[Freeswitch-users] Asterisk 11.20 FreeSWITCH version: 1.5.15b
Brian West
brian at freeswitch.org
Thu Oct 16 09:02:31 MSD 2014
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 at gmail.com> wrote:
>
>
>> 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>
>>>
>>> _________________________________________________________________________
>>> 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
>>>
>>>
>>>
>>>
>>> FreeSWITCH-users mailing list
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>>> http://www.freeswitch.org
>>
>> _________________________________________________________________________
>> 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
>>
>>
>>
>>
>> FreeSWITCH-users mailing list
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>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
>> http://www.freeswitch.org
>
>
>
> --
> Carlos
> http://caruizdiaz.com
> _________________________________________________________________________
> 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
>
>
>
>
> FreeSWITCH-users mailing list
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> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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