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Robert Shilston
2005-Jan-21 09:22 UTC
[Asterisk-Users] Cisco 7960 can't make/receive calls

I've got three 7960s running v6 SIP firmware.  My Asterisk setup has 
worked fine with grandstream devices, and basically, we're just 
upgrading to use nicer phones.

Whilst I can make/receive calls from the 7960 to/from gossiptel).

When I try to place a call, I get the following

Jan 21 11:09:23 NOTICE[19688]: chan_sip.c:7271 handle_request: Failed to 
authenticate user "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d000750732f5f-2c61cb72

We're running Asterisk CVS-v1-0-01/18/05-23:43:27

The SIP<mac>.cnf file contains:
# Proxy Server
proxy1_address: "ginger.assanka.com"

# Line 1 Settings
line1_name: 30
line1_displayname: 30
line1_authname: 30
line1_password: "ciscopassword"


And sip.conf looks like:

[30]
type=friend
username=30
secret=ciscopassword
context=ourphones
host=dynamic
canreinvite=no
nat=yes
mailbox=1



As I said, we've got a couple of grandstream devices working perfectly, 
and we're just trying to upgrade them.  Both ends are behind NAT, with 
the server being the DMZ.  Budgetone 102 and a Handytone 486 both work 
fine.  I've been battling with this for a couple of days and am getting 
no-where.  Any suggestions?

A full transcript when trying to place a call from the Cisco is as 
follows, where
   10.11.185.11 = internal IP of asterisk server (DMZ)
   192.168.123.123 = internal IP of cisco phone
   82.33.200.166 = external IP of cisco phone (DMZ)

server*CLI>

Sip read:
INVITE sip:20@server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 82.33.200.166:5060;branch=z9hG4bK4597e6f2
From: "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20@server.ourdomain.com;user=phone>
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166
Date: Fri, 21 Jan 2005 11:13:19 GMT
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:30@82.33.200.166:5060>
Expires: 180
Content-Type: application/sdp
Content-Length: 247
Accept: application/sdp

v=0
o=Cisco-SIPUA 286 22351 IN IP4 82.33.200.166
s=SIP Call
c=IN IP4 82.33.200.166
t=0 0
m=audio 29280 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

13 headers, 11 lines
Using latest request as basis request
Sending to 82.33.200.166 : 5060 (NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 82.33.200.166:29280
Found description format G729
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10c 
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Reliably Transmitting (NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 
82.33.200.166:5060;branch=z9hG4bK4597e6f2;received=82.33.200.166;rport=5060
From: "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20@server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:20@srv.ext.ip.addr>
Proxy-Authenticate: Digest realm="server.ourdomain.com",
nonce="57d1bb6f"
Content-Length: 0


  to 82.33.200.166:5060
Scheduling destruction of call 
'00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166' in 15000 ms
Found user '30'
server*CLI>

Sip read:
ACK sip:20@server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.123.123:5060;branch=z9hG4bK4597e6f2
From: "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20@server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166
Date: Fri, 21 Jan 2005 11:13:19 GMT
CSeq: 101 ACK
Content-Length: 0


8 headers, 0 lines
server*CLI>

Sip read:
INVITE sip:20@server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 82.33.200.166:5060;branch=z9hG4bK54a10ee6
From: "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20@server.ourdomain.com;user=phone>
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166
Date: Fri, 21 Jan 2005 11:13:19 GMT
CSeq: 102 INVITE
User-Agent: CSCO/6
Contact: <sip:30@82.33.200.166:5060>
Proxy-Authorization: Digest 
username="30",realm="server.ourdomain.com",uri="sip:10.11.185.11",response="7a4852682ebafcd5ac9db349e1fd480a",nonce="57d1bb6f",algorithm=md5
Expires: 180
Content-Type: application/sdp
Content-Length: 247

v=0
o=Cisco-SIPUA 286 22351 IN IP4 82.33.200.166
s=SIP Call
c=IN IP4 82.33.200.166
t=0 0
m=audio 29280 RTP/AVP 18 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

