[Asterisk-Users] Avaya dialing problems
Aaron Johnson
ajohnson01 at cox.net
Wed Aug 25 14:01:59 MST 2004
Aaron Johnson wrote:
> Currently I am having 2 issues with my Avaya 4602 phone:
>
> First, the phone registers with my Asterisk server, but when I start
> dialing I get a busy signal after 4 digits. I specified in the
> dialplan on the phone to expect 10 digits and that solved that
> problem, but I still immediately get a busy after the 10th digit. The
> phone never sends a dial command to asterisk.
>
> Second, asterisk is complaining every few seconds with the message
> "Got SIP response 481 "Call Does Not Exist" back from <insert IP
> address of phone>. I don't know if it is related and I am completely
> stumped.
>
> Thanks,
>
> Aaron
I captured some of the SIP debugging info. It looks as if the phone
never really completes registering with the asterisk server.
---------------------------------------
=========================================================================
[ Booting............ -- SIP Seeding 'test1-avaya' at
test1-avaya at 192.168.1.102:5060 for 60
.......Aug 25 05:12:36 [1;31;40mWARNING[0;37;40m[16384]:
[1;37;40mchan_skinny.c[0;37;40m:[1;37;40m2568[0;37;40m
[1;37;40mreload_config[0;37;40m: Unable to get our IP address, Skinny
disabled
...............................................................................
]
[1;37;40mAsterisk Ready.
[0;37;40m*CLI> sip debug
[0;37;40mSIP Debugging Enabled
*CLI> Retransmitting #2 (no NAT):
NOTIFY sip:test1-avaya at 192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:0;branch=z9hG4bK17e38acc;rport
From: "asterisk" <sip:asterisk at 192.168.1.103:0>;tag=as0044ad38
To: <sip:test1-avaya at 192.168.1.102>
Contact: <sip:asterisk at 192.168.1.103:0>
Call-ID: 222fa122073781c12ebb7f7e3c6fca1e at 192.168.1.103
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36
Messages-Waiting: no
Voicemail: 0/0
to 192.168.1.102:5060
Retransmitting #3 (no NAT):
NOTIFY sip:test1-avaya at 192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:0;branch=z9hG4bK17e38acc;rport
From: "asterisk" <sip:asterisk at 192.168.1.103:0>;tag=as0044ad38
To: <sip:test1-avaya at 192.168.1.102>
Contact: <sip:asterisk at 192.168.1.103:0>
Call-ID: 222fa122073781c12ebb7f7e3c6fca1e at 192.168.1.103
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Event: message-summary
Content-Type: application/simple-message-summary
Content-Length: 36
Messages-Waiting: no
Voicemail: 0/0
to 192.168.1.102:5060
Sip read:
SIP/2.0 500 Server Internal Error
Call-ID: 222fa122073781c12ebb7f7e3c6fca1e at 192.168.1.103
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk at 192.168.1.103>;tag=as0044ad38
To: <sip:test1-avaya at 192.168.1.102>;tag=6b0dbdf1b2816a8
Via: SIP/2.0/UDP 192.168.1.103;branch=z9hG4bK17e38acc;rport
Content-Length: 0
Retry-After: 3
Contact: <sip:test1-avaya at 192.168.1.102>
User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
10 headers, 0 lines
Destroying call '222fa122073781c12ebb7f7e3c6fca1e at 192.168.1.103'
Sip read:
REGISTER sip:192.168.1.103:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK2306e1271
Max-Forwards: 70
Content-Length: 0
To: Asterisk test1-avaya <sip:test1-avaya at 192.168.1.103:5060>
From: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102
CSeq: 197897927 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya at 192.168.1.102>;expires=60
Allow: NOTIFY
Allow: REFER
Allow: OPTIONS
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: BYE
User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
17 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK2306e1271
From: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102
CSeq: 197897927 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya at 192.168.1.103:0>
Content-Length: 0
to 192.168.1.102:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bK2306e1271
From: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102
CSeq: 197897927 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya at 192.168.1.103:0>
WWW-Authenticate: Digest realm="asterisk", nonce="2abf33e8"
Content-Length: 0
to 192.168.1.102:5060
Scheduling destruction of call
'2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102' in 15000 ms
Sip read:
REGISTER sip:192.168.1.103:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKa8a9b10de
Max-Forwards: 70
Content-Length: 0
To: Asterisk test1-avaya <sip:test1-avaya at 192.168.1.103:5060>
From: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102
CSeq: 197897928 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya at 192.168.1.102>;expires=60
Allow: NOTIFY
Allow: REFER
Allow: OPTIONS
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: BYE
Authorization:Digest
response="8e8ffde9979b921307709167cb1b86d0",username="test1-avaya",realm="asterisk",nonce="2abf33e8",uri="sip:192.168.1.103:5060"
User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
18 headers, 0 lines
Using latest request as basis request
Sending to 192.168.1.102 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKa8a9b10de
From: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102
CSeq: 197897928 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya at 192.168.1.103:0>
Content-Length: 0
to 192.168.1.102:5060
Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKa8a9b10de
From: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102
CSeq: 197897928 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:test1-avaya at 192.168.1.103:0>;expires=60
Date: Wed, 25 Aug 2004 12:12:41 GMT
Content-Length: 0
to 192.168.1.102:5060
Scheduling destruction of call
'2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102' in 15000 ms
Sip read:
REGISTER sip:192.168.1.103:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.102;branch=z9hG4bKf145cb7d7
Max-Forwards: 70
Content-Length: 0
To: Asterisk test1-avaya <sip:test1-avaya at 192.168.1.103:5060>
From: Asterisk test1-avaya
<sip:test1-avaya at 192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67 at 192.168.1.102
CSeq: 197897929 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya at 192.168.1.102>;expires=60
Allow: NOTIFY
Allow: REFER
Allow: OPTIONS
Allow: INVITE
Allow: ACK
Allow: CANCEL
Allow: BYE
Authorization:Digest
response="8e8ffde9979b921307709167cb1b86d0",username="test1-avaya",realm="asterisk",nonce="2abf33e8",uri="sip:192.168.1.103:5060"
User-Agent: Elite 1.0 Brcm Callctrl/1.5.1.0 MxSF/v3.2.6.26
More information about the asterisk-users
mailing list