[Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server

Zanzamar Majere phoneman at wbtllc.com
Wed Mar 9 01:23:55 MST 2005 

I have made all the changes to sip.conf for my broadvoice peer
friend(and I have tried it as peer) and I am still seeing this response
(on call out).  Any suggestions?  I don't think it is a problem with the
phones themselves authenticating, as Asterisk takes care of all the
authentication from my understanding.  

Free world does work for calling out however.  So I know at least that
works.



-- Got SIP response 400 "Bad request" back from 147.135.0.128
Mar  9 01:11:09 NOTICE[12284]: chan_sip.c:6798 handle_response: Failed
to authenticate on INVITE to '"PPPPPPPPPP"
<sip:PPPPPPPPPP at sip.broadvoice.com>;tag=as5b80cade'

On Wed, 2005-03-09 at 00:48, Chris Nibeck wrote:
> First off...  please cancel previous amplification request.  I have  
> implemented your ideas with the same errored result.
> 
> I am not sure that we are not making it thru authentication.  From my  
> digging and comparing packet dumps comparing the soft phone to asterisk  
> they have identical transactions through  the ACK reply (the last one  
> on the debug below).  The softphone seems to be authenticated after the  
> ACK.  I am a newbie to debugging this stuff. I just want to get it  
> working.
> 
> Thanks everyone in advance for your help.  I am certainly very very  
> happy to try anything.
> 
> Based on Luki's suggestions I...
> 
> Changed sip.conf...
> 
> [broadvoice1]
> type=peer
> ;user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=8475100139
> secret=zjh018g8f8
> username=8475100139
> insecure=very
> context=default
> authname=8475100139
> dtmfmode=inband
> dtmf=inband
> ;Disable canreinvite if you are behind a NAT
> canreinvite=no
> nat=no
> 
> Changed extensions.conf...
> 
> exten => _8X.,1, dial(SIP/${EXTEN:1}@broadvoice1,30) ; Dial Broadvoice  
> for 30 seconds
> exten => _8X.,2, congestion() ; No answer, nothing
> exten => _8X., 102, busy() ;
> 
> End result...
> 
> Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
> to authenticate on INVITE to '"6050"  
> <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3'
> 
> 
> SIP debug...
> 
>      -- Executing Dial("SIP/6050-132b",  
> "SIP/18475098263 at broadvoice1|30") in new stack
> We're at xxx.xxx.xxx.xxx port 18212
> Answering with capability 2
> Answering with capability 4
> Answering with capability 8
> 12 headers, 10 lines
> Reliably Transmitting:
> INVITE sip:18475098263 at sip.broadvoice.com SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
> To: <sip:18475098263 at sip.broadvoice.com>
> Contact: <sip:8475100139 at xxx.xxx.xxx.xxx>
> Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Date: Wed, 09 Mar 2005 07:30:41 GMT
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Content-Type: application/sdp
> Content-Length: 205
> 
> v=0
> o=root 3998 3998 IN IP4 xxx.xxx.xxx.xxx
> s=session
> c=IN IP4 xxx.xxx.xxx.xxx
> t=0 0
> m=audio 18212 RTP/AVP 3 0 8
> a=rtpmap:3 GSM/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=silenceSupp:off - - - -
>   (no NAT) to 147.135.8.128:5060
>      -- Called 18475098263 at broadvoice1
> com*CLI>
> 
> Sip read:
> INVITE sip:818475098263 at com.imediainc.net SIP/2.0
> Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
> From: 6050 <sip:6050 at com.imediainc.net>;tag=7e2776985d5a0891o0
> To: <sip:818475098263 at com.imediainc.net>
> Call-ID: 26c50864-232ec135 at 64.4.192.110
> CSeq: 102 INVITE
> Max-Forwards: 70
> Proxy-Authorization: Digest  
> username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: 
> 818475098263 at com.imediainc.net",algorithm=MD5,response="420e39b35648a10c 
> 129dd4fb5f97ec47"
> Contact: 6050 <sip:6050 at 64.4.192.110:5060>
> Expires: 240
