SIP Interoperability
Posted by on 06 May 2011 08:39 AM

Numbers format

  • When calling out, the called phone number should be formatted as "1" followed by 10 digits, "011" plus digits for international calls, etc., as is standard US practice.
  • When calling Emergency Number (911), the To Number of the call should be formatted as "1911".
  • Alternatively, full E.164 number formatting is also supported ("+1" and 10 digits US number, "+44" and UK number, etc.).
    Note: this is the only supported format for OCS interoperability.
  • Inbound numbers (DID or toll-free) will be presented to your system as "1" plus 10 digits in the Request-URI, except for OCS interconnection where the numbers will be presented as "+1" followed by 10 digits (standard Microsoft interop).
  • Your originating number (Caller-ID) must be formatted as "1" followed by 10 digits and must be present at least in the username field of the SIP From header. Calls without a valid From calling number will be rejected (because they cannot be billed).
  • Calls from a toll-free number to a toll-free number (i.e. calls where the called and calling numbers are both toll-free numbers) will be rejected (because they cannot be billed).
  • For outbound calls, the called number must be present in the username field of the SIP Request URI, and the username field of the To header. (This is standard practice.)
  • Use P-Asserted-Identity (PAI) and Privacy headers (RFC3323-3325) to specify or suppress caller information.
  • Specifically, use "Privacy: id" to request that the caller-id not be transmitted at the far-end called interface (the calling number will still transit over SIP and SS7). Please note that using this option for business-to-customer calls in the US is not recommended.

SIP headers

  • Request-URI, From, To, and Contact headers should not contain any Display information (preferred configuration if your platform allows it).
  • Only SIP URIs are supported in the Request-URI, From, To, and Contact headers (in other words we do not currently support "tel:" URIs).

SIP Notes

  • Codecs available are G.711 and G.729a, RFC2833 DTMF, and T.38 fax.
  • SoTel SIP Service does not support PRACK. (Please disable RFC3262 support in your device.)
  • SoTel SIP Service does not support Session Timers. (Please disable RFC4028 support in your device.)
  • Support for REFER is limited; properly securing and billing REFER messages present challenges. (Please disable RFC3515 support in your device.)
  • Re-INVITE messages should be limited to changes between voice and T.38 mode; we cannot guarantee that other re-INVITEs will be properly handled. (In other words we expect your IP-PBX to pin the media and handle media changes, such as transfers, hunting, etc., transparently to us.) Using late negotiation in re-INVITEs currently leads to calls being dropped.
  • IMPORTANT: SoTel Systems LLC does not support modem, ATM nor any credit card transaction traffic over our network.
  • IMPORTANT: We highly recommend turning off SIP ALG on your firewall/routers.  This setting may cause numerous issues including dropped calls and audio.

Please note that prior to acknowledging an order for Local Number Portability or Toll-free port, SoTel will verify that the system(s) specified in your instructions actually is (are) ready to receive traffic for the numbers you are requesting. This avoids the loss of calls around the time of the port, and the risks of missing a port date because of interoperability tests by the porting carrier.

(8 vote(s))
This article was helpful
This article was not helpful

Help Desk Software by Kayako Fusion
ERROR: This domain name does not match domain registered in the license key file (, allowed domains:, please change the product path to match the domain under Admin CP > Settings > General Settings
This product will not work properly unless untill that value is changed.

For more information please contact Kayako support at