Fast Busy Signal

Asterisk Asterisk Support
1

when ever I try to dial out via my sip trunk, I get a fast busy signal. from the logs I see this error;

[Sep 18 14:18:31] NOTICE[29037]: chan_sip.c:18160 handle_request_invite: Call from ‘1002’ to extension ‘912128329127’ rejected because extension not found.
Scheduling destruction of SIP dialog ‘ffd19494-1371ff39-330e6caa@000.000.000.000’ in 32000 ms (Method: INVITE)
senna*CLI>
<— SIP read from UDP://000.000.000.000:5060 —>
ACK sip:912128329127@000.000.000.000:5060 SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000;branch=z9hG4bK878dd7a5D421CDE6
From: “temp” sip:1002@000.000.000.;tag=145E3037-146B05F0
To: sip:912128329127@000.000.000.000;user=phone;tag=as19956de2
CSeq: 2 ACK
Call-ID: ffd19494-1371ff39-330e6caa@000.000.000.000
Contact: sip:1002@000.000.000.000
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“412c1927”, uri=“sip:912128329127@000.000.000.000:5060;user=phone”, response=“25f41afe48875ed7a3e0cbedcdc8a84b”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

2

The SIP trace is irrelevant. In fact it doesn’t indicate any problem.

The context selected by sip.conf for calls from 1002 does not contain a dialplan rule that matches the pattern 912128329127. The CLI output just before the bit you included would probably have told you which context it thought 1002 was in.

3

david55 , below is the complete out put form the console. thanks for your help;

<— SIP read from UDP://000.000.000.000:5060 —>
INVITE sip:912128329127@000.000.000.000:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000;branch=z9hG4bKe5feb3d0605DD205
From: “Yan” sip:1002@000.000.000.000;tag=F857CC31-246A96C2
To: sip:912128329127@000.000.000.000;user=phone
CSeq: 1 INVITE
Call-ID: b31852c6-abe59263-f3f811ec@000.000.000.000
Contact: sip:1002@000.000.000.000
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 978406510 978406510 IN IP4 000.000.000.000
s=Polycom IP Phone
c=IN IP4 000.000.000.000
t=0 0
m=audio 2222 RTP/AVP 18 0 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<------------->
— (14 headers 11 lines) —
Sending to 000.000.000.000 : 5060 (no NAT)
Using INVITE request as basis request - b31852c6-abe59263-f3f811ec@000.000.000.000
Found peer ‘1002’ for ‘1002’ from 000.000.000.000:5060

<— Reliably Transmitting (no NAT) to 000.000.000.000:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 000.000.000.000;branch=z9hG4bKe5feb3d0605DD205;received=000.000.000.000
From: “Yan” sip:1002@000.000.000.000;tag=F857CC31-246A96C2
To: sip:912128329127@000.000.000.000;user=phone;tag=as66b02065
Call-ID: b31852c6-abe59263-f3f811ec@000.000.000.000
CSeq: 1 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="3b94fd72"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘b31852c6-abe59263-f3f811ec@000.000.000.000’ in 32000 ms (Method: INVITE)
senna*CLI>
<— SIP read from UDP://000.000.000.000:5060 —>
ACK sip:912128329127@000.000.000.000:5060 SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000;branch=z9hG4bKe5feb3d0605DD205
From: “Yan” sip:1002@000.000.000.000;tag=F857CC31-246A96C2
To: sip:912128329127@000.000.000.000;user=phone;tag=as66b02065
CSeq: 1 ACK
Call-ID: b31852c6-abe59263-f3f811ec@000.000.000.000
Contact: sip:1002@000.000.000.000
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
Max-Forwards: 70
Content-Length: 0

<------------->
— (11 headers 0 lines) —
senna*CLI>
<— SIP read from UDP://000.000.000.000:5060 —>
INVITE sip:912128329127@000.000.000.000:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000;branch=z9hG4bK7c68deef6ED9D2C8
From: “Yan” sip:1002@000.000.000.000;tag=F857CC31-246A96C2
To: sip:912128329127@000.000.000.000;user=phone
CSeq: 2 INVITE
Call-ID: b31852c6-abe59263-f3f811ec@000.000.000.000
Contact: sip:1002@000.000.000.000
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
Supported: 100rel,replaces
Allow-Events: talk,hold,conference
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“3b94fd72”, uri=“sip:912128329127@000.000.000.000:5060;user=phone”, response=“1d74e55d6e1d971b8883da1f3f6a737e”, algorithm=MD5
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 249

v=0
o=- 978406510 978406510 IN IP4 000.000.000.000
s=Polycom IP Phone
c=IN IP4 000.000.000.000
t=0 0
m=audio 2222 RTP/AVP 18 0 8 101
a=sendrecv
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000

<------------->
— (15 headers 11 lines) —
Sending to 000.000.000.000 : 5060 (no NAT)
Using INVITE request as basis request - b31852c6-abe59263-f3f811ec@000.000.000.000
Found peer ‘1002’ for ‘1002’ from 000.000.000.000:5060
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 000.000.000.000:2222
Found audio description format G729 for ID 18
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 000.000.000.000:2222
Looking for 912128329127 in phones (domain 000.000.000.000)

<— Reliably Transmitting (no NAT) to 000.000.000.000:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 000.000.000.000;branch=z9hG4bK7c68deef6ED9D2C8;received=000.000.000.000
From: “Yan” sip:1002@000.000.000.000;tag=F857CC31-246A96C2
To: sip:912128329127@000.000.000.000;user=phone;tag=as66b02065
Call-ID: b31852c6-abe59263-f3f811ec@000.000.000.000
CSeq: 2 INVITE
Server: Asterisk PBX 1.6.1.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
Content-Length: 0

<------------>
[Sep 18 15:47:38] NOTICE[29718]: chan_sip.c:18160 handle_request_invite: Call from ‘1002’ to extension ‘912128329127’ rejected because extension not found.
Scheduling destruction of SIP dialog ‘b31852c6-abe59263-f3f811ec@000.000.000.000’ in 32000 ms (Method: INVITE)
senna*CLI>
<— SIP read from UDP://000.000.000.000:5060 —>
ACK sip:912128329127@000.000.000.000:5060 SIP/2.0
Via: SIP/2.0/UDP 000.000.000.000;branch=z9hG4bK7c68deef6ED9D2C8
From: “Yan” sip:1002@000.000.000.000;tag=F857CC31-246A96C2
To: sip:912128329127@000.000.000.000;user=phone;tag=as66b02065
CSeq: 2 ACK
Call-ID: b31852c6-abe59263-f3f811ec@000.000.000.000
Contact: sip:1002@000.000.000.000
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER
User-Agent: PolycomSoundPointIP-SPIP_320-UA/2.1.2.0049
Authorization: Digest username=“1002”, realm=“asterisk”, nonce=“3b94fd72”, uri=“sip:912128329127@000.000.000.000:5060;user=phone”, response=“1d74e55d6e1d971b8883da1f3f6a737e”, algorithm=MD5
Max-Forwards: 70
Content-Length: 0

4

[code]Looking for 912128329127 in phones (domain 000.000.000.000)

<— Reliably Transmitting (no NAT) to 000.000.000.000:5060 —>
SIP/2.0 404 Not Found [/code]

There is no exten entry in the [phones] section of your dialplan that matches 912128329127, or you didn’t intend your phone to be in the phones context.