[Asterisk-Users] Broadvoice latest changes and still notworking-An

Wed Mar 9 12:48:01 MST 2005

on the CLI> sip debug

that's it.
Also you can do:
CLI> sip show peers
Get the name of BV that you have and then
CLI> sip debug <peer>
to have only the debug for that peer.

When you are done CLI> exit or CLI> sip no debug

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From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Scott Wolfe
Sent: Wednesday, March 09, 2005 1:45 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Broadvoice latest changes and still notworking-An

Just wondering. How are you getting this debug. I am having problems to and I cant seem to track it down.
----- Original Message ----- 
From: Joe <mailto:asterisk at usfloorsllc.com>  
To: asterisk-users at lists.digium.com 
Sent: Wednesday, March 09, 2005 10:41 AM
Subject: [Asterisk-Users] Broadvoice latest changes and still not working-An

I’ve tried everything with the * box after this weekend.  I have read every document on the problems people are having with them
after this weekend as well, but none of them address my problem.

I checked my settings in my sips which I have below as well,  

I have changed the host file a few times,  but this was new to me and I never had modified it before.  I have and the same results
happened.

I have always used the CHI proxy until this past weekend.

I get a 404 not found when the invite goes out.   

Below is my debug for broadvoice,  which I think tells the whole story,  but for the life of me, I can not figure out where the 404
is coming from.

I have listed my sip file below as well.

Inbound calls work and I am registered.

Before we go into the debug,  I get this message when I reload my configs files.
Mar  9 13:23:48 NOTICE[3161]: chan_sip.c:6779 handle_response: Outbound Registration: Expiry for sip.broadvoice.com is 1948 sec
(Scheduling reregistration in 1933000 ms)

Below is the debug:

    -- Executing Dial("OSS/dsp", "SIP/xxxxxxxxxx at sip.broadvoice.com|30") in new stack
We're at outsideIPaddress port 14842
Answering with preferred capability 0x4 (ulaw)
12 headers, 8 lines
Reliably Transmitting:
INVITE sip:xxxxxxxxxx at proxy.lax.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk" <sip:BBBBBBBBBB at sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx at proxy.lax.broadvoice.com>
Contact: <sip:BBBBBBBBBB at outsideIPaddress>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab at sip.broadvoice.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Wed, 09 Mar 2005 18:15:18 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 164

v=0
o=root 17647 17647 IN IP4 outsideIPaddress
s=session
c=IN IP4 outsideIPaddress
t=0 0
m=audio 14842 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=silenceSupp:off - - - -
 (no NAT) to 147.135.8.128:5060
    -- Called xxxxxxxxxx at sip.broadvoice.com
asterisk1*CLI>

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk" <sip:BBBBBBBBBB at sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx at proxy.lax.broadvoice.com>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab at sip.broadvoice.com
CSeq: 102 INVITE

6 headers, 0 lines
asterisk1*CLI>

Sip read:
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk" <sip:BBBBBBBBBB at sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx at proxy.lax.broadvoice.com>;tag=SD4ou5a99-
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab at sip.broadvoice.com
CSeq: 102 INVITE
Content-Length: 0

7 headers, 0 lines
    -- Got SIP response 404 "Not Found" back from 147.135.8.128
Transmitting:
ACK sip:xxxxxxxxxx at proxy.lax.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk" <sip:BBBBBBBBBB at sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx at proxy.lax.broadvoice.com>;tag=SD4ou5a99-
Contact: <sip:BBBBBBBBBB at outsideIPaddress>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab at sip.broadvoice.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

 (no NAT) to 147.135.8.128:5060
    -- SIP/sip.broadvoice.com-2a2c is circuit-busy
  == Everyone is busy/congested at this time
    -- Executing Busy("OSS/dsp", "") in new stack
Destroying call '0674f2a33bfee57a7c9232e10282b5ab at sip.broadvoice.com'
asterisk1*CLI> hangup
  == Spawn extension (default, 509, 102) exited non-zero on 'OSS/dsp'
 << Hangup on console >>

[sip.broadvoice.com]
type=peer
host=proxy.lax.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser= BBBBBBBBBB
username= BBBBBBBBBB
;authuser= BBBBBBBBBB
secret= secret
context=sip
nat=no
insecure=very
dtmfmode=inband

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