[Freeswitch-users] Asterisk 11.20 FreeSWITCH version: 1.5.15b
Carlos Ruiz Díaz
carlos.ruizdiaz at gmail.com
Thu Oct 16 08:53:14 MSD 2014
On Wed, Oct 15, 2014 at 11:28 PM, Brian West <brian at 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 at 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 at 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 at 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 at 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 at 10.237.192.68>;tag=5jt9mZ5cD93pr
> To: <sip:gsvoice01 at 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 at 10.237.192.68>;tag=5jt9mZ5cD93pr
> To: <sip:gsvoice01 at 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 at 10.237.192.68>;tag=6UK2ptpgajt9K
> To: <sip:gsvoice01 at 10.237.192.68>
> Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf
> CSeq: 66373229 REGISTER
> Contact: <sip:gw+ips_voice at 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 at 10.237.192.68>;tag=6UK2ptpgajt9K
> To: <sip:gsvoice01 at 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 at 10.237.192.68>;tag=6UK2ptpgajt9K
> To: <sip:gsvoice01 at 10.237.192.68>
> Call-ID: aac7e260-54c9-11e4-bc61-81d0d9943fdf
> CSeq: 66373230 REGISTER
> Contact: <sip:gw+ips_voice at 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 at 10.237.192.68>;tag=6UK2ptpgajt9K
> To: <sip:gsvoice01 at 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 at 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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>
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>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
> http://www.freeswitch.org
> http://confluence.freeswitch.org
> http://www.cluecon.com
>
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--
Carlos
http://caruizdiaz.com
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