[Blink] sip2sip.info unreachable via TLS connection

Adrian Georgescu ag at ag-projects.com
Mon Dec 16 19:40:13 CET 2019


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
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