Voicemail problems please help

Asterisk Asterisk Support
1

My voicemail used to work perfect i have 7 extensions set up using Polycom 430’s all of them being set up the same exact way with time one by one the voicemail stopped working for all of them without making any changes in the configuration files. When i dial *97 i get the voice prompt to enter password and after i enter it i cannot hear anything the call just stays on without hearing anything.
Asterisk version 1.2.9
This is what i get in console with max debug
1 -

Core debug was 99999999 and is now 10

1	+

<-- SIP read from 192.168.123.202:5060:

2 -

-- Executing VoiceMailMain("SIP/704-0e45", "704") in new stack

2	+

INVITE sip:*97@192.168.123.56:5060 SIP/2.0

3 -

-- Playing 'vm-password' (language 'en')

that’s it and it stays there as long as i leave the call on

Here’s what i get with sip debug

1 +

<-- SIP read from 192.168.123.202:5060:

2 -

-- Executing VoiceMailMain("SIP/704-0e45", "704") in new stack

2	+

INVITE sip:*97@192.168.123.56:5060 SIP/2.0

3 -

-- Playing 'vm-password' (language 'en')

3	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bKd4ed47b7BA947A86

4	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

5	+

To: sip:*97@192.168.123.56

6	+

CSeq: 1 INVITE

7	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

8	+

Contact: sip:704@192.168.123.202

9	+

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

10	+

User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094

11	+

Supported: 100rel,replaces

12	+

Allow-Events: talk,hold,conference

13	+

Max-Forwards: 70

14	+

Content-Type: application/sdp

15	+

Content-Length: 253

17	+

v=0

18	+

o=- 978312888 978312888 IN IP4 192.168.123.202

19	+

s=Polycom IP Phone

20	+

c=IN IP4 192.168.123.202

21	+

t=0 0

22	+

a=sendrecv

23	+

m=audio 2228 RTP/AVP 0 8 18 101

24	+

a=rtpmap:0 PCMU/8000

25	+

a=rtpmap:8 PCMA/8000

26	+

a=rtpmap:18 G729/8000

27	+

a=rtpmap:101 telephone-event/8000

29	+

— (14 headers 11 lines)—

30	+

Using INVITE request as basis request - e2af03e5-e95f02d4-2e102653@192.168.123.202

31	+

Sending to 192.168.123.202 : 5060 (non-NAT)

32	+

Reliably Transmitting (no NAT) to 192.168.123.202:5060:

33	+

SIP/2.0 407 Proxy Authentication Required

34	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bKd4ed47b7BA947A86;received=192.168.123.202

35	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

36	+

To: sip:*97@192.168.123.56;tag=as13ccf307

37	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

38	+

CSeq: 1 INVITE

39	+

User-Agent: Asterisk PBX

40	+

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

41	+

Contact: sip:*97@192.168.123.56

42	+

Proxy-Authenticate: Digest realm=“asterisk”, nonce=“6561c784”

43	+

Content-Length: 0

46	+	

47	+

Scheduling destruction of call ‘e2af03e5-e95f02d4-2e102653@192.168.123.202’ in 15000 ms

48	+

Found user ‘704’

49	+

asterisk1*CLI>

50	+

<-- SIP read from 192.168.123.202:5060:

51	+

ACK sip:*97@192.168.123.56:5060 SIP/2.0

52	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bKd4ed47b7BA947A86

53	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

54	+

To: sip:*97@192.168.123.56;tag=as13ccf307

55	+

CSeq: 1 ACK

56	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

57	+

Contact: sip:704@192.168.123.202

58	+

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

59	+

User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094

60	+

Max-Forwards: 70

61	+

Content-Length: 0

64	+

— (11 headers 0 lines)—

65	+

asterisk1*CLI>

66	+

<-- SIP read from 192.168.123.202:5060:

67	+

INVITE sip:*97@192.168.123.56:5060 SIP/2.0

68	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bK63b9b23094607DEF

69	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

70	+

To: sip:*97@192.168.123.56

71	+

CSeq: 2 INVITE

72	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

73	+

Contact: sip:704@192.168.123.202

74	+

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

75	+

User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094

76	+

Supported: 100rel,replaces

77	+

Allow-Events: talk,hold,conference

78	+

Proxy-Authorization: Digest username=“704”, realm=“asterisk”, nonce=“6561c784”, uri=“sip:*97@192.168.123.56:5060”, response=“83eac4b09b67e0d3fe28af906e5e0a80”, algorithm=MD5

79	+

Max-Forwards: 70

80	+

Content-Type: application/sdp

81	+

Content-Length: 253

83	+

v=0

84	+

o=- 978312888 978312888 IN IP4 192.168.123.202

85	+

s=Polycom IP Phone

86	+

c=IN IP4 192.168.123.202

87	+

t=0 0

88	+

a=sendrecv

89	+

m=audio 2228 RTP/AVP 0 8 18 101

90	+

a=rtpmap:0 PCMU/8000

91	+

a=rtpmap:8 PCMA/8000

92	+

a=rtpmap:18 G729/8000

93	+

a=rtpmap:101 telephone-event/8000

95	+

— (15 headers 11 lines)—

96	+

Using INVITE request as basis request - e2af03e5-e95f02d4-2e102653@192.168.123.202

97	+

Sending to 192.168.123.202 : 5060 (non-NAT)

98	+

Found user ‘704’

