Asterisk 11.20 FreeSWITCH version: 1.5.15b
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 ---> 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@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) 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 ---> 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@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) 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 ---> 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@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>
403 == bad password 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 ---> 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@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) 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 ---> 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@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) 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 ---> 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@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>
_________________________________________________________________________ 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
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 ---> 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@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) 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 ---> 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@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) 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 ---> 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@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>
_________________________________________________________________________ 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
-- Carlos http://caruizdiaz.com
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 ---> 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@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) 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 ---> 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@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) 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 ---> 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@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>
_________________________________________________________________________ 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
-- Carlos http://caruizdiaz.com _________________________________________________________________________ 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
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-jav... Atenciosamente / Best regards, Fernando da Silva Santos CEO - NextBilling IP Solutions $ 55 (11) 3090-0077 $ 0800 580-9200 www.nextbilling.com.br 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 ---> 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@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) 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 ---> 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@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) 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 ---> 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@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>
_________________________________________________________________________ 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
-- Carlos http://caruizdiaz.com _________________________________________________________________________ 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
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 $ 0800 580-9200 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@gmail.com> escreveu:
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 ---> 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@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) 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 ---> 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@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) 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 ---> 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@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>
_________________________________________________________________________ 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
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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-- Carlos http://caruizdiaz.com _________________________________________________________________________ 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
Chris, try: [gs-sbc1] username=gsvoice01 type=friend insecure=port,invite secret=ipscape@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@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@gmail.com> escreveu:
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 ---> 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@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) 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 ---> 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@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) 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 ---> 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@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>
_________________________________________________________________________ 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
-- Carlos http://caruizdiaz.com _________________________________________________________________________ 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
-- Ítalo Rossi
participants (5)
-
Brian West
-
Carlos Ruiz Díaz
-
Chris Allison
-
Fernando - NextBilling IP Solutions
-
Ítalo Rossi