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*77 Service not available


nathans
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This one is trance as I'm pretty sure it was working and somehow we did something and it is not longer working. Basically we are trying to use the default star code for call forwarding *77 or call pick up and we get a recording for "service is not available"

 

On the snomone trace below extension 9999 is trying to forward a call to extension 45.

 

 

 

 

7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024:

INVITE sip:*7745@192.168.30.2:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport

From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9

To: <sip:*7745@192.168.30.2:5060;user=phone>

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1

X-Serialnumber: 000413412E6B

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom870/8.4.33

Accept: application/sdp

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

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Call-Info: <sip:187.174.100.120>;appearance-index=1

Session-Expires: 3600;refresher=uas

Min-SE: 90

Content-Type: application/sdp

Content-Length: 528

 

v=0

o=root 100582088 100582088 IN IP4 75.149.181.121

s=call

c=IN IP4 75.149.181.121

t=0 0

m=audio 53466 RTP/AVP 0 8 9 99 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XzhHQKdn2K8ILVlcwvTbc6XTO0iWl7pmUhQ4ijEM

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:99 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv

[7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport=1024

From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9

To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 1 INVITE

Content-Length: 0

 

[7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024:

SIP/2.0 401 Authentication Required

Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport=1024

From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9

To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 1 INVITE

User-Agent: snom-PBX/2011-4.2.1.4025

WWW-Authenticate: Digest realm="192.168.30.2",nonce="059ba8b53b1b0982fd128db9b5ef8e58",domain="sip:*7745@192.168.30.2:5060;user=phone",algorithm=MD5

Content-Length: 0

 

[7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024:

ACK sip:*7745@192.168.30.2:5060;user=phone SIP/2.0

Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-iv9kko3xkr2u;rport

From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9

To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1

Content-Length: 0

 

[7] 2012/01/09 17:42:37: SIP Rx udp:75.149.181.121:1024:

INVITE sip:*7745@187.174.100.120;user=phone SIP/2.0

Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport

From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9

To: <sip:*7745@187.174.100.120;user=phone>

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 2 INVITE

Max-Forwards: 70

Contact: <sip:9999@192.168.88.107:3072;line=4r9hwkd9>;reg-id=1

X-Serialnumber: 000413412E6B

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom870/8.4.33

Accept: application/sdp

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

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Call-Info: <sip:187.174.100.120>;appearance-index=1

Session-Expires: 3600;refresher=uas

Min-SE: 90

Authorization: Digest username="9999",realm="192.168.30.2",nonce="059ba8b53b1b0982fd128db9b5ef8e58",uri="sip:*7745@192.168.30.2:5060;user=phone",response="7d4d1b35a0321805252b0311df97c9e3",algorithm=MD5

Content-Type: application/sdp

Content-Length: 528

 

v=0

o=root 100582088 100582088 IN IP4 192.168.88.107

s=call

c=IN IP4 192.168.88.107

t=0 0

m=audio 53466 RTP/AVP 0 8 9 99 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:XzhHQKdn2K8ILVlcwvTbc6XTO0iWl7pmUhQ4ijEM

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:9 G722/8000

a=rtpmap:99 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv

[7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024:

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport=1024;received=75.149.181.121

From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9

To: <sip:*7745@187.174.100.120;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 2 INVITE

Content-Length: 0

 

[7] 2012/01/09 17:42:37: SIP Tx udp:75.149.181.121:1024:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 192.168.88.107:3072;branch=z9hG4bK-gtcchuhh9pd4;rport=1024;received=75.149.181.121

From: "9999 Bondojito" <sip:9999@187.174.100.120>;tag=ngoms7doq9

To: <sip:*7745@187.174.100.120;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 2 INVITE

Contact: <sip:9999@192.168.30.2:5060>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4025

