Cisco 7940 phones always busy/congested?

Asterisk Asterisk Support
1

Hey guys.

I’ve set up two Cisco 7940s - normally they use the SCCP/skinny protocol, but I’ve managed to get the SIP firmware (v8.2) installed on them, and they both get registered with Asterisk properly, and they can both make calls which Asterisk handles. However, trying to actually call the phone always results in this message (one phone’s SIP name is colin, the other is ivan):

– Executing Dial(“SIP/colin-0889b048”, “SIP/ivan”) in new stack
Sep 8 09:30:01 NOTICE[20914]: app_dial.c:1069 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing Playback(“SIP/colin-0889b048”, “tt-allbusy”) in new stack
– Playing ‘tt-allbusy’ (language ‘en’)

The SIP.conf entries for these two phones are:

[ivan]
type=friend
username=ivan
secret=password
host=dynamic
context=internal
callerid="ivan"
pickupgroup=1
callgroup=1

[colin]
type=friend
username=colin
secret=password
host=dynamic
context=internal
callerid="colin"
pickupgroup=1
callgroup=1

Has anyone had any problems like this before?

2

Here’s a typical SIP debug when calling:

<-- SIP read from 192.168.1.139:50718:
INVITE sip:*675@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK6c8ebba7
From: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
To: sip:*675@192.168.1.20
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
Max-Forwards: 70
Date: Fri, 08 Sep 2006 10:00:28 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:ivan@192.168.1.139:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 6640 0 IN IP4 192.168.1.139
s=SIP Call
t=0 0
m=audio 30470 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.139
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

— (17 headers 13 lines)—
Using INVITE request as basis request - 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
Sending to 192.168.1.139 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.1.139:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK6c8ebba7;received=192.168.1.139
From: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
To: sip:*675@192.168.1.20;tag=as170fcb35
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:*675@192.168.1.20
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="39586108"
Content-Length: 0

Scheduling destruction of call ‘00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139’ in 15000 ms
Found user 'ivan’
asterisk*CLI>
<-- SIP read from 192.168.1.139:50726:
ACK sip:*675@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK6c8ebba7
From: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
To: sip:*675@192.168.1.20;tag=as170fcb35
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
Date: Fri, 08 Sep 2006 10:00:28 GMT
CSeq: 101 ACK
Content-Length: 0

— (8 headers 0 lines)—
asterisk*CLI>
<-- SIP read from 192.168.1.139:50718:
INVITE sip:*675@192.168.1.20 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK03200203
From: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
To: sip:*675@192.168.1.20
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
Max-Forwards: 70
Date: Fri, 08 Sep 2006 10:00:28 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:ivan@192.168.1.139:5060;transport=udp
Proxy-Authorization: Digest username=“ivan”,realm=“asterisk”,uri=“sip:*675@192.168.1.20”,response=“64fb82dc4b3dd0555528e2fc60eda1a1”,nonce=“39586108”,algorithm=MD5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces,join,norefersub
Content-Length: 274
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 6640 0 IN IP4 192.168.1.139
s=SIP Call
t=0 0
m=audio 30470 RTP/AVP 0 8 18 101
c=IN IP4 192.168.1.139
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/0
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv

— (18 headers 13 lines)—
Using INVITE request as basis request - 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
Sending to 192.168.1.139 : 5060 (non-NAT)
Found user 'ivan’
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Peer audio RTP is at port 192.168.1.139:30470
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x10c (ulaw|alaw|g729)/video=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for *675 in internal (domain 192.168.1.20)
list_route: hop: sip:ivan@192.168.1.139:5060;transport=udp
Transmitting (no NAT) to 192.168.1.139:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK03200203;received=192.168.1.139
From: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
To: sip:*675@192.168.1.20
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:*675@192.168.1.20
Content-Length: 0

-- Executing GotoIf("SIP/ivan-0889aff0", "1?2:3") in new stack
-- Goto (internal,*675,2)
-- Executing Dial("SIP/ivan-0889aff0", "SIP/colin") in new stack

