Cisco 7940 busy when offhook but not when in call with *

Asterisk Asterisk Support
1

Hello,

I am using Asterisk 1.2.9 and recently purchased a Cisco 7940 IP Phone (IOS version 08.2.00), It has 2 lines. Everything works great. I can call out and receive calls. I can also be on a call on Line 1 and will be notified with a beep in the earpeace when another call comes in. I can also see the call on the Cisco Display on line 2. Both lines on the phone are configured to use the same sip account.

My issue is when I am not in a call, but have taken the handset offhook and about to make a call and a call comes in while I am offhook, then the call will go straight to my voicemail and not to line2 (Asterisk logs say all lines buzy/congested). I noticed also that the Dial Tone dissapeared when the incomming call came in.

This also happens while i am calling another extension, but they have not yet picked up.

Very odd… anyone know a fix, or have similar issue?

Thank you.

2

Does anyone on this forum use a Cisco phone with Asterisk?

3

try sip debug and see what asterisk says to the phone. I have a feeling the phone is rejecting the call…

4

I cannot see anything obvious, besides the SIP/100-ac18 is busy.

I have my rtp.conf set to use port range 8000-8100.
I noticed on the Cisco phone, they are set to 16384-32766… even though i have specified 8000-8100 in SIPDefault.cnf. Could this be an issue?

If I ue a softphone (Express Talk). I am able to call the second line on the softphone no problmes. Are you able to call yourself on your Cisco phone?

Here’s the DEBUG of trying to call myself from extension 100 to extensoin 100 on the Cisco Phone. it has 2 lines (both set to use ext 100). But it says im Busy. It works fine if I am in a call and the other end has picked up.

lucasCLI> sip debug peer 100
SIP Debugging Enabled for IP: 192.168.2.16:5060
lucasCLI>
<-- SIP read from 192.168.2.16:5060:
INVITE sip:100@asterisk.mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK10d12eb5
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
Max-Forwards: 70
Date: Sat, 15 Jul 2006 14:21:13 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:100@192.168.2.16:5060;transport=udp
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “John Line 1” sip:100@192.168.2.16;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 273
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 19744 0 IN IP4 192.168.2.16
s=SIP Call
t=0 0
m=audio 19194 RTP/AVP 0 8 18 101
c=IN IP4 192.168.2.16
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 - 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
Sending to 192.168.2.16 : 5060 (NAT)
Reliably Transmitting (NAT) to 192.168.2.16:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK10d12eb5;received=192.168.2.16
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com;tag=as799688fc
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:100@192.168.2.1
Proxy-Authenticate: Digest realm=“asterisk.mydomain.com”, nonce="154138e8"
Content-Length: 0

Scheduling destruction of call ‘000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16’ in 15000 ms
Found user '100’
lucas*CLI>
<-- SIP read from 192.168.2.16:5060:
ACK sip:100@asterisk.mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK10d12eb5
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com;tag=as799688fc
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
Date: Sat, 15 Jul 2006 14:21:13 GMT
CSeq: 101 ACK
Content-Length: 0

— (8 headers 0 lines)—
lucas*CLI>
<-- SIP read from 192.168.2.16:5060:
INVITE sip:100@asterisk.mydomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK2f9576a5
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
Max-Forwards: 70
Date: Sat, 15 Jul 2006 14:21:13 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7940G/8.0
Contact: sip:100@192.168.2.16:5060;transport=udp
Proxy-Authorization: Digest username=“100”,realm=“asterisk.mydomain.com”,uri="sip:100@asterisk.mydomain.com",response=“9aa72a9ba1ef2ee036a5fa1383d1ee2e”,nonce=“154138e8”,algorithm=md5
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “John Line 1” sip:100@192.168.2.16;party=calling;id-type=subscriber;privacy=off;screen=yes
Supported: replaces,join,norefersub
Content-Length: 273
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 19744 0 IN IP4 192.168.2.16
s=SIP Call
t=0 0
m=audio 19194 RTP/AVP 0 8 18 101
c=IN IP4 192.168.2.16
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

