$25 Reward! Everyone is busy/congested at this time [solved]

jjblodg · January 16, 2007, 5:45pm

Ok. Getting somewhere now. Now I have a different message thanks to help from a local friend… Now I get the messages listed below when trying to dial out.

Anyone have any idea if this is a dialplan issue or if it’s possibly an issue with the Sipura I’m using to dial out? I don’t think the hardware is a problem since I can call in just fine through it…

(NOTE: replaced actual numbers with [xxxx])

-- Executing [390[xxxx]@outgoing:1] Dial("SIP/601-b5923f18", "SIP/390[xxxx]@trunk_1|30|r") in new stack
-- Called 390[xxxx]@trunk_1

NOTICE[11094]: chan_sip.c:11719 handle_response_invite: Failed to authenticate on INVITE to ‘“J. J. Blodgett (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as0c3cd771’
– SIP/trunk_1-b5706ac0 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [390[xxxx]@outgoing:2] Busy(“SIP/601-b5923f18”, “”) in new stack
== Spawn extension (outgoing, 390[xxxx], 2) exited non-zero on ‘SIP/601-b5923f18’

<— Transmitting (no NAT) to 192.168.0.74:5060 —>
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-3afd48a5;received=192.168.0.74
From: MY NAME (Sipura 1L2) sip:601@192.168.0.76;tag=3680b5a642dd0a5o1
To: sip:390[xxxx]@192.168.0.76;tag=as0f364fb3
Call-ID: 34b3467a-5b3655e1@192.168.0.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:390[xxxx]@192.168.0.76
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21

[[[extensions.conf]]]
[default]
include = voicemenu-custom-1
include = outgoing
exten => 600,1,Goto(voicemenu-custom-1|s|1)

[outgoing]
include = default
exten => _1NXXNXXXXXX,1,Dial(SIP/${EXTEN}@trunk_1,30,r)
exten => _1NXXNXXXXXX,2,congestion()
exten => _NXXNXXXXXX,1,Dial(SIP/${EXTEN}@trunk_1,30,r)
exten => _NXXNXXXXXX,2,congestion()
exten => _NXXXXXX,1,Dial(SIP/${EXTEN}@trunk_1,30,r)
exten => _NXXXXXX,2,congestion()

[DID_trunk_1]
include = default
include = voicemenu-custom-1
exten => 478287[xxxx],1,Dial(SIP/601&SIP/602,30,Ttr)

[[[users.conf]]]
[trunk_1]
secret = [my eyes only]
provider =
trunkstyle = customvoip
username = 478287[xxxx]@sip.broadvoice.com
authname = 478287[xxxx]
trunkname = Custom - Broadvoice 478287[xxxx] (ATL) DO NOT EDIT VIA WEB
callerid =
hasexten = no
hassip = yes
hasiax = no
registeriax = no
registersip = no
host = sip.broadvoice.com
dialformat = ${EXTEN}
context = DID_trunk_1
group =
insecure = very
fromuser = 478287[xxxx]
fromdomain = sip.broadvoice.com
qualify = yes
pedantic = no
type = friend

[601]
fullname = ME (Office Sipura)
secret = xxxxx
email =
cid_number = 478287[xxxx]
zapchan =
context = outgoing
hasvoicemail = yes
hasdirectory = no
hassip = yes
hasiax = no
hasmanager = no
callwaiting = yes
threewaycalling = yes
mailbox = 601
hasagent = no
group =
host = dynamic
nat = no
type = friend

jjblodg · January 18, 2007, 3:31pm

OK. Fine. I’ll send anyone $25 US via PayPal if you can help figure this out in the next 24 hours. I need to call out and haven’t had any outbound business calls for a week now! Surely somebody can figure out what’s going on???

yusuf · January 18, 2007, 4:35pm

Hi,

So inbound calls from this provider work fine? You have given only the second-half of the SIP debug, post the whole thing.

jjblodg · January 18, 2007, 5:19pm

Yes. Inbound works fine. I can call my main number(s), I get my auto attendant, it reads messages, then dials extensions, I can answer the extensions, and talk to parties on other end with no problem.

