[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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