to 192.168.1.165 : 5806 (NAT)
Using INVITE request as basis request - OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
Found user 'user' for 'user'
Found RTP audio format 107
Found RTP audio format 119
Found RTP audio format 0
Found RTP audio format 98
Found RTP audio format 8
Found RTP audio format 3
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.165:35172
Found description format BV32 for ID 107
Found description format BV32-FEC for ID 119
Found description format iLBC for ID 98
Found description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x40e 
(gsm|ulaw|alaw|ilbc)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 
(telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.1.165:35172
Looking for *8 in internal (domain 192.168.1.2)
list_route: hop: <sip:user at 192.168.1.165:5806>
ord*CLI>
<--- Transmitting (no NAT) to 192.168.1.165:5806 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;received=192.168.1.165;rport=5806
From: "SIP User"<sip:user at 192.168.1.2>;tag=2a71c861
To: "*8"<sip:*8 at 192.168.1.2>
Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*8 at 192.168.1.2>
Content-Length: 0

<------------>
[Jan 28 20:39:10] NOTICE[31924]: chan_sip.c:13519 handle_request_invite: Nothing to 
pick up for OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
ord*CLI>
<--- Reliably Transmitting (no NAT) to 192.168.1.165:5806 --->
SIP/2.0 503 Unavailable
Via: SIP/2.0/UDP 
192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;received=192.168.1.165;rport=5806
From: "SIP User"<sip:user at 192.168.1.2>;tag=2a71c861
To: "*8"<sip:*8 at 192.168.1.2>;tag=as235f0837
Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:*8 at 192.168.1.2>
Content-Length: 0

<------------>
ord*CLI>
<--- SIP read from 192.168.1.165:5806 --->
ACK sip:*8 at 192.168.1.2 SIP/2.0
Via: SIP/2.0/UDP 
192.168.1.165:5806;branch=z9hG4bK-d87543-dc51447ec8163e0a-1--d87543-;rport
To: "*8"<sip:*8 at 192.168.1.2>;tag=as235f0837
From: "SIP User"<sip:user at 192.168.1.2>;tag=2a71c861
Call-ID: OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.
CSeq: 2 ACK
Content-Length: 0

<------------->
--- (7 headers 0 lines) ---
Really destroying SIP dialog 'OWE2MmFkYzYxYmYwNGM2ZGI4ODYzYWU4ODU2MzNhNmI.' Method: 
ACK

On Mon, Jan 29, 2007 at 09:59:14AM +0800, Leo Ann Boon wrote:
> check that your phone is not using *8 in its own dial plan. Also, do a 
> sip debug and see that the phone is actually sending *8 to asterisk.
> 
> Leo
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