I dialed out and captured the entire session log I believe. Hopefully this is what you need (I can also provide full SIP, EXTENSIONS, or USERS conf files if need once logs have been reviewed:

<— SIP read from 192.168.0.74:5060 —>
INVITE sip:390[xxxx]@192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-a5c8d148
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76
Remote-Party-ID: J. J. (Sipura 1L2) sip:601@192.168.0.76;screen=yes;party=calling
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 101 INVITE
Max-Forwards: 70
Contact: J. J. (Sipura 1L2) sip:601@192.168.0.74:5060
Expires: 240
User-Agent: Linksys/SPA2100-3.3.7
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 41570579 41570579 IN IP4 192.168.0.74
s=-
c=IN IP4 192.168.0.74
t=0 0
m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->
— (15 headers 20 lines) —

Sending to 192.168.0.74 : 5060 (NAT)
Using INVITE request as basis request - a4b0378a-1134a190@192.168.0.74
Found peer ‘601’

<— Reliably Transmitting (NAT) to 192.168.0.74:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-a5c8d148;received=192.168.0.74
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76;tag=as6bba992d
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="73c0c2b9"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘a4b0378a-1134a190@192.168.0.74’ in 32000 ms (Method: INVITE)

<— SIP read from 192.168.0.74:5060 —>
ACK sip:390[xxxx]@192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-a5c8d148
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76;tag=as6bba992d
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 101 ACK
Max-Forwards: 70
Contact: J. J. (Sipura 1L2) sip:601@192.168.0.74:5060
User-Agent: Linksys/SPA2100-3.3.7
Content-Length: 0

<------------->

— (10 headers 0 lines) —

<— SIP read from 192.168.0.74:5060 —>
INVITE sip:390[xxxx]@192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-aeb0c5a4
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76
Remote-Party-ID: J. J. (Sipura 1L2) sip:601@192.168.0.76;screen=yes;party=calling
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username=“601”,realm=“asterisk”,nonce=“73c0c2b9”,uri=“sip:390[xxxx]@192.168.0.76”,algorithm=MD5,response="76f1665bb138ce05ae2a70731f7c475b"
Contact: J. J. (Sipura 1L2) sip:601@192.168.0.74:5060
Expires: 240
User-Agent: Linksys/SPA2100-3.3.7
Content-Length: 446
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 41570579 41570579 IN IP4 192.168.0.74
s=-
c=IN IP4 192.168.0.74
t=0 0
m=audio 16428 RTP/AVP 0 2 4 8 18 96 97 98 100 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=fmtp:100 192-193
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

<------------->

— (16 headers 20 lines) —
Sending to 192.168.0.74 : 5060 (NAT)
Using INVITE request as basis request - a4b0378a-1134a190@192.168.0.74
Found peer '601’
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 96
Found RTP audio format 97
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.74:16428
Found description format PCMU for ID 0
Found description format G726-32 for ID 2
Found description format G723 for ID 4
Found description format PCMA for ID 8
Found description format G729a for ID 18
Found description format G726-40 for ID 96
Found description format G726-24 for ID 97
Found description format G726-16 for ID 98
Found description format NSE for ID 100
Found description format telephone-event for ID 101
Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xd0d (g723|ulaw|alaw|g726|g729|ilbc)/video=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 192.168.0.74:16428

Looking for 390[xxxx] in outgoing (domain 192.168.0.76)
list_route: hop: sip:601@192.168.0.74:5060

<— Transmitting (NAT) to 192.168.0.74:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-aeb0c5a4;received=192.168.0.74
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:390[xxxx]@192.168.0.76
Content-Length: 0

