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:[EMAIL PROTECTED]>;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:[EMAIL PROTECTED],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:[EMAIL PROTECTED]>;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:[EMAIL PROTECTED],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:[EMAIL PROTECTED]>;tag=as545ccba3'
SIP debug...
-- Executing Dial("SIP/6050-132b", "SIP/[EMAIL PROTECTED]|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:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0
From: "6050" <sip:[EMAIL PROTECTED]>;tag=as545ccba3
To: <sip:[EMAIL PROTECTED]>
Contact: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
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 [EMAIL PROTECTED] com*CLI>
Sip read:
INVITE sip:[EMAIL PROTECTED] SIP/2.0
Via: SIP/2.0/UDP 64.4.192.110:5060;branch=z9hG4bK-1b5b3221
From: 6050 <sip:[EMAIL PROTECTED]>;tag=7e2776985d5a0891o0
To: <sip:[EMAIL PROTECTED]>
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="6050",realm="asterisk",nonce="42d82e9b",uri="sip: [EMAIL PROTECTED]",algorithm=MD5,response="420e39b35648a 10c 129dd4fb5f97ec47"
Contact: 6050 <sip:[EMAIL PROTECTED]: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:[EMAIL PROTECTED]>;tag=7e2776985d5a0891o0 To: <sip:[EMAIL PROTECTED]>;tag=as2f065f18 Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Contact: <sip:[EMAIL PROTECTED]> 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:[EMAIL PROTECTED]>;tag=as545ccba3 To: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;tag=as545ccba3
To: <sip:[EMAIL PROTECTED]>;tag=SD38rq699-
Call-ID: [EMAIL PROTECTED]
CSeq: 102 INVITE
WWW-Authenticate: DIGEST realm="BroadWorks",algorithm=MD5,nonce="1110353299563"
Content-Length: 0
8 headers, 0 lines Transmitting: ACK sip:[EMAIL PROTECTED] SIP/2.0 Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK3ca6ade0 From: "6050" <sip:[EMAIL PROTECTED]>;tag=as545ccba3 To: <sip:[EMAIL PROTECTED]>;tag=SD38rq699- Contact: <sip:[EMAIL PROTECTED]> Call-ID: [EMAIL PROTECTED] 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:[EMAIL PROTECTED]>;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([EMAIL PROTECTED])
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 [email protected] 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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