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Aaron Johnson

2004-Aug-25 10:42 UTC

[Asterisk-Users] Avaya dialing problems

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

Aaron Johnson

2004-Aug-25 12:21 UTC

head link

[Asterisk-Users] Avaya dialing problems (w/ SIP debugging)

I ran SIP debug on the asterisk server and this is what I got.  It looks 
as the avaya phone is trying and retrying to register, even though the 
phone shows that it is registered.  I'm stumped.
_______________________________________________________________________________________
[0;37;40mAsterisk CVS-D2004.06.20.07.00.00-06/22/04-10:57:30, Copyright 
(C) 1999-2004 Digium.

Written by Mark Spencer <markster@digium.com>

========================================================================
[ Booting............    -- SIP Seeding 'test1-avaya' at 
test1-avaya@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@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:0;branch=z9hG4bK17e38acc;rport
From: "asterisk" <sip:asterisk@192.168.1.103:0>;tag=as0044ad38
To: <sip:test1-avaya@192.168.1.102>
Contact: <sip:asterisk@192.168.1.103:0>
Call-ID: 222fa122073781c12ebb7f7e3c6fca1e@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@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:0;branch=z9hG4bK17e38acc;rport
From: "asterisk" <sip:asterisk@192.168.1.103:0>;tag=as0044ad38
To: <sip:test1-avaya@192.168.1.102>
Contact: <sip:asterisk@192.168.1.103:0>
Call-ID: 222fa122073781c12ebb7f7e3c6fca1e@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@192.168.1.103
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk@192.168.1.103>;tag=as0044ad38
To: <sip:test1-avaya@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@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@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@192.168.1.103:5060>
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897927 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897927 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897927 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya@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@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@192.168.1.103:5060>
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897928 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897928 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897928 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:test1-avaya@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@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@192.168.1.103:5060>
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897929 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya@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=z9hG4bKf145cb7d7
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897929 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:test1-avaya@192.168.1.103:0>;expires=60
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=z9hG4bKf145cb7d7
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897929 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:test1-avaya@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@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=z9hG4bK74f881ea3
Max-Forwards: 70
Content-Length: 0
To: Asterisk test1-avaya <sip:test1-avaya@192.168.1.103:5060>
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897930 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya@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

Aaron Johnson

2004-Aug-25 14:01 UTC

head link

[Asterisk-Users] Avaya dialing problems

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@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@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:0;branch=z9hG4bK17e38acc;rport
From: "asterisk" <sip:asterisk@192.168.1.103:0>;tag=as0044ad38
To: <sip:test1-avaya@192.168.1.102>
Contact: <sip:asterisk@192.168.1.103:0>
Call-ID: 222fa122073781c12ebb7f7e3c6fca1e@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@192.168.1.102 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.103:0;branch=z9hG4bK17e38acc;rport
From: "asterisk" <sip:asterisk@192.168.1.103:0>;tag=as0044ad38
To: <sip:test1-avaya@192.168.1.102>
Contact: <sip:asterisk@192.168.1.103:0>
Call-ID: 222fa122073781c12ebb7f7e3c6fca1e@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@192.168.1.103
CSeq: 102 NOTIFY
From: "asterisk" <sip:asterisk@192.168.1.103>;tag=as0044ad38
To: <sip:test1-avaya@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@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@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@192.168.1.103:5060>
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897927 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897927 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897927 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya@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@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@192.168.1.103:5060>
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897928 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897928 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:test1-avaya@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@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
To: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=as0a2edad0
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897928 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 60
Contact: <sip:test1-avaya@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@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@192.168.1.103:5060>
From: Asterisk test1-avaya 
<sip:test1-avaya@192.168.1.103:5060>;tag=dbc5a65cf5c1b60
Call-ID: 2c87ec00ca20bd9ff98558cb237fda67@192.168.1.102
CSeq: 197897929 REGISTER
Contact: Asterisk test1-avaya <sip:test1-avaya@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

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