Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
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 'PP
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=zjh018g8f8
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=420e39b35648a10c
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
[Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server
CHris, I had the exact same problem with the exact same error. My password was entered incorrectly in context section. The register line had the correct password. That is why you get incoming calls. and not outgoing. Jerry ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
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=XXX
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 'PP
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=420e39b35648a10c
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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Thank you for the response. I still have the errors mentioned below, sip
response and Failed to authenticate on INVITE
[PP]
type=peer
username=PP
fromuser=PP
authuser=PP
fromdomain=sip.broadvoice.com
secret=XX
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=XXX
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 'PP
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=420e39b35648a10c
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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX 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? ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Broadvoice latest changes and still not working-An Additional Server
Jerry- Thank you I accidently sent my password on the LISTSERV last night so I just changed (pasted) the new one in. Still the same problem... Mar 9 09:51:13 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to 'Chris Nibeck sip:[EMAIL PROTECTED];tag=as4b70f2e7' Incoming works fine still. Anyone can call me at that number. Please do. It is a free call from another BV account. Chris On Mar 9, 2005, at 7:42 AM, Jerry Geis wrote: CHris, I had the exact same problem with the exact same error. My password was entered incorrectly in context section. The register line had the correct password. That is why you get incoming calls. and not outgoing. Jerry ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
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=x
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=XXX
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 'PP
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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX 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? ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Fwd: Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****SOLVED****
This configuration solved my problem. I could have sworn I tried this before. I guess not. I did not need to apply the patch. Also, I am using a regular Registration setup in my sip.conf not broadvoice's funky one... The only thing I can surmise is that order of the variables matters. This is what worked for me: [PP] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=PP secret=XX username=PP insecure=very context=sip authname=PP dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no Thank you On Wednesday 09 March 2005 08:23 am, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX 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? ___ Asterisk-Users mailing list Asterisk-Users@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 --- ___ Asterisk-Users mailing list Asterisk-Users@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
***SOLVED*** [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****Solved*****!
thank you everyone! It does not seen that it was configuration problems at all. It appears it was the CVS that I was using from yesterday. I decided to start over, downloaded the latest CVS, recompiled, and voila! * started working Indeed even a Cisco ATA that was never working before started working! Thanks to everyone! Chris On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote: there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX 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? ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
Re: ***SOLVED*** [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server ****Solved*****!
I concur. I rebuilt today and now I seem to be able to dial out. MARK. Chris Nibeck wrote: thank you everyone! It does not seen that it was configuration problems at all. It appears it was the CVS that I was using from yesterday. I decided to start over, downloaded the latest CVS, recompiled, and voila! * started working Indeed even a Cisco ATA that was never working before started working! Thanks to everyone! Chris On Mar 9, 2005, at 10:28 AM, Chris Nibeck wrote: there are two of us with the same problem so I will answer for me. Yes I tried the below instructions. The current thinking by multiple people is * never tries authenticating so removing the FQDN will force * to go to the related section named by either a phone number or a non Fully Qualified Domain Name. But I still don't have it working so who knows. Anyone that wishes to call me via BV my number is 8475100139 and it is up. Chris On Mar 9, 2005, at 9:23 AM, Mike Matthews wrote: Have you tried this: http://foo.robotics.net/mediawiki-1.3.10/index.php/Asterisk_Setup Zanzamar Majere wrote: Thank you for the response. I still have the errors mentioned below, sip response and Failed to authenticate on INVITE [PP] type=peer username=PP fromuser=PP authuser=PP fromdomain=sip.broadvoice.com secret=XX 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? ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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 ___ Asterisk-Users mailing list Asterisk-Users@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
[Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
I have been going crazy with this also since Sat. Our server was working perfectly with BV but will now not place calls to BV. Incoming from BV works fine. I felt sad rebooting it today, it had been running for almost 200 days! Here is my error message from the console... Notice I am running today's CVS Connected to Asterisk CVS-03/08/05-14:32:39 currently running on com (pid = 1624) -- Executing Dial(SIP/6050-5bc9, SIP/[EMAIL PROTECTED]|30) in new stack -- Called [EMAIL PROTECTED] Mar 8 23:11:55 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '6050 sip:[EMAIL PROTECTED];tag=as20911f6e' I have tried many versions of sip.conf, here is the current... [sip.broadvoice.com] type=peer user=phone host=sip.broadvoice.com fromdomain=sip.broadvoice.com fromuser=8475100139 secret=blah username=8475100139 insecure=very context=default authname=8475100139 dtmfmode=inband dtmf=inband ;Disable canreinvite if you are behind a NAT canreinvite=no I have tried the different proxies proxy.dca.broadvoice.com, lax, mia, and was originally using chi when the system worked. BV told me Mon that chi is considered a test server that should not be used for production, it is expected to go up and down. My hosts file points to one of the working ones. I verified my account through a softphone. It works fine to BV. There is something wrong with the authentication. Here is the SIP debug... -- Executing Dial(SIP/6050-019c, SIP/[EMAIL PROTECTED]|30) in new stack We're at xxx.xxx.xxx.xxx port 16776 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=z9hG4bK5364668c From: 6050 sip:[EMAIL PROTECTED];tag=as292b9469 To: sip:[EMAIL PROTECTED] Contact: sip:[EMAIL PROTECTED] Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 09 Mar 2005 05:35:15 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 205 v=0 o=root 3501 3501 IN IP4 xxx.xxx.xxx.xxx s=session c=IN IP4 xxx.xxx.xxx.xxx t=0 0 m=audio 16776 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: SIP/2.0 100 Trying Via: SIP/2.0/UDP xxx.xxx.xxx.xxx:5060;branch=z9hG4bK5364668c From: 6050 sip:[EMAIL PROTECTED];tag=as292b9469 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=z9hG4bK5364668c From: 6050 sip:[EMAIL PROTECTED];tag=as292b9469 To: sip:[EMAIL PROTECTED];tag=SD38ad399- Call-ID: [EMAIL PROTECTED] CSeq: 102 INVITE WWW-Authenticate: DIGEST realm=BroadWorks,algorithm=MD5,nonce=1110346372627 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=z9hG4bK5364668c From: 6050 sip:[EMAIL PROTECTED];tag=as292b9469 To: sip:[EMAIL PROTECTED];tag=SD38ad399- 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 8 23:35:15 NOTICE[15376]: chan_sip.c:5047 handle_response: Failed to authenticate on INVITE to '6050 sip:[EMAIL PROTECTED];tag=as292b9469' TIA Chris ___ Asterisk-Users mailing list Asterisk-Users@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
Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
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:
[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
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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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
First, thanks for your help.
I have been changing these to different values but not getting it.
Could you further amplify your statement...
Try changing yours to say broadvoice and then the corresponding
section in sip.conf.
Thanks!
Chris
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:
[360350]
type=peer
dtmfmode=inband
username=360350
fromuser=360350
secret=XX
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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Re: [Asterisk-Users] Broadvoice latest changes and still not working- An Additional Server
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=zjh018g8f8
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=420e39b35648a10c
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
