Still no luck using asterisk as a proxy.
48 hours solid working on this. I'm beginning to think asterisk isn't
going
to be compatible with the provider I'm using :(
Has anyone got *any* clues as to what can cause this message? It's
definately
provider specific (voiptalk works, pipecall doesn't) but confusingly
seems to
be caused by something in the client phone app.
I guess I didn't give enough detail in my last message, so here's as
much as
I've done so far:
1. I've reconfigured to network to non-NAT (was 1:1 NAT) so there's no
rewriting going on.
2. I've tried various combinations of
'fromuser','fromdomain',
'username' and
got nowhere. There's no authuser option on the outgoing call so this
may be
the issue (in which case I'll have to use a different provider as
authuser!=username. Pity as they're the cheapest by far...).
3. Tried recompiling asterisk from source, just in case the debian
package was
broken.
I still get the error:
May 17 23:20:27 NOTICE[1110916016]: chan_sip.c:5059 handle_response:
Failed to
authenticate on INVITE to '"Tony Hoyle"
<sip:6001@213.208.99.114>;tag=as5c348356'
Relevant chunks here of data are:
[pipecall]
type=peer
secret=xxxx
username=xxxx
host=sipproxy.pipecall.com
[6001]
type=friend
username=6001
secret=xxxx
host=dynamic
context=inbound-from-local
The log looks like:
Sip read:
INVITE sip:8378@asterisk SIP/2.0
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>
Contact: <sip:6001@213.208.99.115:5060>
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1567 INVITE
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103a
Content-Length: 295
v=0
o=6001 8049593 8049593 IN IP4 213.208.99.115
s=X-Lite
c=IN IP4 213.208.99.115
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
11 headers, 13 lines
Using latest request as basis request
Sending to 213.208.99.115 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 524302, them - 1550/0, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>;tag=as7d10bfb2
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1567 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8378@213.208.99.114>
Proxy-Authenticate: Digest realm="asterisk",
nonce="7b551e23"
Content-Length: 0
to 213.208.99.115:5060
sisko*CLI>
Sip read:
ACK sip:8378@asterisk SIP/2.0
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK280F039561C44F4A93B15B494551D18A
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>;tag=as7d10bfb2
Contact: <sip:6001@213.208.99.115:5060>
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1567 ACK
Max-Forwards: 70
Content-Length: 0
9 headers, 0 lines
sisko*CLI>
Sip read:
INVITE sip:8378@asterisk SIP/2.0
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>
Contact: <sip:6001@213.208.99.115:5060>
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1568 INVITE
Proxy-Authorization: Digest
username="6001",realm="asterisk",nonce="7b551e23",response="3f2a64418952e18bbb69bb8a5189384f",uri="sip:8378@asterisk"
Max-Forwards: 70
Content-Type: application/sdp
User-Agent: X-Lite release 1103a
Content-Length: 295
v=0
o=6001 8049593 8049593 IN IP4 213.208.99.115
s=X-Lite
c=IN IP4 213.208.99.115
t=0 0
m=audio 8000 RTP/AVP 0 8 3 98 97 101
a=rtpmap:0 pcmu/8000
a=rtpmap:8 pcma/8000
a=rtpmap:3 gsm/8000
a=rtpmap:98 iLBC/8000
a=rtpmap:97 speex/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
12 headers, 13 lines
Using latest request as basis request
Sending to 213.208.99.115 : 5060 (non-NAT)
Found audio format UNKN
Found audio format ALAW
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found audio format UNKN
Found description format pcmu
Found description format pcma
Found description format gsm
Found description format iLBC
Found description format speex
Found description format telephone-event
Capabilities: us - 524302, them - 1550/0, combined - 14
Non-codec capabilities: us - 1, them - 1, combined - 1
Looking for 8378 in inbound-from-local
list_route: hop: <sip:6001@213.208.99.115:5060>
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>;tag=as47ab5787
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1568 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8378@213.208.99.114>
Content-Length: 0
to 213.208.99.115:5060
We're at 213.208.99.114 port 10306
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
12 headers, 12 lines
Reliably Transmitting:
INVITE sip:8378@sipproxy.pipecall.com SIP/2.0
Via: SIP/2.0/UDP 213.208.99.114:5060;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Contact: <sip:6001@213.208.99.114>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Mon, 17 May 2004 22:16:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 6973 6973 IN IP4 213.208.99.114
s=session
c=IN IP4 213.208.99.114
t=0 0
m=audio 10306 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 217.31.129.144:5060
sisko*CLI>
Sip read:
SIP/2.0 407 Proxy authentication required
Via: SIP/2.0/UDP 213.208.99.114;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 102 INVITE
Content-Length: 0
Proxy-Authenticate: Digest realm="213.208.99.114",
nonce="000000fc9509e3569yx1EXM7fFX+lgZ6Byvq7g==",
opaque="MTAyNjBmOWE2MTY3MTk3MQ==", stale=false, algorithm=MD5,
qop="auth"
8 headers, 0 lines
Transmitting:
ACK sip:8378@sipproxy.pipecall.com SIP/2.0
Via: SIP/2.0/UDP 213.208.99.114:5060;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Contact: <sip:6001@213.208.99.114>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 217.31.129.144:5060
