[Freeswitch-users] Asterisk 11.20 FreeSWITCH version: 1.5.15b
Ítalo Rossi
italorossib at gmail.com
Thu Oct 16 18:12:02 MSD 2014
Chris,
try:
[gs-sbc1]
username=gsvoice01
type=friend
insecure=port,invite
secret=ipscape at 2014
qualify=yes
*host=dynamic*
;192.168.202.60
dtmfmode=rfc2833
disallow=all
canreinvite=no
allow=alaw
On Thu, Oct 16, 2014 at 2:21 AM, Fernando - NextBilling IP Solutions <
freeswitch at nextbilling.com.br> wrote:
> Sorry by the earlier link.
>
>
> This is the correct one: http://www.sieraybould.net/Software/SipMD5Calc/
>
>
>
>
> *Atenciosamente / Best regards,Fernando da Silva Santos*
> *CEO - NextBilling IP Solutions*
> *$ 55 (11) 3090-0077 <55%20%2811%29%203090-0077>*
> *$ 0800 580-9200 <0800%20580-9200> *
> *www.nextbilling.com.br <http://www.nextbilling.com.br>*
>
> *P Antes de imprimir pense em sua responsabilidade e compromisso com
> o Meio Ambiente!*
>
> Em 16/10/2014, à(s) 01:53, Carlos Ruiz Díaz <carlos.ruizdiaz at gmail.com>
> escreveu:
>
>
> 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
>>
>>
>>
>>
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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://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
> FreeSWITCH-users at 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 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
> FreeSWITCH-users at lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
--
Ítalo Rossi
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