[Asterisk-Users] Cisco 7960 can't make/receive calls
Robert Shilston
robert at shilston.com
Fri Jan 21 09:22:26 MST 2005
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 at 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 at server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 82.33.200.166:5060;branch=z9hG4bK4597e6f2
From: "30"
<sip:30 at server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20 at server.ourdomain.com;user=phone>
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a at 82.33.200.166
Date: Fri, 21 Jan 2005 11:13:19 GMT
CSeq: 101 INVITE
User-Agent: CSCO/6
Contact: <sip:30 at 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 at server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20 at server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a at 82.33.200.166
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:20 at 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 at 82.33.200.166' in 15000 ms
Found user '30'
server*CLI>
Sip read:
ACK sip:20 at server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.123.123:5060;branch=z9hG4bK4597e6f2
From: "30"
<sip:30 at server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20 at server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a at 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 at server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 82.33.200.166:5060;branch=z9hG4bK54a10ee6
From: "30"
<sip:30 at server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20 at server.ourdomain.com;user=phone>
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a at 82.33.200.166
Date: Fri, 21 Jan 2005 11:13:19 GMT
CSeq: 102 INVITE
User-Agent: CSCO/6
Contact: <sip:30 at 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 at 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 at server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20 at server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a at 82.33.200.166
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:20 at srv.ext.ip.addr>
Content-Length: 0
to 82.33.200.166:5060
server*CLI>
Sip read:
ACK sip:20 at server.ourdomain.com;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.123.123:5060;branch=z9hG4bK54a10ee6
From: "30"
<sip:30 at server.ourdomain.com>;tag=00078599323d00057ddbf1bd-56b51a90
To: <sip:20 at server.ourdomain.com;user=phone>;tag=as63cf85a2
Call-ID: 00078599-323d0005-0dc35ec5-5770d68a at 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 at 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 at 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 at server.ourdomain.com>;tag=as1348ad44
To: <sip:30 at 82.33.200.166:5060;user=phone>
Contact: <sip:23 at srv.ext.ip.addr>
Call-ID: 56cbde81699aebed514115512a263413 at 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 at server.ourdomain.com>;tag=as1348ad44
To: <sip:30 at 82.33.200.166:5060;user=phone>
Call-ID: 56cbde81699aebed514115512a263413 at 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 at 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 at server.ourdomain.com>;tag=as1348ad44
To: <sip:30 at 82.33.200.166:5060;user=phone>
Contact: <sip:23 at srv.ext.ip.addr>
Call-ID: 56cbde81699aebed514115512a263413 at 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 at server.ourdomain.com'
server*CLI>
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