When I try to check voicemail by dialing 8500 I get no audio. I’m a newbie to this so it may be something silly like the phone config. Here is the output from the sip debug:
minint-hd3fuqac*CLI>
SIP Debugging enabled
minint-hd3fuqac*CLI>
<— SIP read from 192.168.192.74:5060 —>
INVITE sip:8500@192.168.192.141:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK369c69433E5A5626
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone
CSeq: 1 INVITE
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
Contact: sip:6002@192.168.192.74
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 228
v=0
o=- 1175178588 1175178588 IN IP4 192.168.192.74
s=Polycom IP Phone
c=IN IP4 192.168.192.74
t=0 0
a=sendrecv
m=audio 10008 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
<------------->
minint-hd3fuqac*CLI>
— (14 headers 10 lines) —
minint-hd3fuqac*CLI>
Sending to 192.168.192.74 : 5060 (no NAT)
minint-hd3fuqac*CLI>
Using INVITE request as basis request - 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
minint-hd3fuqac*CLI>
Found peer ‘6002’
minint-hd3fuqac*CLI>
<— Reliably Transmitting (no NAT) to 192.168.192.74:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK369c69433E5A5626;received=192.168.192.74
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone;tag=as0b5a50a7
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“7eca8fe2”
Content-Length: 0
<------------>
minint-hd3fuqac*CLI>
Scheduling destruction of SIP dialog ‘1bf9cff5-6c0e3be0-fab209d7@192.168.192.74’ in 32000 ms (Method: INVITE)
minint-hd3fuqac*CLI>
<— SIP read from 192.168.192.74:5060 —>
ACK sip:8500@192.168.192.141:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK369c69433E5A5626
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone;tag=as0b5a50a7
CSeq: 1 ACK
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
Contact: sip:6002@192.168.192.74
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Max-Forwards: 70
Content-Length: 0
<------------->
minint-hd3fuqac*CLI>
— (11 headers 0 lines) —
minint-hd3fuqac*CLI>
<— SIP read from 192.168.192.74:5060 —>
INVITE sip:8500@192.168.192.141:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK4283e3a4FA5052B
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone
CSeq: 2 INVITE
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
Contact: sip:6002@192.168.192.74
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Proxy-Authorization: Digest username=“6002”, realm=“asterisk”, nonce=“7eca8fe2”, uri=“sip:8500@192.168.192.141:5060;user=phone”, response=“a9b87fddd38ce072b2479338d193a779”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 228
v=0
o=- 1175178588 1175178588 IN IP4 192.168.192.74
s=Polycom IP Phone
c=IN IP4 192.168.192.74
t=0 0
a=sendrecv
m=audio 10008 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
<------------->
minint-hd3fuqac*CLI>
— (15 headers 10 lines) —
Sending to 192.168.192.74 : 5060 (no NAT)
Using INVITE request as basis request - 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
minint-hd3fuqac*CLI>
Found peer ‘6002’
minint-hd3fuqac*CLI>
Found RTP audio format 0
minint-hd3fuqac*CLI>
Found RTP audio format 8
Found RTP audio format 101
minint-hd3fuqac*CLI>
Peer audio RTP is at port 192.168.192.74:10008
Found description format PCMU for ID 0
minint-hd3fuqac*CLI>
Found description format PCMA for ID 8
Found description format telephone-event for ID 101
minint-hd3fuqac*CLI>
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
minint-hd3fuqac*CLI>
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 192.168.192.74:10008
minint-hd3fuqac*CLI>
Looking for 8500 in numberplan-custom-1 (domain 192.168.192.141)
minint-hd3fuqac*CLI>
list_route: hop: sip:6002@192.168.192.74
minint-hd3fuqac*CLI>
