Joe
2005-Mar-09  11:41 UTC
[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@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@proxy.lax.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
<sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>
Contact: <sip:BBBBBBBBBB@outsideIPaddress>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@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@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99-
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@proxy.lax.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
<sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99-
Contact: <sip:BBBBBBBBBB@outsideIPaddress>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@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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Scott Wolfe
2005-Mar-09  11:44 UTC
[Asterisk-Users] Broadvoice latest changes and still not working-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 
  To: asterisk-users@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@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@proxy.lax.broadvoice.com SIP/2.0
  Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk"
<sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9
  To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>
  Contact: <sip:BBBBBBBBBB@outsideIPaddress>
  Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@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@sip.broadvoice.com>;tag=as6ed673e9
  To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>
  Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@sip.broadvoice.com>;tag=as6ed673e9
  To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99-
  Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@proxy.lax.broadvoice.com SIP/2.0
  Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
  From: "asterisk"
<sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9
  To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99-
  Contact: <sip:BBBBBBBBBB@outsideIPaddress>
  Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@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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Marios Andreou
2005-Mar-09  12:45 UTC
[Asterisk-Users] Broadvoice latest changes and still not working-An
The problem that you have it was the one that I stabled across the very first
time tried to setup BV.
The 404 not found that you are getting is because there is no such phone number
xxxxx@proxy.lax.broadvoice.com
But there is a xxxxx@sip.broadvoice.com.
 
This is like saying xxxxx@yourownlocaldomain.tld (you are going to get a 404)
 
The chi worked because it was a test server (beta/debug) that I read somewhere
in this list.
 
So the fix for you will be to change the 
 
host=proxy.lax.broadvoice.com
to 
host=sip.broadvoice.com
 
 
Now if you are getting better responses from lax then change your host file to
147.135.8.128  sip.broadvoice.com
 
This is because sip.broadvoice.com resolves to proxy.dca.broadvoice.com.
 
  _____  
From: asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Joe
Sent: Wednesday, March 09, 2005 1:41 PM
To: asterisk-users@lists.digium.com
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@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@proxy.lax.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
<sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>
Contact: <sip:BBBBBBBBBB@outsideIPaddress>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@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@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99-
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@proxy.lax.broadvoice.com SIP/2.0
Via: SIP/2.0/UDP outsideIPaddress:5060;branch=z9hG4bK1572d8bc
From: "asterisk"
<sip:BBBBBBBBBB@sip.broadvoice.com>;tag=as6ed673e9
To: <sip:xxxxxxxxxx@proxy.lax.broadvoice.com>;tag=SD4ou5a99-
Contact: <sip:BBBBBBBBBB@outsideIPaddress>
Call-ID: 0674f2a33bfee57a7c9232e10282b5ab@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@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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