— (19 headers 13 lines)—
Using INVITE request as basis request - 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
Sending to 192.168.2.16 : 5060 (NAT)
Found user '100’
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 192.168.2.16:19194
Found description format PCMU
Found description format PCMA
Found description format G729
Found description format telephone-event
Capabilities: us - 0xe (gsm|ulaw|alaw), 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 100 in sip (domain asterisk.mydomain.com)
list_route: hop: sip:100@192.168.2.16:5060;transport=udp
Transmitting (NAT) to 192.168.2.16:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK2f9576a5;received=192.168.2.16
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:100@192.168.2.1
Content-Length: 0

-- Executing Dial("SIP/100-fda6", "SIP/100|25|Ttr") in new stack

We’re at 192.168.2.1 port 8062
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 12 lines
Reliably Transmitting (NAT) to 192.168.2.16:5060:
INVITE sip:100@192.168.2.16:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK3de30596;rport
From: “John” sip:100@asterisk.mydomain.com;tag=as2064fb94
To: sip:100@192.168.2.16:5060;user=phone;transport=udp
Contact: sip:100@192.168.2.1
Call-ID: 4c0c292c19c11ff53b75718a745593fb@asterisk.mydomain.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 15 Jul 2006 14:21:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 30398 30398 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 8062 RTP/AVP 0 8 3 101
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 - - - -

-- Called 100

Retransmitting #1 (NAT) to 192.168.2.16:5060:
INVITE sip:100@192.168.2.16:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK3de30596;rport
From: “John” sip:100@asterisk.mydomain.com;tag=as2064fb94
To: sip:100@192.168.2.16:5060;user=phone;transport=udp
Contact: sip:100@192.168.2.1
Call-ID: 4c0c292c19c11ff53b75718a745593fb@asterisk.mydomain.com
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sat, 15 Jul 2006 14:21:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 260

v=0
o=root 30398 30398 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 8062 RTP/AVP 0 8 3 101
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 - - - -

lucas*CLI>
<-- SIP read from 192.168.2.16:5060:
SIP/2.0 486 Busy here
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK3de30596;rport
From: “John” sip:100@asterisk.mydomain.com;tag=as2064fb94
To: sip:100@192.168.2.16:5060;user=phone;transport=udp;tag=000dedd80b2c08ae715709d5-624b4d5e
Call-ID: 4c0c292c19c11ff53b75718a745593fb@asterisk.mydomain.com
Date: Sat, 15 Jul 2006 14:21:14 GMT
CSeq: 102 INVITE
Server: Cisco-CP7940G/8.0
Contact: sip:100@192.168.2.16:5060;transport=udp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Remote-Party-ID: “John Line 1” sip:100@192.168.2.16;party=called;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

— (12 headers 0 lines)—
– Got SIP response 486 “Busy here” back from 192.168.2.16
Transmitting (NAT) to 192.168.2.16:5060:
ACK sip:100@192.168.2.16:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK3de30596;rport
From: “John” sip:100@asterisk.mydomain.com;tag=as2064fb94
To: sip:100@192.168.2.16:5060;user=phone;transport=udp;tag=000dedd80b2c08ae715709d5-624b4d5e
Contact: sip:100@192.168.2.1
Call-ID: 4c0c292c19c11ff53b75718a745593fb@asterisk.mydomain.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

-- SIP/100-ac18 is busy

== Everyone is busy/congested at this time (1:1/0/0)
– Executing VoiceMail(“SIP/100-fda6”, “b1”) in new stack
We’re at 192.168.2.1 port 8006
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.2.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK2f9576a5;received=192.168.2.16
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com;tag=as578c1b01
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:100@192.168.2.1
Content-Type: application/sdp
Content-Length: 237

v=0
o=root 30398 30398 IN IP4 192.168.2.1
s=session
c=IN IP4 192.168.2.1
t=0 0
m=audio 8006 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 '/var/spool/asterisk/voicemail/default/1/greet' (language 'en')

lucas*CLI>
<-- SIP read from 192.168.2.16:5060:
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK3de30596;rport
From: “John” sip:100@asterisk.mydomain.com;tag=as2064fb94
To: sip:100@192.168.2.16:5060;user=phone;transport=udp
Call-ID: 4c0c292c19c11ff53b75718a745593fb@asterisk.mydomain.com
Date: Sat, 15 Jul 2006 14:21:14 GMT
CSeq: 102 INVITE
Content-Length: 0

