[Asterisk-Users] Broadvoice latest changes and still not working-
An Additional Server
MF Hulber
asterisk-admin at hulber.com
Wed Mar 9 06:56:05 MST 2005
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
>>>_______________________________________________
>>>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
>>>
>>>
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>>
>
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