[Public/Private IP for Freeswitch in cluster mode]

Did you try setting sip-ip to your private ip and ext-sip-ip to your public ip? and the same for rtp? I have a private IPs on my servers and proxy with public IP and have no issues. Right now I have public IP in both sip-ip and ext-sip-ip since I have no NAT and direct protocol forwarding, but before with NAT I think I had mix of both in those.

freesiwtch won't listen on sip interface until you load sofia into it. It's done when ecallmgr connects to fs.

[4.3.22 update and call forwarding issues]

Barnaby,

Could you post your INVITES? Are you on 4.3.22?

 

Arek

[4.3.22 update and call forwarding issues]

Sure, here is before:
 

INVITE sip:forwarded_to_number@xxx SIP/2.0
Via: SIP/2.0/UDP XXX:11000;rport;branch=z9hG4bK89av9KNXj176g
Max-Forwards: 70
From: "Caller Name" <sip:caller_number@xxx>;tag=yp3tSBpp9Xrtg
To: <sip:forwarded_to_number@xxxx>
Call-ID: 011c88ba-3ef8-11e9-b763-c17ba6c41aeb
CSeq: 1324733 INVITE
Contact: <sip:mod_sofia@xxx:11000>
User-Agent: 2600hz
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248
X-FS-Support: update_display,send_info
Remote-Party-ID: "Caller Name" <sip:caller_number@xxx>;party=calling;screen=yes;privacy=off

and here is after update:

INVITE sip:forwarded_to_number@xxx SIP/2.0
Via: SIP/2.0/UDP xxx:11000;rport;branch=z9hG4bKgXQFF73QH5y8m
Max-Forwards: 70
From: "caller name" <sip:caller_number@xxxx>;tag=N7eKFHF7QXa4a
To: <sip:forwarded_to_number@xxx>
Call-ID: a4c6701a-3edb-11e9-9c45-c17ba6c41aeb
CSeq: 1318642 INVITE
Contact: <sip:mod_sofia@xxx:11000>
User-Agent: 2600hz
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 248
X-FS-Support: update_display,send_info
P-Asserted-Identity: "Account Name" <sip:account_number@kazoo.realm>

 

[4.3.22 update and call forwarding issues]

Hi everyone,

4.3.22 update adds P-Asserted-Identity to INVITE and when you call forward with "Keep Original Caller ID" option checked it will still show account caller Id instead of original caller ID.
And that's because many carriers will use P-Asserted-Identity as legit caller id information and use it instead of FROM header. 

I'm wondering if carriers I use should not be using it to set caller id or Kazoo should not be sending it this way. 

 

Arek

[Advanced e911 - granular based on room or building]

On 10/5/2017 at 10:12 PM, Darren Schreiber said:

How does that accomplish telling the PSAP what room or floor the person is on?

Darren,

You are right, it wont deliver address. It will route to correct PSAP and deliver subscriber name only. That is I guess more for mobile devices.

There is another option: 

Quote

Bandwidth provides a service where your subscribers can call 911 without using phone numbers. Instead of the typical ANI, you register a SIP URI username for the subscriber, along with their name and location, in our databases. When that caller needs help, you send the call to us with the SIP URI instead of the ANI in one of the caller-identifying SIP headers (See section "Supported Privacy Types). Upon receipt of the SIP INVITE from that SIP URI, Bandwidth will send the registered address for that caller to the PSAP. In addition, since most PSAPs are unable to make direct connections to SIP proxies, Bandwidth will temporarily reserve a number from our own PSTN pool to support PSAP callback. If the PSAP needs to call your subscriber back, Bandwidth will bridge the call from the PSAP to your subscriber's SIP UA.

Those are just options and I'm thinking out loud. You could register username with room number. 

Arek

[Advanced e911 - granular based on room or building]

Hi,

You can try with Bandwidth. They support sip header: X-Geolocation:<geo: 39.752913, -104.996080;timestamp=20161213164900> that will route call to correct PSAP based on geolocation. If kazoo supports custom headers maybe that can be added to specific device and then transmitted with 911 call.

From documentation:

Quote

The caller will be routed to the PSAP matching the lat-lng in the geolocation header. If the latlng is invalid or the timestamp older than 60 minutes, the call will go to the emergency call center.

Arek