> User-Agent: Sipura/SPA3000-2.0.10(GWf)
> Content-Length: 241
> Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
> Supported: x-sipura
> Content-Type: application/sdp
> 
> v=0
> o=- 1138990026 1138990026 IN IP4 64.4.192.110
> s=-
> c=IN IP4 64.4.192.110
> t=0 0
> m=audio 16388 RTP/AVP 0 100 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:100 NSE/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-15
> a=ptime:30
> a=sendrecv
> 
> 15 headers, 12 lines
> Ignoring this request
> Transmitting (no NAT):
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
> From: 6050 <sip:6050 at com.imediainc.net>;tag=7e2776985d5a0891o0
> To: <sip:818475098263 at com.imediainc.net>;tag=as2f065f18
> Call-ID: 26c50864-232ec135 at 64.4.192.110
> CSeq: 102 INVITE
> User-Agent: Asterisk PBX
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
> Contact: <sip:818475098263 at xxx.xxx.xxx.xxx>
> Content-Length: 0
> 
> 
>   to 64.4.192.110:5060
> com*CLI>
> 
> Sip read:
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
> To: <sip:18475098263 at sip.broadvoice.com>
> Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
> CSeq: 102 INVITE
> 
> 
> 6 headers, 0 lines
> com*CLI>
> 
> Sip read:
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
> To: <sip:18475098263 at sip.broadvoice.com>;tag=SD38rq699-
> Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
> CSeq: 102 INVITE
> WWW-Authenticate: DIGEST  
> realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
> Content-Length: 0
> 
> 
> 8 headers, 0 lines
> Transmitting:
> ACK sip:18475098263 at sip.broadvoice.com SIP/2.0
> Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
> From: "6050" <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3
> To: <sip:18475098263 at sip.broadvoice.com>;tag=SD38rq699-
> Contact: <sip:8475100139 at xxx.xxx.xxx.xxx>
> Call-ID: 06508e9743b24399385d5543410b1a06 at xxx.xxx.xxx.xxx
> CSeq: 102 ACK
> User-Agent: Asterisk PBX
> Content-Length: 0
> 
>   (no NAT) to 147.135.8.128:5060
> Mar  9 01:30:42 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed  
> to authenticate on INVITE to '"6050"  
> <sip:8475100139 at sip.broadvoice.com>;tag=as545ccba3'
> 
> 
> 
> On Mar 9, 2005, at 12:08 AM, Luki wrote:
> 
> > Chris,
> >
> > first of all, if your server has been up for 200 days, I suggest you
> > update the kernel -- you don't say if it's Linux, but chances are that
> > yes... and there have been some security bugs patched recently.
> >
> > That aside. I'm not sure, but it's possible that since you are using a
> > valid host name ('sip.broadvoice.com') in your dial statement, perhaps
> > * tried to talk to it directly and does not consider the section in
> > sip.conf. Just a guess. You will notice from the the sip debug output
> > that * does not even try to authenticate, as if it didn't know about
> > the user/secret.
> >
> > I use the BV number as the section name, so the dial statement
> > essentially looks like: Dial(${EXTEN}@${BV_LINE})
> >
> > Try changing yours to say "broadvoice" and then the corresponding
> > section in sip.conf. I'm using the DCA server, and didn't have an
> > issue at all when they introduced INVITE authentication on the
> > weekend. This is how my section looks like:
> >
> > [360350XXXX]
> > type=peer
> > dtmfmode=inband
> > username=360350XXXX
> > fromuser=360350XXXX
> > secret=XXXXXXXXXX
> > host=sip.broadvoice.com
> > fromdomain=sip.broadvoice.com
> > canreinvite=no
> > nat=no
> > insecure=very
> > context=incoming
> > outgoinglimit=2
> >
> > In /etc/hosts I have:
> > 147.135.0.128           sip.broadvoice.com
> >
> > It's the proxy.dca.broadvoice.com server. Hope this helps...
> >
> > --Luki
> > _______________________________________________
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