We're at 213.208.99.114 port 10306
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
Reliably Transmitting:
INVITE sip:8378@sipproxy.pipecall.com SIP/2.0
Via: SIP/2.0/UDP 213.208.99.114:5060;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Contact: <sip:6001@213.208.99.114>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 103 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="8378",
realm="213.208.99.114",
algorithm="MD5", uri="sip:8378@sipproxy.pipecall.com",
nonce="000000fc9509e3569yx1EXM7fFX+lgZ6Byvq7g==",
response="66ed3e637cb5597849619365543ee80c",
opaque="MTAyNjBmOWE2MTY3MTk3MQ=="
Date: Mon, 17 May 2004 22:16:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 6973 6974 IN IP4 213.208.99.114
s=session
c=IN IP4 213.208.99.114
t=0 0
m=audio 10306 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 217.31.129.144:5060
sisko*CLI>
Sip read:
SIP/2.0 407 Proxy authentication required
Via: SIP/2.0/UDP 213.208.99.114;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 103 INVITE
Content-Length: 0
Proxy-Authenticate: Digest realm="213.208.99.114",
nonce="000000fc9509e3b3sluyMhHjWcVjVa5+JQDKTQ==",
opaque="MTNlMzAwNmY4NzM5ZjI5Nw==", stale=false, algorithm=MD5,
qop="auth"
8 headers, 0 lines
Transmitting:
ACK sip:8378@sipproxy.pipecall.com SIP/2.0
Via: SIP/2.0/UDP 213.208.99.114:5060;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Contact: <sip:6001@213.208.99.114>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 103 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 217.31.129.144:5060
We're at 213.208.99.114 port 10306
Answering with capability 2
Answering with capability 4
Answering with capability 8
Answering with non-codec capability 1
Reliably Transmitting:
INVITE sip:8378@sipproxy.pipecall.com SIP/2.0
Via: SIP/2.0/UDP 213.208.99.114:5060;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Contact: <sip:6001@213.208.99.114>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 104 INVITE
User-Agent: Asterisk PBX
Proxy-Authorization: Digest username="8378",
realm="213.208.99.114",
algorithm="MD5", uri="sip:8378@sipproxy.pipecall.com",
nonce="000000fc9509e3b3sluyMhHjWcVjVa5+JQDKTQ==",
response="a9be2b85880e3791a8b5429c5c19064b",
opaque="MTNlMzAwNmY4NzM5ZjI5Nw=="
Date: Mon, 17 May 2004 22:16:52 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 265
v=0
o=root 6973 6975 IN IP4 213.208.99.114
s=session
c=IN IP4 213.208.99.114
t=0 0
m=audio 10306 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
(no NAT) to 217.31.129.144:5060
sisko*CLI>
Sip read:
SIP/2.0 407 Proxy authentication required
Via: SIP/2.0/UDP 213.208.99.114;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 104 INVITE
Content-Length: 0
Proxy-Authenticate: Digest realm="213.208.99.114",
nonce="000000fc9509e411oOmwwi4/8N7HVCuHl8blqA==",
opaque="NmUzMGM4Mjk3YzViNA==", stale=false, algorithm=MD5,
qop="auth"
8 headers, 0 lines
Transmitting:
ACK sip:8378@sipproxy.pipecall.com SIP/2.0
Via: SIP/2.0/UDP 213.208.99.114:5060;branch=z9hG4bK7e8863fb
From: "Tony Hoyle" <sip:6001@213.208.99.114>;tag=as6e93ec5f
To: <sip:8378@sipproxy.pipecall.com>
Contact: <sip:6001@213.208.99.114>
Call-ID: 2dc65c830aaf5d615ff6cf647f59ab7b@213.208.99.114
CSeq: 104 ACK
User-Agent: Asterisk PBX
Content-Length: 0
(no NAT) to 217.31.129.144:5060
May 17 23:16:52 NOTICE[1110916016]: chan_sip.c:5059 handle_response:
Failed to
authenticate on INVITE to '"Tony Hoyle"
<sip:6001@213.208.99.114>;tag=as6e93ec5f'
sisko*CLI>
Sip read:
CANCEL sip:8378@asterisk SIP/2.0
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>
Contact: <sip:6001@213.208.99.115:5060>
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1568 CANCEL
Proxy-Authorization: Digest
username="6001",realm="asterisk",nonce="7b551e23",response="080be7e06049d363d78b565d22dff8d5",uri="sip:8378@asterisk"
Max-Forwards: 70
User-Agent: X-Lite release 1103a
Content-Length: 0
11 headers, 0 lines
Sending to 213.208.99.115 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>;tag=as47ab5787
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1568 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8378@213.208.99.114>
Content-Length: 0
to 213.208.99.115:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>;tag=as47ab5787
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1568 CANCEL
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:8378@213.208.99.114>
Content-Length: 0
to 213.208.99.115:5060
sisko*CLI>
Sip read:
ACK sip:8378@asterisk SIP/2.0
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>;tag=as47ab5787
Contact: <sip:6001@213.208.99.115:5060>
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1568 ACK
Max-Forwards: 70
Content-Length: 0
9 headers, 0 lines
sisko*CLI>
Sip read:
ACK sip:8378@asterisk SIP/2.0
Via: SIP/2.0/UDP
213.208.99.115:5060;rport;branch=z9hG4bK1B1C28C7A366423997B53E3520F71ACC
From: Tony Hoyle <sip:6001@asterisk>;tag=3751201687
To: <sip:8378@asterisk>;tag=as47ab5787
Contact: <sip:6001@213.208.99.115:5060>
Call-ID: 6D3C9176-5684-4F40-8620-D7A105CD0A42@213.208.99.115
CSeq: 1568 ACK
Max-Forwards: 70
Content-Length: 0
9 headers, 0 lines
sisko*CLI>
Tony
--
Te audire no possum. Musa sapientum fixa est in aure.
Tony Hoyle <tmh@nodomain.org> Key ID: 104D/4F4B6917 2003-09-13
Fingerprint: 063C AFB4 3026 F724 0AA2 02B8 E547 470E 4F4B 6917
