[Asterisk-Users] Broadvoice latest changes and still not working-
An Additional Server
Chris Nibeck
nibeck at interaccess.com
Wed Mar 9 09:12:54 MST 2005
Thanks MF,
Yes that was me that sent my PW :-) It is changed now.
Same error...
Mar 9 10:12:46 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed
to authenticate on INVITE to '"Chris Nibeck"
<sip:8475100139 at sip.broadvoice.com>;tag=as0cefa74c'
Sip.conf...
[*8475100139*]
type=peer
;user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=8475100139
secret=xxxxx
username=8475100139
insecure=very
context=default
authname=8475100139
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no
nat=no
extensions.conf...
exten => _8X.,1, dial(SIP/${EXTEN:1}@*8475100139*,30) ; Dial Broadvoice
for 30 seconds
exten => _8X.,2, congestion() ; No answer, nothing
exten => _8X., 102, busy() ;
On Mar 9, 2005, at 7:56 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="420e39b35648a
>>> 10c 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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