Polycom IP430 no sound from asterisk server

aorszula · March 29, 2007, 2:39pm

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>
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>

baconbuttie · March 29, 2007, 2:53pm

what codecs do you have setup in sip.conf in [general] and for the phone in question ?

aorszula · March 29, 2007, 3:12pm

G711u and G711a on the phones

i changed sip.conf to
allow=all

Bender · March 29, 2007, 9:07pm

Try

pedantic=yes

aorszula · March 30, 2007, 1:29pm

I set pedantic = yes and restarted asterisk. still does not work

Bender · March 30, 2007, 3:00pm

On each of the two individual channels you are dealing with try this each one of this one at a time. After each change, instead of restarting Asterisk go to your CLI> sip reload. and your sip file will reload without having to restart asterisk.

[8500]
pedantic=yes ; try this first
canreinvite=no
dtmfmode=rfc2833
progressinband=no

[6002]
pedantic=yes
canreinvite=no
dtmfmode=rfc2833
progressinband=no

aorszula · March 30, 2007, 3:27pm

Do I do this in users.conf or sip.conf?