Incoming calls busy signal

etechllc · August 30, 2008, 6:57pm

Hello everyone,

I had incoming calls working fine with my BroadVoice account and Asterisk 1.6.0-beta9. After making several test calls, I now get a busy signal when I try to call in. I’ve made no changes to my dial plan.

sip show registry
Host Username Refresh State Reg.Time
sip.broadvoice.com:5060 1234567890 24 Registered Sat, 30 Aug 2008 13:54:36
1 SIP registrations.

The only thing I see when I make an incoming call is:

*CLI> == Using SIP RTP CoS mark 5

This is with three levels of verbosity. Is there any information I can provide that would help troubleshoot this issue, or does anyone have an idea from what I’ve provided? Thanks in advance for your help, it’s much appreciated!

Best Regards,
Martin Schultz

ianplain · August 30, 2008, 7:28pm

Try upping verbose to at least 12 and enable sip debug for the peer.

Ian

etechllc · August 31, 2008, 7:39pm

Hello ianplain, thanks for your help so far. I’ve enabled verbosity up to 12 and SIP debug is now on. Here is the sequence of notifications I get when I attempt to call from my cell phone (2165551212) to my Asterisk (2165881234). I don’t know if this helps, but I am able to call out from my Asterisk to my cell phone.

Any help you guys can offer would be great, I’m very new to this and just trying to follow the instructions in the setup guide…

-Martin

*CLI> sip set debug on
SIP Debugging enabled
*CLI> Really destroying SIP dialog ‘ad02e4-ad@147.135.12.128’ Method: ACK

<— SIP read from UDP://147.135.12.128:5060 —>
INVITE sip:2165881234@74.XX.XX.34:5060 SIP/2.0
Call-ID: 4a01d7-4a@147.135.12.128
CSeq: 1 INVITE
From: "Cleveland OH"sip:2165551212@147.135.12.128;user=phone;tag=mpqs
To: "Martin Schultz"sip:s@74.XX.XX.34
Via: SIP/2.0/UDP 147.135.12.128:5060
Contact: sip:2165551212@147.135.12.128:5060
Supported: 100rel
Content-Length: 272
Content-Type: application/sdp

v=0
o=2475101431 10 10 IN IP4 147.135.12.247
s=-
c=IN IP4 147.135.12.248
t=0 0
m=audio 10410 RTP/AVP 0 8 18 96 97
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:96 iLBC/8000
a=fmtp:96 mode=30
a=rtpmap:97 t38/8000

<------------->
— (10 headers 13 lines) —
== Using SIP RTP CoS mark 5
Sending to 147.135.12.128 : 5060 (no NAT)
Using INVITE request as basis request - 4a01d7-4a@147.135.12.128
No user ‘2165551212’ in SIP users list
Found peer ‘bv-0143’ for ‘2165551212’ from 147.135.12.128:5060

<— Reliably Transmitting (no NAT) to 147.135.12.128:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 147.135.12.128:5060;received=147.135.12.128
From: "Cleveland OH"sip:2165551212@147.135.12.128;user=phone;tag=mpqs
To: "Martin Schultz"sip:s@74.XX.XX.34;tag=as672635f0
Call-ID: 4a01d7-4a@147.135.12.128
CSeq: 1 INVITE
User-Agent: Asterisk PBX 1.6.0-beta9
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="7b975dc8"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘4a01d7-4a@147.135.12.128’ in 32000 ms (Method: INVITE)

<— SIP read from UDP://147.135.12.128:5060 —>
ACK sip:2165881234@74.XX.XX.34:5060 SIP/2.0
Call-ID: 4a01d7-4a@147.135.12.128
CSeq: 1 ACK
From: "Cleveland OH"sip:2165551212@147.135.12.128;user=phone;tag=mpqs
To: "Martin Schultz"sip:s@74.XX.XX.34;tag=as672635f0
Via: SIP/2.0/UDP 147.135.12.128:5060
Content-Length: 0

<------------->
— (7 headers 0 lines) —
[Aug 31 14:32:51] NOTICE[21780]: chan_sip.c:9088 sip_reregister: – Re-registration for 2165881234@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.12.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 74.XX.XX.34:5060;branch=z9hG4bK42faca2f;rport
Max-Forwards: 70
From: sip:2165881234@sip.broadvoice.com;tag=as2556ddac
To: sip:2165881234@sip.broadvoice.com
Call-ID: 4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34
CSeq: 118 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Authorization: Digest username=“2165881234”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfkk20h6eT12dythBW”, response=“d1ee64b30f6e8685226eb904f6b30df2”, qop=auth, cnonce=“2d9a8d1b”, nc=00000010
Expires: 120
Contact: sip:s@74.XX.XX.34
Event: registration
Content-Length: 0

