[Asterisk-Users] weird outbound problem through broadvoice (new)

[Paul P. Pongco]

Hello,

Have a weird problem when using asterisk (1.0.6). There are certain
numbers I cannot dial when using asterisk with my broadvoice account.
No problems with inbound. With outbound calls, I can call some numbers
(for example broadvoice customer support number) and unsuccessfully with
some. However, when I configure my account directly on x-lite, I dont
see these outbound problems.
Here is a snapshot of my sip.conf

register => UUUUUUUUUU at sip.broadvoice.com:PPPPPPPPPP:UUUUUUUUUU at sip.broadvoice.com
 
 
[sip.broadvoice.com]
type=peer
host=sip.broadvoice.com
fromuser=UUUUUUUUUU
fromdomain=sip.broadvoice.com
secret=PPPPPPPPPP
username=UUUUUUUUUU
port=5060
dtmfmode=inband
dtmf=inband
insecure=very
context=incoming
authname=UUUUUUUUUU
canreinvite=no
qualify=no
nat=no

extensions.conf
[outgoing]
exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()

A portion of sip debug during successful calls (calling broadvoice
support)

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a
From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as65b65920
To: <sip:19784187300 at sip.broadvoice.com>
Call-ID: 2007fca97e36e72b54818caa377e6dcc at sip.broadvoice.com
CSeq: 103 INVITE
  
6 headers, 0 lines
CLI>
  
Sip read:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK27bcee7a
From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as65b65920
To:
<sip:19784187300 at sip.broadvoice.com>;tag=SD58a8499-104694000-1110784950009
Call-ID: 2007fca97e36e72b54818caa377e6dcc at sip.broadvoice.com
CSeq: 103 INVITE
Allow: ACK,BYE,CANCEL,INFO,INVITE,OPTIONS,PRACK,REFER,UPDATE,NOTIFY
Supported: 100rel,timer
Contact:
<sip:19784187300 at 147.135.8.128:5060;bvoice=ACME-ntqjclfhfev2b;ep=147.135.8.129;transport=udp>
Remote-Party-ID: "Auto Attendant
PrimaryAttendant"<sip:9784187395 at 147.135.8.129;user=phone;bvoice=ACME-06t5tpji5ub7e>;screen=yes;party=called;privacy=off;id-type=subscriber
Content-Length: 0

A portion of sip debug during unsuccessful calls, where TTTTTTTTT is the
target phone number

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18
From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as6f6dba69
To: <sip:1TTTTTTTTTT at sip.broadvoice.com>
Call-ID: 095981b26d97329e4155ccd529617e5c at sip.broadvoice.com
CSeq: 103 INVITE
  
  
6 headers, 0 lines
Reliably Transmitting:
CANCEL sip:1TTTTTTTTTT at sip.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP x.x.x.x:5060;branch=z9hG4bK0c7f8b18
From: "1001" <sip:UUUUUUUUUU at sip.broadvoice.com>;tag=as6f6dba69
To: <sip:1TTTTTTTTTT at sip.broadvoice.com>
Contact: <sip:UUUUUUUUUU at x.x.x.x>
Call-ID: 095981b26d97329e4155ccd529617e5c at sip.broadvoice.com
CSeq: 103 CANCEL
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="UUUUUUUUUU", realm="BroadWorks",
algorithm=MD5,
uri="sip:1TTTTTTTTTT at sip.broadvoice.com", nonce="1110785211206",
response="f68a31735aec843b9ef68b7909fcf178", opaque=""
Content-Length: 0
  
 (no NAT) to 147.135.8.128:5060
Scheduling destruction of call
'095981b26d97329e4155ccd529617e5c at sip.broadvoice.com' in 15000 ms
Transmitting (no NAT):
SIP/2.0 503 Service Unavailable
Via: SIP/2.0/UDP x.x.x.x;branch=z9hG4bK01853115f3033a3c
From: <sip:1001 at x.x.x.x>;tag=9d9e03fd7b4508e9
To: <sip:1TTTTTTTTTT at x.x.x.x>;tag=as79fd7936
Call-ID: 3512f0bb5f5ebf20 at x.x.x.x
CSeq: 7327 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:1TTTTTTTTT at x.x.x.x>
Content-Length: 0
   
to x.x.x.x:5060

Asterisk box not behind firewall. No iptables filters either. It seems
that asterisk is sending CANCEL due to call timeout after the 2nd 100
Trying during INVITE message flow. I am not sure what is causing the
timeout. Anyone experienced this before? Tried using ethereal to debug
the problem deeply, but I can only see the same flow as the sip debug.
Hoping for your assistance. Thanks.