— (8 headers 0 lines)—
– Got SIP response 481 “Call Leg/Transaction Does Not Exist” back from 192.168.2.16
Transmitting (NAT) to 192.168.2.16:5060:
ACK sip:100@192.168.2.16:5060;user=phone;transport=udp SIP/2.0
Via: SIP/2.0/UDP 192.168.2.1:5060;branch=z9hG4bK3de30596;rport
From: “John” sip:100@asterisk.mydomain.com;tag=as2064fb94
To: sip:100@192.168.2.16:5060;user=phone;transport=udp
Contact: sip:100@192.168.2.1
Call-ID: 4c0c292c19c11ff53b75718a745593fb@asterisk.mydomain.com
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

Destroying call '4c0c292c19c11ff53b75718a745593fb@asterisk.mydomain.com’
lucas*CLI>
<-- SIP read from 192.168.2.16:5060:
ACK sip:100@192.168.2.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK54408da7
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com;tag=as578c1b01
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
Max-Forwards: 70
Date: Sat, 15 Jul 2006 14:21:14 GMT
CSeq: 102 ACK
User-Agent: Cisco-CP7940G/8.0
Proxy-Authorization: Digest username=“100”,realm=“asterisk.mydomain.com”,uri="sip:100@asterisk.mydomain.com",response=“9aa72a9ba1ef2ee036a5fa1383d1ee2e”,nonce=“154138e8”,algorithm=md5
Remote-Party-ID: “John Line 1” sip:100@192.168.2.16;party=calling;id-type=subscriber;privacy=off;screen=yes
Content-Length: 0

— (12 headers 0 lines)—
– Playing ‘vm-isonphone’ (language ‘en’)
lucas*CLI>
<-- SIP read from 192.168.2.16:5060:
BYE sip:100@192.168.2.1:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK147fbf8b
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com;tag=as578c1b01
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
Max-Forwards: 70
Date: Sat, 15 Jul 2006 14:21:16 GMT
CSeq: 103 BYE
User-Agent: Cisco-CP7940G/8.0
Content-Length: 0
RTP-RxStat: Dur=3,Pkt=119,Oct=19040,LatePkt=0,LostPkt=0,AvgJit=38
RTP-TxStat: Dur=2,Pkt=102,Oct=16320
Proxy-Authorization: Digest username=“100”,realm=“asterisk.mydomain.com”,uri=“sip:100@192.168.2.1:5060”,response=“2a47a13981d85f3237966dda137cb699”,nonce=“154138e8”,algorithm=md5

— (13 headers 0 lines)—
Sending to 192.168.2.16 : 5060 (NAT)
Transmitting (NAT) to 192.168.2.16:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.16:5060;branch=z9hG4bK147fbf8b;received=192.168.2.16
From: “John Line 1” sip:100@asterisk.mydomain.com;tag=000dedd80b2c08ad69c09ff5-74171224
To: sip:100@asterisk.mydomain.com;tag=as578c1b01
Call-ID: 000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16
CSeq: 103 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:100@192.168.2.1
Content-Length: 0
X-Asterisk-HangupCause: User busy

== Spawn extension (sip, 100, 102) exited non-zero on 'SIP/100-fda6’
Destroying call '000dedd8-0b2c003d-254b90e4-060672db@192.168.2.16’
lucas*CLI> sip no debug
SIP Debugging Disabled

5

Hello,

I’ve managed the get the second line working on the phone by using the RetryDial() function in Asterisk 1.2.

So now if the phone is off-hook, and not in a call. The caller gets put on hold while it tries to continually call the off-hook extension. When I place the phone on-hook, the call comes through.

I’m fairly new to Asterisk and I guess this is normal.

Thank you for reading.