<— SIP read from UDP://147.135.12.128:5060 —>
SIP/2.0 200 OK
Call-ID: 4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34
CSeq: 118 REGISTER
From: sip:2165881234@sip.broadvoice.com;tag=as2556ddac
To: sip:2165881234@sip.broadvoice.com
Via: SIP/2.0/UDP 74.XX.XX.34:5060;branch=z9hG4bK42faca2f
Contact: sip:s@74.XX.XX.34
Expires: 30
Event: registration
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog '4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34’ in 32000 ms (Method: REGISTER)
[Aug 31 14:32:51] NOTICE[21780]: chan_sip.c:15029 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog '4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34’ Method: REGISTER
[Aug 31 14:33:15] NOTICE[21780]: chan_sip.c:9088 sip_reregister: – Re-registration for 2165881234@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.12.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 74.XX.XX.34:5060;branch=z9hG4bK6a6923e5;rport
Max-Forwards: 70
From: sip:2165881234@sip.broadvoice.com;tag=as73daf674
To: sip:2165881234@sip.broadvoice.com
Call-ID: 4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34
CSeq: 119 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Authorization: Digest username=“2165881234”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfkk20h6eT12dythBW”, response=“17c0974c195f37d8bcf7ac79fb09c343”, qop=auth, cnonce=“4e19d92c”, nc=00000011
Expires: 120
Contact: sip:s@74.XX.XX.34
Event: registration
Content-Length: 0

<— SIP read from UDP://147.135.12.128:5060 —>
SIP/2.0 200 OK
Call-ID: 4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34
CSeq: 119 REGISTER
From: sip:2165881234@sip.broadvoice.com;tag=as73daf674
To: sip:2165881234@sip.broadvoice.com
Via: SIP/2.0/UDP 74.XX.XX.34:5060;branch=z9hG4bK6a6923e5
Contact: sip:s@74.XX.XX.34
Expires: 30
Event: registration
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog '4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34’ in 32000 ms (Method: REGISTER)
[Aug 31 14:33:15] NOTICE[21780]: chan_sip.c:15029 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog ‘4a01d7-4a@147.135.12.128’ Method: ACK
Really destroying SIP dialog '4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34’ Method: REGISTER
[Aug 31 14:33:39] NOTICE[21780]: chan_sip.c:9088 sip_reregister: – Re-registration for 2165881234@sip.broadvoice.com
REGISTER 13 headers, 0 lines
Reliably Transmitting (no NAT) to 147.135.12.128:5060:
REGISTER sip:sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 74.XX.XX.34:5060;branch=z9hG4bK1d0acf1f;rport
Max-Forwards: 70
From: sip:2165881234@sip.broadvoice.com;tag=as56aefae3
To: sip:2165881234@sip.broadvoice.com
Call-ID: 4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34
CSeq: 120 REGISTER
User-Agent: Asterisk PBX 1.6.0-beta9
Authorization: Digest username=“2165881234”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:sip.broadvoice.com”, nonce=“BroadWorksXfkk20h6eT12dythBW”, response=“f7cfe52100d30b6bac95fd25049eb000”, qop=auth, cnonce=“4a09de54”, nc=00000012
Expires: 120
Contact: sip:s@74.XX.XX.34
Event: registration
Content-Length: 0

<— SIP read from UDP://147.135.12.128:5060 —>
SIP/2.0 200 OK
Call-ID: 4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34
CSeq: 120 REGISTER
From: sip:2165881234@sip.broadvoice.com;tag=as56aefae3
To: sip:2165881234@sip.broadvoice.com
Via: SIP/2.0/UDP 74.XX.XX.34:5060;branch=z9hG4bK1d0acf1f
Contact: sip:s@74.XX.XX.34
Expires: 30
Event: registration
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog '4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34’ in 32000 ms (Method: REGISTER)
[Aug 31 14:33:39] NOTICE[21780]: chan_sip.c:15029 handle_response_register: Outbound Registration: Expiry for sip.broadvoice.com is 30 sec (Scheduling reregistration in 24 s)
Really destroying SIP dialog '4aaf22141bbfc7de2baeaf105ce2fac6@74.XX.XX.34’ Method: REGISTER

etechllc · August 31, 2008, 8:54pm

Here is my sip.conf:

[general]
register => 2165881234:BVPASS@sip.broadvoice.com
pedantic=no

[bv-1234]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=2165881234
secret=BVPASS
username=2165881234
insecure=very
context=incoming
authname=2165881234
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=yes

[1000]
type=friend
context=phones
host=dynamic
secret=EXTPASS

[6000]
type=friend
context=phones
host=dynamic
secret=EXTPASS

[6001]
type=friend
context=phones
host=dynamic
secret=EXTPASS

etechllc · August 31, 2008, 8:55pm

And here is my extensions.conf:

[incoming]
exten => _216.,1,Answer()
exten => _216.,n,Background(en/enter-ext-of-person)
exten => _216.,n,WaitExten()
exten => i,1,Playback(en/pbx-invalid)
exten => i,n,Goto(incoming,_216.,1)
exten => t,1,Playback(en/vm-goodbye)
exten => t,n,Hangup()
include => phones

[phones]
exten => 123,1,Answer()
exten => 123,n,Background(en/enter-ext-of-person)
exten => 123,n,WaitExten()
exten => 1,1,Playback(digits/1)
exten => 1,n,Goto(phones,123,1)
exten => 2,1,Playback(digits/2)
exten => 2,n,Goto(phones,123,1)
exten => 3,1,Playback(digits/3)
exten => 3,n,Goto(phones,123,1)
exten => 1000,1,Dial(SIP/1000)
exten => 6000,1,Dial(SIP/6000)
exten => 6001,1,Dial(SIP/6001)
exten => i,1,Playback(en/pbx-invalid)
exten => i,n,Goto(phones,123,1)
exten => t,1,Playback(en/vm-goodbye)
exten => t,n,Hangup()
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@BV-1234,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()

etechllc · September 1, 2008, 4:23am

Anyone? I’m still very stuck…