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Phriction Welcome SIP2SIP Configurations Snom 370

Snom 370
Updated 2,886 Days AgoPublic
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Goal is to create a reliable and secure configuration for inbound calls, where the account is shared with another SIP client Groundwire.

Login Information

Identity activeOn
Displayname<empty>
Username2233XXXXXX
Registrarsip2sip.info:443;transport=tls
Outbound Proxyproxy.sipthor.net:443;transport=tls
Failover IdentityNone
Authentication Username<empty>
Mailbox<empty>

SIP Identity Settings

Send hold as inactiveOff
Alert Info URL<empty>
Dial-Plan String<empty>
ENUM SupportOff
Countrycode31
Areacode10
Proxy Require<empty>
Additional supported headers<empty>
Q-Value1.0
Proposed Expiry43200
Auto AnswerOff
Long SIP-Contact (RFC3840)On
Support broken RegistrarOff
Shared LineOn
Publish Presence on bootupOff
DTMF via SIP INFOOn
Send display name on INVITEOff
Extension Monitoring Call Pickup ListOff
Extension Monitoring Call Pickup List URI<empty>
Contact ListOff
Contact List URI<empty>
Server Type SupportDefault
Remove all bindings on unregisterOff
Subscription Expiry (s)43200
Failed Subscription Retry Time (s)30
Enable hook flashOff

Other possible values for "Server Type Support" are:

  • CCM
  • Asterisk
  • Broadsoft
  • MetaSwitch
  • PBXnSIP
  • Sylantro
  • Telepo
  • Nortel
  • Teles
  • bria

NAT Identity Settings

Offer ICEOn
STUN server (IP-addr:port)<empty>
STUN interval (seconds)<empty>
Keepalive interval (seconds)<empty>

RTP Identity Settings

Codec 1G.722
Codec 2G.711a
Codec 3G.711u
Codec 4G.726-32
Codec 5GSM FR
Codec 6G.729AB
Codec 7G.723.1
Packet Size20 ms
Full SDP AnswerOn
Summetrical RTPOff
RTP EncryptionOn
Dynamic G.726 payloadOn
G.726 Byte OrderRFC3551
SRTP Auth-tagAES-32
RTP/SAVPOff
Media Transport OfferUDP
Media Transport Offer SetupActive
Last Author
tijmen
Last Edited
Dec 27 2016, 12:06 PM

Event Timeline

tijmen created this document.Dec 27 2016, 12:03 PM
tijmen edited the content of this document. (Show Details)
tijmen edited the content of this document. (Show Details)Dec 27 2016, 12:05 PM
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