[Asterisk-Users] Broadvoice latest changes and still not
working-An
Scott Wolfe
scottwolfe at orbus.net
Wed Mar 9 11:44:32 MST 2005
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 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
------------------------------------------------------------------------------
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users at lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050309/d04f78e0/attachment.htm
More information about the asterisk-users
mailing list