99	+

Found RTP audio format 0

100	+

Found RTP audio format 8

101	+

Found RTP audio format 18

102	+

Found RTP audio format 101

103	+

Peer audio RTP is at port 192.168.123.202:2228

104	+

Found description format PCMU

105	+

Found description format PCMA

106	+

Found description format G729

107	+

Found description format telephone-event

108	+

Capabilities: us - 0x108 (alaw|g729), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x108 (alaw|g729)

109	+

Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)

110	+

Looking for *97 in from-internal (domain 192.168.123.56)

111	+

list_route: hop: sip:704@192.168.123.202

112	+

Transmitting (no NAT) to 192.168.123.202:5060:

113	+

SIP/2.0 100 Trying

114	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bK63b9b23094607DEF;received=192.168.123.202

115	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

116	+

To: sip:*97@192.168.123.56

117	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

118	+

CSeq: 2 INVITE

119	+

User-Agent: Asterisk PBX

120	+

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

121	+

Contact: sip:*97@192.168.123.56

122	+

Content-Length: 0

125	+	

126	+	

-- Executing VoiceMailMain("SIP/704-4f80", "704") in new stack

127	+

We’re at 192.168.123.56 port 10388

128	+

Adding codec 0x8 (alaw) to SDP

129	+

Adding codec 0x100 (g729) to SDP

130	+

Adding non-codec 0x1 (telephone-event) to SDP

131	+

Reliably Transmitting (no NAT) to 192.168.123.202:5060:

132	+

SIP/2.0 200 OK

133	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bK63b9b23094607DEF;received=192.168.123.202

134	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

135	+

To: sip:*97@192.168.123.56;tag=as2b733235

136	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

137	+

CSeq: 2 INVITE

138	+

User-Agent: Asterisk PBX

139	+

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

140	+

Contact: sip:*97@192.168.123.56

141	+

Content-Type: application/sdp

142	+

Content-Length: 265

144	+

v=0

145	+

o=root 2402 2402 IN IP4 192.168.123.56

146	+

s=session

147	+

c=IN IP4 192.168.123.56

148	+

t=0 0

149	+

m=audio 10388 RTP/AVP 8 18 101

150	+

a=rtpmap:8 PCMA/8000

151	+

a=rtpmap:18 G729/8000

152	+

a=fmtp:18 annexb=no

153	+

a=rtpmap:101 telephone-event/8000

154	+

a=fmtp:101 0-16

155	+

a=silenceSupp:off - - - -

157	+	

158	+	

-- Playing 'vm-password' (language 'en')

159	+

asterisk1*CLI>

160	+

<-- SIP read from 192.168.123.202:5060:

161	+

ACK sip:*97@192.168.123.56 SIP/2.0

162	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bK4a380e8b316C0E1A

163	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

164	+

To: sip:*97@192.168.123.56;tag=as2b733235

165	+

CSeq: 2 ACK

166	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

167	+

Contact: sip:704@192.168.123.202

168	+

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER

169	+

User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094

170	+

Proxy-Authorization: Digest username=“704”, realm=“asterisk”, nonce=“6561c784”, uri=“sip:*97@192.168.123.56:5060”, response=“83eac4b09b67e0d3fe28af906e5e0a80”, algorithm=MD5

171	+

Max-Forwards: 70

172	+

Content-Length: 0

175	+

— (12 headers 0 lines)—

176	+

asterisk1*CLI>

177	+

<-- SIP read from 192.168.123.202:5060:

178	+

BYE sip:*97@192.168.123.56 SIP/2.0

179	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bKf777f839A94EABE8

180	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

181	+

To: sip:*97@192.168.123.56;tag=as2b733235

182	+

CSeq: 3 BYE

183	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

184	+

Contact: sip:704@192.168.123.202

185	+

User-Agent: PolycomSoundPointIP-SPIP_430-UA/1.6.7.0094

186	+

Proxy-Authorization: Digest username=“704”, realm=“asterisk”, nonce=“6561c784”, uri=“sip:*97@192.168.123.56:5060”, response=“a75bbbde2c6790f43559a1a97244a796”, algorithm=MD5

187	+

Max-Forwards: 70

188	+

Content-Length: 0

191	+

— (11 headers 0 lines)—

192	+

Sending to 192.168.123.202 : 5060 (non-NAT)

193	+

Transmitting (no NAT) to 192.168.123.202:5060:

194	+

SIP/2.0 200 OK

195	+

Via: SIP/2.0/UDP 192.168.123.202;branch=z9hG4bKf777f839A94EABE8;received=192.168.123.202

196	+

From: “704” sip:704@192.168.123.56;tag=1BE6AD62-BEBE6701

197	+

To: sip:*97@192.168.123.56;tag=as2b733235

198	+

Call-ID: e2af03e5-e95f02d4-2e102653@192.168.123.202

199	+

CSeq: 3 BYE

200	+

ser-Agent: Asterisk PBX

201	+

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY

202	+

Contact: sip:*97@192.168.123.56

203	+

Content-Length: 0

206	+

Any ideas?I’m desperate

2

Can anybody try to help me please ?

3

Real strange. I would try upgrading to the latest version of 1.2.X and see what happens. Also what are you using for DTMF ?

4

Check your voice recordings. They may have been corrupted. Re-copy them over the files that you have.

Also check to make sure that the codec module for the voice recordings has been loaded properly. (Reload it or specifically load it if necessary)