<------------>
– Executing [390[xxxx]@outgoing:1] Dial(“SIP/601-081fe540”, “SIP/390[xxxx]@trunk_1|30|r”) in new stack
Audio is at 204.96.xxx.xxx port 14680
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.28.128:5060:
INVITE sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK1c8ca75c;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Thu, 18 Jan 2007 17:08:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 11039 11039 IN IP4 204.96.xxx.xxx
s=session
c=IN IP4 204.96.xxx.xxx
t=0 0
m=audio 14680 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

-- Called 390[xxxx]@trunk_1

<— Transmitting (NAT) to 192.168.0.74:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-aeb0c5a4;received=192.168.0.74
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76;tag=as6e2194cb
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: sip:390[xxxx]@192.168.0.76
Content-Length: 0

<------------>

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 100 Trying
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 102 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK1c8ca75c
Content-Length: 0

<------------->

— (7 headers 0 lines) —

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 401 Unauthorized
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 102 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=ortv
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK1c8ca75c
WWW-Authenticate: DIGEST realm=“BroadWorks”,qop=“auth”,algorithm=MD5,nonce="BroadWorksXex3fu8zlT45i31sBW"
Content-Length: 0

<------------->

— (8 headers 0 lines) —

Transmitting (NAT) to 147.135.28.128:5060:
ACK sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK1c8ca75c;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=ortv
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Audio is at 204.96.xxx.xxx port 14680
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.28.128:5060:
INVITE sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK54e99d53;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“478287[xxxx]@sip.broadvoice.com”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:390[xxxx]@sip.broadvoice.com”, nonce=“BroadWorksXex3fu8zlT45i31sBW”, response=“c213799d9c16de99287baf1059d692d9”, opaque="", qop=auth, cnonce=“75938509”, nc=00000001
Date: Thu, 18 Jan 2007 17:08:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 11039 11040 IN IP4 204.96.xxx.xxx
s=session
c=IN IP4 204.96.xxx.xxx
t=0 0
m=audio 14680 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

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 100 Trying
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 103 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK54e99d53
Content-Length: 0

<------------->

— (7 headers 0 lines) —

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 401 Unauthorized
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 103 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=twxz
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK54e99d53
WWW-Authenticate: DIGEST realm=“BroadWorks”,qop=“auth”,algorithm=MD5,nonce="BroadWorksXex3fu95eT36pzsnBW"
Content-Length: 0

<------------->

— (8 headers 0 lines) —

Transmitting (NAT) to 147.135.28.128:5060:
ACK sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK54e99d53;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=twxz
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 103 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Audio is at 204.96.xxx.xxx port 14680
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.28.128:5060:
INVITE sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK2a738caa;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“478287[xxxx]@sip.broadvoice.com”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:390[xxxx]@sip.broadvoice.com”, nonce=“BroadWorksXex3fu95eT36pzsnBW”, response=“a73dae91abf601080039f8c2f4f66814”, opaque="", qop=auth, cnonce=“70b0fcd1”, nc=00000001
Date: Thu, 18 Jan 2007 17:08:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 11039 11041 IN IP4 204.96.xxx.xxx
s=session
c=IN IP4 204.96.xxx.xxx
t=0 0
m=audio 14680 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

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 100 Trying
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 104 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK2a738caa
Content-Length: 0

<------------->

— (7 headers 0 lines) —

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 401 Unauthorized
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 104 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=acef
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK2a738caa
WWW-Authenticate: DIGEST realm=“BroadWorks”,qop=“auth”,algorithm=MD5,nonce="BroadWorksXex3fu9alT3msiijBW"
Content-Length: 0

