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Enhanced 911 for OpenSIP

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alexxst
Occasional Contributor

Enhanced 911 for OpenSIP

Hello,

I'm trying to configure my VVX/Trio phones working in Generic SIP mode (ver: 6.4.1.2280) to send the switch MAC address & Port ID to server in SIP REGISTER request ('X-switch-info' header).

I've set below, but see nothing in REGISTER message.

feature.E911.enabled="1"
locInfo.source="LLDP" 

 

Expected to a similar behavior like with Yealink phones, am I missing something?

SteffenBaierUK_0-1641886882568.png

 

Message 1 of 4
1 ACCEPTED SOLUTION

Accepted Solutions
SteffenBaierUK
Polycom Employee & Community Manager

Re: Enhanced 911 for OpenSIP

Hello @alexxst ,

 

I have since then looked into this and it seems you only need to enable the following for the phone to send the x-switch-info in its register message:

 

<web voIpProt.SIP.header.switchInfo.enable="1"/>

 

I am getting a Doc Bug raised as I believe the Admin guide states this should be enabled by default.

 

Enabling this will send the register message with the Switch Port. To my own personal knowledge, this was added for our Partner Zoom. 

 

Any issues, if you are not using Zoom, for the supported use case please follow this up as explained via our support organisation.

 

0118200700|sip  |0|00|>>> Data Send to 172.21.177.17:5060
0118200700|sip  |0|00|    REGISTER sip:172.21.177.17 SIP/2.0
0118200700|sip  |0|00|    Via: SIP/2.0/UDP 10.252.149.59;branch=z9hG4bK9ba9ab27482E408A
0118200700|sip  |0|00|    From: "3034" <sip:3034@172.21.177.17>;tag=94836B96-4838D646
0118200700|sip  |0|00|    To: <sip:3034@172.21.177.17>
0118200700|sip  |0|00|    CSeq: 61 REGISTER
0118200700|sip  |0|00|    Call-ID: c9464b584d17bbebf611b62d4ff2b007
0118200700|sip  |0|00|    Contact: <sip:3034@10.252.149.59>;methods="INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER"
0118200700|sip  |0|00|    User-Agent: PolyCCX-CCX_400-UA/7.2.0.12360
0118200700|sip  |0|00|    Accept-Language: en
0118200700|sip  |0|00|    X-switch-info: mac=50:06:ab:92:4b:00,port=Fa0/8
0118200700|sip  |0|00|    Authorization: Digest username="3034", realm="asterisk", nonce="3e8748d3", uri="sip:172.21.177.17", response="0925a48212d7f935c4cb1a9f6482b5c4", algorithm=MD5
0118200700|sip  |0|00|    Max-Forwards: 70
0118200700|sip  |0|00|    Expires: 3600
0118200700|sip  |0|00|    Content-Length: 0
0118200700|sip  |0|00|    
0118200700|sip  |0|00|<<< End of data send

 

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

 

----------------

If official support is required please check how to phone or open a case here

----------------
The title Poly Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

----------------


⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓SIGNATURE ⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓
Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
Please also ensure you always check the VoIP , Video Endpoint , Microsoft Voice , PSTN or other FAQ's in the different sections

View solution in original post

Message 3 of 4
3 REPLIES 3
SteffenBaierUK
Polycom Employee & Community Manager

Re: Enhanced 911 for OpenSIP

Hello @alexxst ,

 

Welcome to the Poly Community. Pictures inside the topic need to be approved by a moderator before they appear.

 

From our latest 6.4.0 Admin Guide >here<

 

  • feature.E911.enabled
    0 (default) - Disable the E.911 feature.

    1 - Enable the E.911 feature.

    The INVITE sent for emergency calls from the phone includes the geolocation header defined in RFC 6442 and PIDF presence element as specified in RFC3863 with a GEOPRIV location object specified in RFC4119 for in Open SIP environments.

I would only expect an INVITE to include this.

 

If you still believe we are violating an RFC or there is a bug please go ahead and open a support ticket in your region and/or if the phone is out of warranty our team should explain to you who to contact to pay for PPI/PayPerIncident support.

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

 

----------------

If official support is required please check how to phone or open a case here

----------------
The title Poly Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

----------------


⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓SIGNATURE ⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓
Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
Please also ensure you always check the VoIP , Video Endpoint , Microsoft Voice , PSTN or other FAQ's in the different sections
Message 2 of 4
SteffenBaierUK
Polycom Employee & Community Manager

Re: Enhanced 911 for OpenSIP

Hello @alexxst ,

 

I have since then looked into this and it seems you only need to enable the following for the phone to send the x-switch-info in its register message:

 

<web voIpProt.SIP.header.switchInfo.enable="1"/>

 

I am getting a Doc Bug raised as I believe the Admin guide states this should be enabled by default.

 

Enabling this will send the register message with the Switch Port. To my own personal knowledge, this was added for our Partner Zoom. 

 

Any issues, if you are not using Zoom, for the supported use case please follow this up as explained via our support organisation.

 

0118200700|sip  |0|00|>>> Data Send to 172.21.177.17:5060
0118200700|sip  |0|00|    REGISTER sip:172.21.177.17 SIP/2.0
0118200700|sip  |0|00|    Via: SIP/2.0/UDP 10.252.149.59;branch=z9hG4bK9ba9ab27482E408A
0118200700|sip  |0|00|    From: "3034" <sip:3034@172.21.177.17>;tag=94836B96-4838D646
0118200700|sip  |0|00|    To: <sip:3034@172.21.177.17>
0118200700|sip  |0|00|    CSeq: 61 REGISTER
0118200700|sip  |0|00|    Call-ID: c9464b584d17bbebf611b62d4ff2b007
0118200700|sip  |0|00|    Contact: <sip:3034@10.252.149.59>;methods="INVITE,ACK,BYE,CANCEL,OPTIONS,INFO,MESSAGE,SUBSCRIBE,NOTIFY,PRACK,UPDATE,REFER"
0118200700|sip  |0|00|    User-Agent: PolyCCX-CCX_400-UA/7.2.0.12360
0118200700|sip  |0|00|    Accept-Language: en
0118200700|sip  |0|00|    X-switch-info: mac=50:06:ab:92:4b:00,port=Fa0/8
0118200700|sip  |0|00|    Authorization: Digest username="3034", realm="asterisk", nonce="3e8748d3", uri="sip:172.21.177.17", response="0925a48212d7f935c4cb1a9f6482b5c4", algorithm=MD5
0118200700|sip  |0|00|    Max-Forwards: 70
0118200700|sip  |0|00|    Expires: 3600
0118200700|sip  |0|00|    Content-Length: 0
0118200700|sip  |0|00|    
0118200700|sip  |0|00|<<< End of data send

 

 

Please ensure to provide some feedback if this reply has helped you so other users can profit from your experience.

Best Regards

Steffen Baier

 

----------------

If official support is required please check how to phone or open a case here

----------------
The title Poly Employee & Community Manager is a community setting and does not reflect my role. I am just a simple volunteer in the community like everybody else. All posts and words are my own & do not represent the views of Employer.

----------------


⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓SIGNATURE ⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓⇓
Notice: This community forum is not an official Poly support resource, thus responses from Poly employees, partners, and customers alike are best-effort in attempts to share learned knowledge.
Please also ensure you always check the VoIP , Video Endpoint , Microsoft Voice , PSTN or other FAQ's in the different sections
Message 3 of 4
alexxst
Occasional Contributor

Re: Enhanced 911 for OpenSIP

Thanks SteffenBaierUK - that's exactly what I looked for.

Message 4 of 4