Try removing realm from gateway on FS side.

Try an password that do not contain @ character.

You can generate the digest of your Authorization header and compare with your sip capture:
 http://www.kapejod.org/en/2013/02/reversing-sip-digest-authentication-in-javascript-node-js/

Atenciosamente / Best regards,

Fernando da Silva Santos
CEO - NextBilling IP Solutions
$ 55 (11) 3090-0077
$ 0800 580-9200 

P Antes de imprimir pense em sua responsabilidade e  compromisso com o Meio Ambiente!

Em 16/10/2014, à(s) 02:02, Brian West <brian@freeswitch.org> escreveu:

Again 403 == bad password, period end of story.

Anything beyond this is probably environmental!

Sent from my iPhone

On Oct 15, 2014, at 11:53 PM, Carlos Ruiz Díaz <carlos.ruizdiaz@gmail.com> wrote:


On Wed, Oct 15, 2014 at 11:28 PM, Brian West <brian@freeswitch.org> wrote:
403 == bad password

Not necessarily.

Since you have control over the credentials in the Asterisk box, try switching passwords just to discard this variable. Try "1234" for example.

Also, remove the realm parameter and let it decide what value to use, or specify "asterisk". Finally, add "<param name="register" value="true/>".

Maybe, you could also add "insecure=very" on the Asterisk side.
  
Regards,
Carlos


Sent from my iPhone

On Oct 15, 2014, at 7:19 PM, Chris Allison <chris.allison@ipscape.asia> wrote:

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 --->
Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bK87tBt612BpgNp
Max-Forwards: 70
From: <sip:gsvoice01@10.237.192.68>;tag=5jt9mZ5cD93pr
Call-ID: 747dfaba-54c8-11e4-bc60-81d0d9943fdf
CSeq: 66373227 REGISTER
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 --->
Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bK9gm4U1j68y67H
Max-Forwards: 70
From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K
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 --->
Via: SIP/2.0/UDP 10.237.192.68;rport;branch=z9hG4bKatDXXv3957vtD
Max-Forwards: 70
From: <sip:gsvoice01@10.237.192.68>;tag=6UK2ptpgajt9K
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>

_________________________________________________________________________
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http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
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FreeSWITCH-powered IP PBX: The CudaTel Communication Server
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FreeSWITCH-users mailing list
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_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

FreeSWITCH-users mailing list
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--
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
consulting@freeswitch.org
http://www.freeswitchsolutions.com

Official FreeSWITCH Sites
http://www.freeswitch.org
http://confluence.freeswitch.org
http://www.cluecon.com

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
http://www.cudatel.com

FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org