From sj at d4.dk Tue Dec 3 09:17:03 2019 From: sj at d4.dk (=?iso-8859-1?Q?Stephan_J=F6nsson?=) Date: Tue, 3 Dec 2019 08:17:03 +0000 Subject: [Blink] Windows 10 Callto / TEL protocol Message-ID: Hi, I have begun using the software and I wanted to integrate it with windows so as when i use the click call on phone numbers online it will pass it to Blink as it does with programs like LYNC and skype. I been looking into it and found that you can edit the registration database of win 10 but the only thing I can't find is if it is possible to parse a number via the exe file. Is this possible and would you like to see the code snip I have working with other programs ? Venlig hilsen | Best regards Stephan J?nsson Teamleader Technical department sj at d4.dk [D4] D4 Enterprise Solutions ApS H?gevej 10 | DK-3400 Hiller?d +45 45 90 31 50 | www.d4infonet.dk Vi er certificeret efter ISO 27001 & 9001 F?lg os p? LinkedIn L?s vores persondatapolitik her -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 2378 bytes Desc: image001.png URL: From promike1987 at gmail.com Mon Dec 16 19:27:31 2019 From: promike1987 at gmail.com (Michael Nagie) Date: Mon, 16 Dec 2019 19:27:31 +0100 Subject: [Blink] sip2sip.info unreachable via TLS connection Message-ID: <20191216182731.ndmsi4ojyd26fmha@arch> 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 From ag at ag-projects.com Mon Dec 16 19:40:13 2019 From: ag at ag-projects.com (Adrian Georgescu) Date: Mon, 16 Dec 2019 15:40:13 -0300 Subject: [Blink] sip2sip.info unreachable via TLS connection In-Reply-To: <20191216182731.ndmsi4ojyd26fmha@arch> References: <20191216182731.ndmsi4ojyd26fmha@arch> Message-ID: 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 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 > From promike1987 at gmail.com Tue Dec 17 11:07:16 2019 From: promike1987 at gmail.com (Michael Nagie) Date: Tue, 17 Dec 2019 11:07:16 +0100 Subject: [Blink] sip2sip.info unreachable via TLS connection In-Reply-To: References: <20191216182731.ndmsi4ojyd26fmha@arch> Message-ID: <20191217100716.eu4vlhpddkoobh6a@arch> 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 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 > > > > _______________________________________________ > Blink mailing list > Blink at lists.ag-projects.com > https://lists.ag-projects.com/mailman/listinfo/blink