I have been at this for days now, I cant figure our why the FreeSWITCH-> Asterisk SIP registration is failing as the passwords look ok. Any help is much appreciated.​



Asterisk(10.237.192.53) sip.conf
[gs-sbc1]
username=gsvoice01
type=friend
insecure=port,invite
secret=ipscape@2014
qualify=yes
host=10.237.192.68
;192.168.202.60
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=alaw

FreeSWITCH(10.237.192.68) sip internal profile
<include>
  <gateway name="ips_voice">
    <param name="username" value="gsvoice01"/>
    <param name="realm" value="10.237.192.53"/>
    <param name="from-domain" value="10.237.192.68"/>
    <param name="password" value="ipscape@2014"/>
    <param name="expire-seconds" value="60"/>
    <param name="retry-seconds" value="30"/>
    <param name="ping" value="25"/>
  </gateway>
</include>

FreeSWITCH console error
2014-10-16 00:07:46.105383 [ERR] sofia_reg.c:2312 ips_voice Registration Failed with status Forbidden [403]. failure #1

freeswitch@internal> 2014-10-16 00:07:47.105382 [WARNING] sofia_reg.c:502 ips_voice Failed Registration [403], setting retry to 30 seconds.​

Asterisk console error - debug
<--- SIP read from UDP:10.237.192.68:5060 --->
REGISTER sip:10.237.192.53;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bK87tBt612BpgNp
Max-Forwards: 70
From: <sip:gsvoice01@10.237.192.68>;tag=5jt9mZ5cD93pr
To: <sip:gsvoice01@10.237.192.53>
Call-ID: 747dfaba-54c8-11e4-bc60-81d0d9943fdf
CSeq: 66373227 REGISTER
Contact: <sip:10.237.192.68>
Expires: 0
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141008T204520Z~63734bcde0~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 10.237.192.68:5060 (no NAT)
Sending to 10.237.192.68:5060 (no NAT)

<--- Transmitting (no NAT) to 10.237.192.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.237.192.68;branch=z9hG4bK87tBt612BpgNp;received=10.237.192.68;rport=5060
From: <sip:gsvoice01@10.237.192.68>;tag=5jt9mZ5cD93pr
To: <sip:gsvoice01@10.237.192.53>;tag=as0b2496ec
Call-ID: 747dfaba-54c8-11e4-bc60-81d0d9943fdf
CSeq: 66373227 REGISTER
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="56dafaed"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '747dfaba-54c8-11e4-bc60-81d0d9943fdf' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.237.192.68:5060 --->
REGISTER sip:10.237.192.53;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bK9gm4U1j68y67H
Max-Forwards: 70
From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K
To: <sip:gsvoice01@10.237.192.68>
Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf
CSeq: 66373229 REGISTER
Contact: <sip:gw+ips_voice@10.237.192.68:5060;transport=udp;gw=ips_voice>
Expires: 60
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141008T204520Z~63734bcde0~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Content-Length: 0

<------------->
--- (13 headers 0 lines) ---
Sending to 10.237.192.68:5060 (no NAT)
Sending to 10.237.192.68:5060 (no NAT)

<--- Transmitting (no NAT) to 10.237.192.68:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.237.192.68;branch=z9hG4bK9gm4U1j68y67H;received=10.237.192.68;rport=5060
From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K
To: <sip:gsvoice01@10.237.192.68>;tag=as5eb1b731
Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf
CSeq: 66373229 REGISTER
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="53a1bd9f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'aac7e260-54c9-11e4-bc61-81d0d9943fdf' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:10.237.192.68:5060 --->
REGISTER sip:10.237.192.53;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bKatDXXv3957vtD
Max-Forwards: 70
From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K
To: <sip:gsvoice01@10.237.192.68>
Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf
CSeq: 66373230 REGISTER
Contact: <sip:gw+ips_voice@10.237.192.68:5060;transport=udp;gw=ips_voice>
Expires: 60
User-Agent: FreeSWITCH-mod_sofia/1.5.15b+git~20141008T204520Z~63734bcde0~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, path, replaces
Authorization: Digest username="gsvoice01", realm="asterisk", nonce="53a1bd9f", algorithm=MD5, uri="sip:10.237.192.53;transport=udp", response="4057b65188f937c3a8cdbef65e3d8416"
Content-Length: 0

<------------->
--- (14 headers 0 lines) ---
Sending to 10.237.192.68:5060 (no NAT)

<--- Transmitting (no NAT) to 10.237.192.68:5060 --->
SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 10.237.192.68;branch=z9hG4bKatDXXv3957vtD;received=10.237.192.68;rport=5060
From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K
To: <sip:gsvoice01@10.237.192.68>;tag=as5eb1b731
Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf
CSeq: 66373230 REGISTER
Server: Asterisk PBX 11.12.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
[Oct 16 00:16:26] NOTICE[25991]: chan_sip.c:28059 handle_request_register: Registration from '<sip:gsvoice01@10.237.192.68>' failed for '10.237.192.68:5060' - Wrong password
Scheduling destruction of SIP dialog 'aac7e260-54c9-11e4-bc61-81d0d9943fdf' in 32000 ms (Method: REGISTER)
ip-10-237-192-53*CLI>