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Why is GOIP not registered on freepbx?
Greetings
Available:
Asterisk 13.10.0 assembled from raw materials - 192.168.1.100
Freepbx 13.0.170 - 192.168.1.101
Goip 8 GST1610-1.01-55
Stubbornly does not want to swear goip on the PBX, swears at the wrong password
Trunk:
username=600
authuser=600
defaultuser=600
secret=600
host=dynamic
type=peer
nat=no
qualify=yes
canreinvite=yes
insecure=port,invite
dtmfmode=auto
disallow=all
allow=alaw
Config by line
Authentication ID 600
Password - 600
Routing Prefix 1
SIP Proxy 192.168.1.100
SIP Registrar Server 192.168.1.100
Re-register Period (s) 60
Phone Number 600
Display Name 600
Outbound Proxy
Home Domain
�ò�^B^@^D^@^@^@^@^@^@^@^@^@��^@^@^A^@^@^@^W��W�ڲ
^@�^A^@^@�^A^@^@@�\Ni�8?^P^C��^H^@E^@^A�^@^@@^@@^Q�Q��^Ae��^Ad^S�^S�^An��REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1331228219
From: "600" <sip:[email protected]>;tag=1091296080
To: "600" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 54 REGISTER
Contact: <sip:[email protected]:5060>;expires=60
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0
^W��W��
^@+^B^@^@+^B^@^@8?^P^C��@�\Ni�^H^@E^@^B^]˵^@^@@^Q)^A��^Ad��^Ae^Sĵ^Sĵ^B �wSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1331228219;received=192.168.1.101
From: "600" <sip:[email protected]>;tag=1091296080
To: "600" <sip:[email protected]>;tag=as062a10d8
Call-ID: [email protected]
CSeq: 54 REGISTER
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="603f222d"
Content-Length: 0
^W��W��
^@.^B^@^@.^B^@^@@�\Ni�8?^P^C��^H^@E^@^B ^@^@@^@@^Q����^Ae��^Ad^S�^S�^B^Li
REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK381485677
From: "600" <sip:[email protected]>;tag=1091296080
To: "600" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 55 REGISTER
Contact: <sip:[email protected]:5060>;expires=60
Authorization: Digest username="600", realm="asterisk", nonce="603f222d", uri="sip:192.168.1.100", response="ae249072b6618522d8ed81ca30a749f0", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
^W��W^W�
^@�^A^@^@�^A^@^@8?^P^C��@�\Ni�^H^@E^@^A�˶^@^@@^Q)P��^Ad��^Ae^SĶ^SĶ^A�<,SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK381485677;received=192.168.1.101
From: "600" <sip:[email protected]>;tag=1091296080
To: "600" <sip:[email protected]>;tag=as062a10d8
Call-ID: [email protected]
CSeq: 55 REGISTER
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
^Y��WH�^A^@<^@^@^@<^@^@^@@�\Ni�8?^P^C��^H^@E^@^@^_^@^@@^@@^Q�^W��^A^B��^Ad^S�^S�^@^KG�
^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^Z��W�B^H^@�^A^@^@�^A^@^@@�\Ni�8?^P^C��^H^@E^@^A^?^@^@@^@@^Q�T��^Ae��^Ad^S�^S�^AkU�REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1190187642
From: "600" <sip:[email protected]>;tag=335421696
To: "600" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 56 REGISTER
Contact: <sip:[email protected]:5060>;expires=0
Max-Forwards: 30
User-Agent: dble
Expires: 0
Content-Length: 0
^Z��W^SD^H^@*^B^@^@*^B^@^@8?^P^C��@�\Ni�^H^@E^@^B^\˷^@^@@^Q)^@��^Ad��^Ae^Sķ^Sķ^B^H^FKSIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1190187642;received=192.168.1.101
From: "600" <sip:[email protected]>;tag=335421696
To: "600" <sip:[email protected]>;tag=as062a10d8
Call-ID: [email protected]
CSeq: 56 REGISTER
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="51e34e73"
Content-Length: 0
^^��W�g^H^@-^B^@^@-^B^@^@@�\Ni�8?^P^C��^H^@E^@^B^_^@^@@^@@^Q����^Ae��^Ad^S�^S�^B^K�^]REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1635581886
From: "600" <sip:[email protected]>;tag=22337336
To: "600" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 58 REGISTER
Contact: <sip:[email protected]:5060>;expires=60
Authorization: Digest username="600", realm="asterisk", nonce="59168475", uri="sip:192.168.1.100", response="ebe7c280677e496f52575ee22bd1e141", algorithm=MD5
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0
^^��W^Hi^H^@�^A^@^@�^A^@^@8?^P^C��@�\Ni�^H^@E^@^A�˹^@^@@^Q)N��^Ad��^Ae^SĹ^SĹ^A�HTSIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK1635581886;received=192.168.1.101
From: "600" <sip:[email protected]>;tag=22337336
To: "600" <sip:[email protected]>;tag=as062a10d8
Call-ID: [email protected]
CSeq: 58 REGISTER
Server: Asterisk PBX 13.10.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
#��W^A�^A^@<^@^@^@<^@^@^@@�\Ni�8?^P^C��^H^@E^@^@^_^@^@@^@@^Q�^W��^A^B��^Ad^S�^S�^@^KG�
^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@^@(��W^@�^H^@�^A^@^@�^A^@^@@�\Ni�8?^P^C��^H^@E^@^A�^@^@@^@@^Q�S��^Ae��^Ad^S�^S�^Al��REGISTER sip:192.168.1.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.101:5060;branch=z9hG4bK406696794
From: "600" <sip:[email protected]>;tag=938554630
To: "600" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 59 REGISTER
Contact: <sip:[email protected]:5060>;expires=60
Max-Forwards: 30
User-Agent: dble
Expires: 60
Content-Length: 0
[2016-08-22 09:50:24] NOTICE[4043]: chan_sip.c:28468 handle_request_register: Registration from '"600" <sip:[email protected]>' failed for '192.168.1.101:5060' - Wrong password
[2016-08-22 09:50:24] DEBUG[4043]: chan_sip.c:28914 handle_request_do: SIP message could not be handled, bad request: [email protected]
[2016-08-22 09:50:28] NOTICE[4043]: chan_sip.c:28468 handle_request_register: Registration from '"600" <sip:[email protected]>' failed for '192.168.1.101:5060' - Wrong password
[2016-08-22 09:50:28] DEBUG[4043]: chan_sip.c:28914 handle_request_do: SIP message could not be handled, bad request: [email protected]
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host=dynamic and type=peer are incompatible with each other
in your case you need type=friend
and remove insecure=port,invite
If it’s still relevant,
then check on which port you are listening to sip now, since by default FreePBX port is 5160.
Try registering to 5160 or change to 5060 in FreePBX Settings-Asterisk Sip Settings->Chan Sip Settings->Bind Port tab.
I hope that my situation will help someone, in which I also received the answer "wrong password" from asterisk with goip-4, although everything was rechecked several times and it works on the first module, but not on the others.
The essence of the problem is that when cloning a trunk, freepbx creates it disabled:
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