<— Transmitting (no NAT) to 192.168.192.74:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK4283e3a4FA5052B;received=192.168.192.74
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:8500@192.168.192.141
Content-Length: 0
minint-hd3fuqac*CLI>
<------------>
minint-hd3fuqac*CLI>
– Executing [8500@numberplan-custom-1:1] VoiceMailMain(“SIP/6002-b5e16648”, “”) in new stack
minint-hd3fuqac*CLI>
Audio is at 192.168.192.141 port 11284
minint-hd3fuqac*CLI>
Adding codec 0x4 (ulaw) to SDP
minint-hd3fuqac*CLI>
Adding codec 0x8 (alaw) to SDP
minint-hd3fuqac*CLI>
Adding non-codec 0x1 (telephone-event) to SDP
minint-hd3fuqac*CLI>
<— Reliably Transmitting (no NAT) to 192.168.192.74:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK4283e3a4FA5052B;received=192.168.192.74
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone;tag=as30181dc0
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:8500@192.168.192.141
Content-Type: application/sdp
Content-Length: 268
v=0
o=root 5941 5941 IN IP4 192.168.192.141
s=session
c=IN IP4 192.168.192.141
t=0 0
m=audio 11284 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------>
minint-hd3fuqac*CLI>
– Playing ‘vm-login’ (language ‘en’)
minint-hd3fuqac*CLI>
<— SIP read from 192.168.192.74:5060 —>
ACK sip:8500@192.168.192.141 SIP/2.0
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK8f9d6f3fFD08B792
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone;tag=as30181dc0
CSeq: 2 ACK
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
Contact: sip:6002@192.168.192.74
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Proxy-Authorization: Digest username=“6002”, realm=“asterisk”, nonce=“7eca8fe2”, uri=“sip:8500@192.168.192.141:5060;user=phone”, response=“a9b87fddd38ce072b2479338d193a779”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
minint-hd3fuqac*CLI>
— (12 headers 0 lines) —
minint-hd3fuqac*CLI>
<— SIP read from 192.168.192.74:5060 —>
BYE sip:8500@192.168.192.141 SIP/2.0
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK9ce4f8915DECFE6C
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone;tag=as30181dc0
CSeq: 3 BYE
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
Contact: sip:6002@192.168.192.74
User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094
Proxy-Authorization: Digest username=“6002”, realm=“asterisk”, nonce=“7eca8fe2”, uri=“sip:8500@192.168.192.141:5060;user=phone”, response=“0a8a62af2df1c1d43cc250977cabeb01”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0
<------------->
minint-hd3fuqac*CLI>
— (11 headers 0 lines) —
Sending to 192.168.192.74 : 5060 (no NAT)
minint-hd3fuqac*CLI>
<— Transmitting (no NAT) to 192.168.192.74:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.192.74;branch=z9hG4bK9ce4f8915DECFE6C;received=192.168.192.74
From: “6002” sip:6002@192.168.192.141;tag=3881778A-F535BF69
To: sip:8500@192.168.192.141;user=phone;tag=as30181dc0
Call-ID: 1bf9cff5-6c0e3be0-fab209d7@192.168.192.74
CSeq: 3 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:8500@192.168.192.141
Content-Length: 0
<------------>
minint-hd3fuqac*CLI>
[Mar 29 08:32:24] WARNING[6766]: app_voicemail.c:6006 vm_authenticate: Couldn’t read username
minint-hd3fuqac*CLI>
Really destroying SIP dialog ‘1bf9cff5-6c0e3be0-fab209d7@192.168.192.74’ Method: BYE
minint-hd3fuqac*CLI>
Really destroying SIP dialog ‘c949f1e7-7253f1da-7dc51cf9@192.168.192.74’ Method: REGISTER
minint-hd3fuqac*CLI> sip set debug off
minint-hd3fuqac*CLI>
SIP Debugging Disabled
minint-hd3fuqac*CLI>
– Executing [8500@numberplan-custom-1:1] VoiceMailMain(“SIP/6002-b5e04e70”, “”) in new stack
minint-hd3fuqac*CLI>
– Playing ‘vm-login’ (language ‘en’)
minint-hd3fuqac*CLI>
[Mar 29 08:32:42] WARNING[6773]: app_voicemail.c:6006 vm_authenticate: Couldn’t read username
minint-hd3fuqac*CLI>