13 headers, 11 lines
Using latest request as basis request
Sending to 82.33.200.166 : 5060 (NAT)
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 82.33.200.166:29280
Found description format G729
Found description format PCMU
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x10e (gsm|ulaw|alaw|g729), peer - audio=0x10c 
(ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x10c (ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined - 
0x1 (g723)
Found user '30'
Jan 21 11:13:20 NOTICE[19688]: chan_sip.c:7271 handle_request: Failed to 
authenticate user "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
Reliably Transmitting (NAT):
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 
82.33.200.166:5060;branch=z9hG4bK54a10ee6;received=82.33.200.166;rport=5060
From: "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20@server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:20@srv.ext.ip.addr>
Content-Length: 0


  to 82.33.200.166:5060
server*CLI>

Sip read:
ACK sip:20@server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.123.123:5060;branch=z9hG4bK54a10ee6
From: "30" 
<sip:30@server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20@server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166
Date: Fri, 21 Jan 2005 11:13:19 GMT
CSeq: 102 ACK
Content-Length: 0


8 headers, 0 lines
Destroying call '00078599-323d0005-0dc35ec5-5770d68a@82.33.200.166'
server*CLI>



And when receiving a call:

     -- Executing Dial("SIP/RobHardPhone-6f23", "SIP/30|20")
in new stack
We're at srv.ext.ip.addr port 10592
Video is at srv.ext.ip.addr port 14740
Answering/Requesting with root capability 0x100 (g729)
Answering with preferred capability 0x4 (ulaw)
Answering with preferred capability 0x8 (alaw)
Answering with preferred capability 0x2 (gsm)
Answering with non-codec capability 0x1 (telephone-event)
12 headers, 13 lines
Reliably Transmitting:
INVITE sip:30@82.33.200.166:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP srv.ext.ip.addr:5060;branch=z9hG4bK7875d9c7;rport
From: "Robert Shilston"
<sip:23@server.ourdomain.com>;tag=as1348ad44
To: <sip:30@82.33.200.166:5060;user=phone>
Contact: <sip:23@srv.ext.ip.addr>
Call-ID: 56cbde81699aebed514115512a263413@server.ourdomain.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Date: Fri, 21 Jan 2005 11:44:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Content-Type: application/sdp
Content-Length: 289

v=0
o=root 1650 1650 IN IP4 srv.ext.ip.addr
s=session
c=IN IP4 srv.ext.ip.addr
t=0 0
m=audio 10592 RTP/AVP 18 0 8 3 101
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
  (NAT) to 82.33.200.166:5060
     -- Called 30
server*CLI>


Sip read:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 10.11.185.11:5060;branch=z9hG4bK7875d9c7;rport
From: "Robert Shilston"
<sip:23@server.ourdomain.com>;tag=as1348ad44
To: <sip:30@82.33.200.166:5060;user=phone>
Call-ID: 56cbde81699aebed514115512a263413@server.ourdomain.com
Date: Fri, 21 Jan 2005 11:44:31 GMT
Warning: 399 Bad Request - 'Invalid SDP information'
CSeq: 102 INVITE
Content-Length: 0


9 headers, 0 lines
     -- Got SIP response 400 "Bad Request" back from 82.33.200.166
Transmitting:
ACK sip:30@82.33.200.166:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP srv.ext.ip.addr:5060;branch=z9hG4bK7875d9c7;rport
From: "Robert Shilston"
<sip:23@server.ourdomain.com>;tag=as1348ad44
To: <sip:30@82.33.200.166:5060;user=phone>
Contact: <sip:23@srv.ext.ip.addr>
Call-ID: 56cbde81699aebed514115512a263413@server.ourdomain.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Content-Length: 0

  (NAT) to 82.33.200.166:5060
     -- SIP/30-f1cf is circuit-busy
   == Everyone is busy/congested at this time
Destroying call '56cbde81699aebed514115512a263413@server.ourdomain.com'
server*CLI>

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