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

</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 --->

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)

<--- 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 --->

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)

<--- 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 --->

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>