[Blink] sip2sip.info unreachable via TLS connection

[Michael Nagie]

Thank you for your reply

I do use DNS lookups:

Use DNS SRV: yes	DNS SRV Auto Prefix: yes

DNS SRV:
Whether to use DNS SRV lookup for Proxy and
Outbound Proxy.

DNS SRV Auto Prefix:
If enabled, the phone will automatically prepend the
Proxy or Outbound Proxy name with _sip._udp when
performing a DNS SRV lookup on that name

if you don't see any misconfigurations either I'll try restoring factory 
defults

On 19-12-16 15:40:13, Adrian Georgescu wrote:
> To find out the ports used for SIP TCP and TLS fro a given domain your device must perform the following DNS lookups:
> 
> dig NAPTR naptr sip2sip.info
> sip2sip.info.		3599	IN	NAPTR	15 100 "s" "SIPS+D2T" "" _sips._tcp.sip2sip.info.
> 
> 
> dig SRV _sips._tcp.sip2sip.info.
> _sips._tcp.sip2sip.info. 299	IN	SRV	100 10 443 proxy.sipthor.net.
> 
> The port used for TLS at this moment is 443 but it may change at any time.
> 
> 
> Regards,
> Adrian
> 
> 
> > On 16 Dec 2019, at 15:27, Michael Nagie <promike1987 at gmail.com> wrote:
> > 
> > Hello,
> > I need a little help.
> > I've been unable to connect to sip2sip.info via TLS with my Cisco 
> > SPA504G device for a couple of days now.
> > 
> > sip2sip.info status says: all systems operational
> > yet I can't establish a secure connection.
> > 
> > If I choose TCP transport protocol then it can connect otherwise with 
> > TLS it says 'Failed - Not Reachable'
> > 
> > Here's my configuration, it's pretty basic, I didn't change much:
> > 
> > General
> > Line Enable:  yes
> > 
> > Share Line Appearance
> > Share Ext: private    		Shared User ID:    	
> > Subscription Expires: 3600	Restrict MWI: no   	
> > Monitor User ID:        	SCA Unseize Delay: 0
> > 
> > NAT Settings
> > NAT Mapping Enable: no  	NAT Keep Alive Enable:  no  	
> > NAT Keep Alive Msg: $NOTIFY     NAT Keep Alive Dest: $PROXY
> > 
> > Network Settings
> > SIP TOS/DiffServ Value: 0x68    SIP CoS Value: 3
> > RTP TOS/DiffServ Value: 0xb8    RTP CoS Value: 6
> > Network Jitter Level: high      Jitter Buffer Adjustment: up and down   	
> > 
> > SIP Settings
> > SIP Transport: TLS  		SIP Port: 5060
> > SIP 100REL Enable: no  		EXT SIP Port:    	
> > Auth Resync-Reboot: yes		SIP Proxy-Require:    	
> > SIP Remote-Party-ID: no		Referor Bye Delay: 4
> > Refer-To Target Contact: no	Referee Bye Delay: 0
> > SIP Debug Option: none		Refer Target Bye Delay: 0
> > Sticky 183: no   		Auth INVITE: no
> > Ntfy Refer On 1xx-To-Inv: yes	Use Anonymous With RPID: yes	
> > Set G729 annexb: none  		Voice Quality Report Address:    
> > User Equal Phone: no    			
> > 
> > Call Feature Settings
> > Blind Attn-Xfer Enable: no	MOH Server:    	
> > Message Waiting: no   		Auth Page: no
> > Default Ring: 1    		Auth Page Realm:    	
> > Conference Bridge URL:          Auth Page Password:    	
> > Mailbox ID:    	                Voice Mail Server:    	
> > Voice Mail Subscribe Interval:  86400   State Agent:    	
> > CFWD Notify Serv: no   		CFWD Notifier:    	
> > User ID with Domain: no		Broadsoft ACD: no    	
> > Auto Ans Page On Active Call: yes Feature Key Sync: no
> > HuaWei SBC: yes  		Call Park Monitor Enable: yes	
> > Enable Broadsoft Hoteling: no	Hoteling Sbscrpton Expirs:3600
> > 
> > Proxy and Registration
> > Proxy: sip2sip.info
> > Outbound Proxy:    	
> > Alternate Proxy:    	
> > Alternate Outbound Proxy:0
> > Use Outbound Proxy: no	        Use OB Proxy In Dialog: yes
> > Register: yes          	        Make Call Without Reg: no 	
> > Register Expires: 3600          Ans Call Without Reg: no   	
> > Use DNS SRV: yes   		DNS SRV Auto Prefix: yes    	
> > Proxy Fallback Intvl: 3600      Proxy Redundancy Method:normal   
> > Dual Registration: no  		Auto Register When Failover:no 	
> > 
> > Subscriber Information
> > Display Name: My Name           User ID: user_id
> > Password: ****                  Use Auth ID: no
> > Auth ID: user_id                Reversed Auth Realm:    	
> > Mini Certificate:    	
> > SRTP Private Key:    	
> > Resident Online Number:         SIP URI:    	
> > 
> > Audio Configuration
> > Preferred Codec:  G722  	Use Pref Codec Only: no   	
> > Second Preferred Codec: Unspecified Third Prfrrd Codc:Unspcfd    
> > G711u Enable: yes      		G711a Enable: yes
> > G729a Enable: yes    		G722 Enable: yes    	
> > G726-16 Enable: yes    		G726-24 Enable: yes    	
> > G726-32 Enable: yes    		G726-40 Enable: yes    	
> > Release Unused Codec: yes    	DTMF Process AVT: yes
> > Silence Supp Enable: yes	DTMF Tx Method: Auto
> > DTMF Tx Volume for AVT Packet:0 DTMF AVT Packet Interval:0
> > Use Remote Pref Codec: no	Codec Negotiation: Default   	
> > Rx Payload In 18x Media Session: Use Local SDP   			
> > Dial Plan
> > Dial Plan:    	
> > (*xx|[3469]11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)
> > Caller ID Map:    	
> > Enable IP Dialing: yes   		Emergency Number:    
> > 
> > 
> > -- 
> > Best Regards,
> > 
> > Michael Nagie
> > e: promike1987 at gmail.com
> > _______________________________________________
> > Blink mailing list
> > Blink at lists.ag-projects.com
> > https://lists.ag-projects.com/mailman/listinfo/blink
> > 
> 
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