Content-Type: application/sdp

Content-Length: 396

 

v=0

o=- 405392631 405392631 IN IP4 192.168.30.2

s=-

c=IN IP4 192.168.30.2

t=0 0

m=audio 53968 RTP/AVP 0 8 9 18 99 3 101

a=rtpmap:0 pcmu/8000

a=rtpmap:8 pcma/8000

a=rtpmap:9 g722/8000

a=rtpmap:18 g729/8000

a=fmtp:18 annexb=no

a=rtpmap:99 g726-32/8000

a=rtpmap:3 gsm/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:rcvr-rtt=all voip-metrics

a=sendrecv

[7] 2012/01/09 17:42:38: SIP Rx udp:75.149.181.121:1024:

ACK sip:9999@192.168.30.2:5060 SIP/2.0

Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-lh9yrb0744rt;rport

From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9

To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 2 ACK

Max-Forwards: 70

Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1

Content-Length: 0

 

[7] 2012/01/09 17:42:40: SIP Rx udp:75.149.181.121:1024:

BYE sip:9999@192.168.30.2:5060 SIP/2.0

Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-78xiabcq4825;rport

From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9

To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 3 BYE

Max-Forwards: 70

Contact: <sip:9999@75.149.181.121:1024;line=4r9hwkd9>;reg-id=1

User-Agent: snom870/8.4.33

RTP-RxStat: Total_Rx_Pkts=106,Rx_Pkts=106,Rx_Pkts_Lost=1,Remote_Rx_Pkts_Lost=0

RTP-TxStat: Total_Tx_Pkts=165,Tx_Pkts=121,Remote_Tx_Pkts=0

Content-Length: 0

 

[7] 2012/01/09 17:42:40: SIP Tx udp:75.149.181.121:1024:

SIP/2.0 200 Ok

Via: SIP/2.0/UDP 75.149.181.121:1024;branch=z9hG4bK-78xiabcq4825;rport=1024

From: "9999 Bondojito" <sip:9999@192.168.30.2:5060>;tag=ngoms7doq9

To: <sip:*7745@192.168.30.2:5060;user=phone>;tag=ffafb39193

Call-ID: 8633353c102b-7vv98fylg9ww

CSeq: 3 BYE

Contact: <sip:9999@192.168.30.2:5060>

User-Agent: snom-PBX/2011-4.2.1.4025

Content-Length: 0

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  • 2 weeks later...
  • 4 weeks later...

HI,

 

Actually most of the codes requiring an extension like *83 and *77 are the ones not working anymore.

 

I just found out also that the call forwarding also is not working any more.

 

Is this the screen status you wanted? Let me know if you need more infer before rebooting the system. I will try to get a a log later on .

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some log on the snomone of a failed *83 call pick attempt:

 

 

[7] 2012/02/22 13:49:17: SIP Rx tls:192.168.30.77:4422:

INVITE sip:*877121@pbx.bondojito.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay

To: <sip:*877121@pbx.bondojito.com;user=phone>

Call-ID: 17c0263cd8d7-yvp7g9nueny5

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1

X-Serialnumber: 000413457659

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom821/8.4.32

Accept: application/sdp

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

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 524

 

v=0

o=root 367130050 367130050 IN IP4 192.168.30.77

s=call

c=IN IP4 192.168.30.77

t=0 0

m=audio 64274 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:UKfZKYOjMYhdvmxL3Uk8T2mUEmznx5sPoPLZ0/HE

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv

[7] 2012/02/22 13:49:17: SIP Tx tls:192.168.30.77:4422:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport=4422

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay

To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54

Call-ID: 17c0263cd8d7-yvp7g9nueny5

CSeq: 1 INVITE

Content-Length: 0

 

[7] 2012/02/22 13:49:17: SIP Tx tls:192.168.30.77:4422:

SIP/2.0 404 Not Found

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport=4422

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay

To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54

Call-ID: 17c0263cd8d7-yvp7g9nueny5

CSeq: 1 INVITE

Contact: <sip:7120@192.168.30.2:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4025

Content-Length: 0

 

[7] 2012/02/22 13:49:17: SIP Rx tls:192.168.30.77:4422:

ACK sip:*877121@pbx.bondojito.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-eetzrl6p5054;rport

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=op7imylnay

To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=f9b72edc54

Call-ID: 17c0263cd8d7-yvp7g9nueny5

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

 

[7] 2012/02/22 13:49:29: SIP Rx tls:192.168.30.77:4422:

INVITE sip:*877121@pbx.bondojito.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi

To: <sip:*877121@pbx.bondojito.com;user=phone>

Call-ID: 25c0263c5251-7ubabgx00lt8

CSeq: 1 INVITE

Max-Forwards: 70

Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1

X-Serialnumber: 000413457659

P-Key-Flags: resolution="31x13", keys="4"

User-Agent: snom821/8.4.32

Accept: application/sdp

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

Allow-Events: talk, hold, refer, call-info

Supported: timer, 100rel, replaces, from-change

Session-Expires: 3600;refresher=uas

Min-SE: 90

Proxy-Require: buttons

Content-Type: application/sdp

Content-Length: 526

 

v=0

o=root 1685612402 1685612402 IN IP4 192.168.30.77

s=call

c=IN IP4 192.168.30.77

t=0 0

m=audio 56568 RTP/AVP 9 0 8 2 3 18 4 101

a=crypto:1 AES_CM_128_HMAC_SHA1_32 inline:+gqsKVDkqLc0eaydfsXtEcnTYUf1dCTo+h/mTo34

a=rtpmap:9 G722/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:8 PCMA/8000

a=rtpmap:2 G726-32/8000

a=rtpmap:3 GSM/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:4 G723/8000

a=rtpmap:101 telephone-event/8000

a=fmtp:101 0-16

a=ptime:20

a=rtcp-xr:voip-metrics stat-summary=loss,dup,jitt

a=sendrecv

[7] 2012/02/22 13:49:29: SIP Tx tls:192.168.30.77:4422:

SIP/2.0 100 Trying

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport=4422

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi

To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c

Call-ID: 25c0263c5251-7ubabgx00lt8

CSeq: 1 INVITE

Content-Length: 0

 

[7] 2012/02/22 13:49:29: SIP Tx tls:192.168.30.77:4422:

SIP/2.0 404 Not Found

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport=4422

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi

To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c

Call-ID: 25c0263c5251-7ubabgx00lt8

CSeq: 1 INVITE

Contact: <sip:7120@192.168.30.2:5061;transport=tls>

Supported: 100rel, replaces, norefersub

Allow-Events: refer

Allow: INVITE, ACK, CANCEL, BYE, REFER, PRACK, INFO, UPDATE

Accept: application/sdp

User-Agent: snom-PBX/2011-4.2.1.4025

Content-Length: 0

 

[7] 2012/02/22 13:49:29: SIP Rx tls:192.168.30.77:4422:

ACK sip:*877121@pbx.bondojito.com;user=phone SIP/2.0

Via: SIP/2.0/TLS 192.168.30.77:4422;branch=z9hG4bK-apzk3ou8w7zp;rport

From: "Julio Cesar" <sip:7120@pbx.bondojito.com>;tag=9fmh5ih1pi

To: <sip:*877121@pbx.bondojito.com;user=phone>;tag=fa33140c8c

Call-ID: 25c0263c5251-7ubabgx00lt8

CSeq: 1 ACK

Max-Forwards: 70

Contact: <sip:7120@192.168.30.77:4422;transport=tls;line=0ii3wcy6>;reg-id=1

Proxy-Require: buttons

Content-Length: 0

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Okay, now we know that this is not specific to *77. The version that you are running is somewhat outdated; what could be a problem is that the number of G.729A codecs is exhausted. There was a bug in the older version that could cause this problem.

 

Any chance to upgrade? 4.5 (Beta Corodindis) is available. If you backup your configuration and see if the trunks are working fine after the upgrade (try this after hours to minimize the disturbance), it would be worth giving it a try. If not, you can move back to the previous version.

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