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

Zanzamar Majere Phoneman at wbtllc.com
Wed Mar 9 07:53:33 MST 2005 

Thank you for the response.   I still have the errors mentioned below, sip 
response and Failed to authenticate on INVITE

[PPPPPPPPPP]
type=peer
username=PPPPPPPPPP
fromuser=PPPPPPPPPP
authuser=PPPPPPPPPP
fromdomain=sip.broadvoice.com
secret=XXXXXXXXXX
host=sip.broadvoice.com
dtmfmode=inband
insecure=very
context=sip
qualify=yes
disallow=all
allow=ulaw
allow=gsm
;Disable canreinvite if you are behind a NAT
;canreinvite=no
nat=no

Does anyone else have any other suggestions?


On Wednesday 09 March 2005 06:56 am, MF Hulber wrote:
> Try changing the extension from Broadvoice1 to the actual phone number
> (and don't send your secret in a public email or maybe that's Chris'):
>
> [*8475100139*]
> type=peer
> ;user=phone
> host=sip.broadvoice.com
> fromdomain=sip.broadvoice.com
> fromuser=8475100139
> secret=XXXXXXXXXXX
> username=8475100139
>
> Zanzamar Majere wrote:
> >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=DELETED
> >>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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> >>
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