[Asterisk-Users] Broadvoice latest changes and stillnot working- An Additional Server

Eugene B eb at st-us.com
Thu Mar 10 07:39:28 MST 2005 

Here you can find new settings I got from BV and now works on my *. make 
sure you use correct section in dial command in extensions.conf.

Eugene.
;;=====================================================================
Some modifications since the last time.  Now that asterisk has the secret
in
outbound calls, it seems to want the proxy to authenticate to asterisk for
inbound calls.  Thats fixed with the insecure=very.  As well, its authname
not
authuser.  The username and secret variables stay the same.


--

Sip.conf

In the [general] section of the config file create a line like this:

register => <accountid>@sip.broadvoice.com:<password>:<account
id>@sip.broadvoice.com/<extension>

Replace accountid with your account or phone number, password with your
password and extension with one of your accessible extensions in the dial
plan.

BroadVoice Peer

Add a new section towards the bottom of the file to insure you don't
overwrite
any important data in [general]

[sip.broadvoice.com]
type=peer
user=phone
host=sip.broadvoice.com
fromdomain=sip.broadvoice.com
fromuser=<phone number>
secret=<register password>
username=<phone number>
insecure=very
context=from-broadvoice
authname=<phone number>
dtmfmode=inband
dtmf=inband
;Disable canreinvite if you are behind a NAT
canreinvite=no


Insure you fillin fromuser with your phone number.

/etc/hosts
Finding the right proxy

Ping the following hosts and select for the best time:

- proxy.lax.broadvoice.com
- proxy.dca.broadvoice.com
- proxy.mia.broadvoice.com

After you have chosen the one with the best ping time, do a dnslookup by
running nslookup on the hostname.
[edit]
Modifying /etc/hosts

Using the IP Address you received from nslookup add a line like this to
/etc/hosts:

{ip} sip.broadvoice.com

Insert the IP appropriately.

extensions.conf
Default Dial Plan

Put the following block in your extensions.conf as it will be your default
dial
plan:

[default]

exten => _1NXXNXXXXXX, 1, dial(SIP/${EXTEN}@sip.broadvoice.com,30)
exten => _1NXXNXXXXXX, 2, congestion()
exten => _1NXXNXXXXXX, 102, busy()


;;==================================================================================
----- Original Message ----- 
From: "Zanzamar Majere" <Phoneman at wbtllc.com>
To: <asterisk-admin at hulber.com>; "Asterisk Users Mailing List - 
Non-Commercial Discussion" <asterisk-users at lists.digium.com>
Sent: Wednesday, March 09, 2005 9:53 AM
Subject: Re: [Asterisk-Users] Broadvoice latest changes and stillnot 
working- An Additional Server


>
> 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
>> >>>_______________________________________________
>> >>>Asterisk-Users mailing list
>> >>>Asterisk-Users at lists.digium.com
>> >>>http://lists.digium.com/mailman/listinfo/asterisk-users
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>> >>>   http://lists.digium.com/mailman/listinfo/asterisk-users
>> >>
>> >>_______________________________________________
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>> >
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