<------------->

— (8 headers 0 lines) —

Transmitting (NAT) to 147.135.28.128:5060:
ACK sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK2a738caa;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=acef
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 104 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Audio is at 204.96.xxx.xxx port 14680
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 147.135.28.128:5060:
INVITE sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK510a1025;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 105 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Authorization: Digest username=“478287[xxxx]@sip.broadvoice.com”, realm=“BroadWorks”, algorithm=MD5, uri=“sip:390[xxxx]@sip.broadvoice.com”, nonce=“BroadWorksXex3fu9alT3msiijBW”, response=“b822b65d91a1fa7d52f012231d0f8caa”, opaque="", qop=auth, cnonce=“29cc703f”, nc=00000001
Date: Thu, 18 Jan 2007 17:08:38 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 262

v=0
o=root 11039 11042 IN IP4 204.96.xxx.xxx
s=session
c=IN IP4 204.96.xxx.xxx
t=0 0
m=audio 14680 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

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 100 Trying
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 105 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK510a1025
Content-Length: 0

<------------->

— (7 headers 0 lines) —

<— SIP read from 147.135.28.128:5060 —>
SIP/2.0 401 Unauthorized
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 105 INVITE
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=moqs
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK510a1025
WWW-Authenticate: DIGEST realm=“BroadWorks”,qop=“auth”,algorithm=MD5,nonce="BroadWorksXex3fu9fwTgsjqwiBW"
Content-Length: 0

<------------->

— (8 headers 0 lines) —

Transmitting (NAT) to 147.135.28.128:5060:
ACK sip:390[xxxx]@sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP 204.96.xxx.xxx:5060;branch=z9hG4bK510a1025;rport
From: “J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446
To: sip:390[xxxx]@sip.broadvoice.com;tag=moqs
Contact: sip:478287[xxxx]@204.96.xxx.xxx
Call-ID: 2ee378ef01f7f10f53e151b00fdb79f3@sip.broadvoice.com
CSeq: 105 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

[Jan 18 12:08:39] NOTICE[11094]: chan_sip.c:11719 handle_response_invite: Failed to authenticate on INVITE to ‘“J. J. (Office Sipura)” sip:478287[xxxx]@sip.broadvoice.com;tag=as208e5446’
– SIP/trunk_1-0822b178 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
– Executing [390[xxxx]@outgoing:2] Busy(“SIP/601-081fe540”, “”) in new stack

<— Transmitting (NAT) to 192.168.0.74:5060 —>
SIP/2.0 486 Busy Here
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-aeb0c5a4;received=192.168.0.74
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76;tag=as6e2194cb
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Length: 0
X-Asterisk-HangupCause: Call Rejected
X-Asterisk-HangupCauseCode: 21

<------------>
== Spawn extension (outgoing, 390[xxxx], 2) exited non-zero on ‘SIP/601-081fe540’

<— SIP read from 192.168.0.74:5060 —>
ACK sip:390[xxxx]@192.168.0.76 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.74:5060;branch=z9hG4bK-aeb0c5a4
From: J. J. (Sipura 1L2) sip:601@192.168.0.76;tag=1170cf42656a2d38o1
To: sip:390[xxxx]@192.168.0.76;tag=as6e2194cb
Call-ID: a4b0378a-1134a190@192.168.0.74
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username=“601”,realm=“asterisk”,nonce=“73c0c2b9”,uri=“sip:390[xxxx]@192.168.0.76”,algorithm=MD5,response=“0151873418c3fd5fa95d2e25a4b25b43”

Contact: J. J. (Sipura 1L2) sip:601@192.168.0.74:5060
User-Agent: Linksys/SPA2100-3.3.7
Content-Length: 0

jjblodg · January 19, 2007, 1:47am

Sweet! $25 reward to myself. Bonus Cash!!!

I figured it out. Surpising how simple things can totally screw something up…

In the trunk definition in users.conf, I had:
username = 478287[xxxx]@sip.broadvoice.com

But it SHOULD have been simply:
username = 478287[xxxx]

I think it got entered that way from the GUI somehow and it only affected the outbound calls. I went back and referred to the broadvoice documentation to try to set it up from scratch again (cause it was working at one point last week) and happened to notice that difference.

Go figure! Well, I’ll go spend my $25 on dinner and then find something else to freak out about and post a cry for help as soon as possible.

Happy debugging…