Destroying call '7351326c073f3bd54157435b79514ce2@127.0.0.1’
Sep 8 10:58:57 NOTICE[21801]: app_dial.c:1069 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
– Executing Playback(“SIP/ivan-0889aff0”, “tt-allbusy”) in new stack
We’re at 192.168.1.20 port 17950
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.1.139:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK03200203;received=192.168.1.139
From: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
To: sip:*675@192.168.1.20;tag=as3b73ac82
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:*675@192.168.1.20
Content-Type: application/sdp
ontent-Length: 240

v=0
o=root 20766 20766 IN IP4 192.168.1.20
s=session
c=IN IP4 192.168.1.20
t=0 0
m=audio 17950 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

-- Playing 'tt-allbusy' (language 'en')

asterisk*CLI>
<-- SIP read from 192.168.1.139:50718:
ACK sip:*675@192.168.1.20:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.139:5060;branch=z9hG4bK4a68452a
From: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
To: sip:*675@192.168.1.20;tag=as3b73ac82
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
Max-Forwards: 70
Date: Fri, 08 Sep 2006 10:00:29 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7940G/8.0
Proxy-Authorization: Digest username=“ivan”,realm=“asterisk”,uri=“sip:*675@192.168.1.20”,response=“64fb82dc4b3dd0555528e2fc60eda1a1”,nonce=“39586108”,algorithm=MD5
Content-Length: 0

— (11 headers 0 lines)—
== Primary D-Channel on span 1 down
Sep 8 10:59:00 WARNING[20777]: chan_zap.c:2506 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway!
== Primary D-Channel on span 1 down
Sep 8 10:59:05 WARNING[20777]: chan_zap.c:2506 pri_find_dchan: No D-channels available! Using Primary channel 3 as D-channel anyway!
– Executing Hangup(“SIP/ivan-0889aff0”, “”) in new stack
== Spawn extension (internal, *675, 104) exited non-zero on 'SIP/ivan-0889aff0’
set_destination: Parsing sip:ivan@192.168.1.139:5060;transport=udp for address/port to send to
set_destination: set destination to 192.168.1.139, port 5060
Reliably Transmitting (no NAT) to 192.168.1.139:5060:
BYE sip:ivan@192.168.1.139:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK35593151;rport
From: sip:*675@192.168.1.20;tag=as3b73ac82
To: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
Contact: sip:*675@192.168.1.20
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

asterisk*CLI>
<-- SIP read from 192.168.1.139:50718:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.20:5060;branch=z9hG4bK35593151;rport
From: sip:*675@192.168.1.20;tag=as3b73ac82
To: “ivan” sip:ivan@192.168.1.20;tag=00115c2c16c000025194483e-4b6cb1d7
Call-ID: 00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139
Date: Fri, 08 Sep 2006 10:00:37 GMT
CSeq: 102 BYE
Server: Cisco-CP7940G/8.0
Content-Length: 0

— (9 headers 0 lines)—
Destroying call ‘00115c2c-16c00002-016e4d12-47ca4d97@192.168.1.139’

3

sip show peers seems to say that the phones aren’t registering as peers (even though in sip.conf they’re designed to become both peers and users when you specify type=friend, right?)

sip show peers shows this:

colin/colin (Unspecified) D 0 Unmonitored
ivan/ivan (Unspecified) D 0 Unmonitored
andrew/andrew 192.168.1.141 D 5060 Unmonitored

andrew is a budgetone IP phone that works fine with Asterisk.

Please? Anyone had this problem before?

4

If your phones aren’t registering, obviously you’ll never be able to call them. Please post the section in your sip.conf file that describes the cisco phones as well as the SIP<mac_address>.cnf file that the phone picks up from your tftp server to boot.

Also, when you reboot the phones, do you see any messages on the asterisk console that look like there might be a username/password error?