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 <[email protected]> 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: [email protected] > > _______________________________________________ > > Blink mailing list > > [email protected] > > https://lists.ag-projects.com/mailman/listinfo/blink > > > > _______________________________________________ > Blink mailing list > [email protected] > https://lists.ag-projects.com/mailman/listinfo/blink _______________________________________________ Blink mailing list [email protected] https://lists.ag-projects.com/mailman/listinfo/blink
