E.164 on IP Office?

[Bivman]

Is there a way to get the IP Office to use E.164?

When I add a + sign to the number field, it inserts a 00
in front of the sip invite.

I need the invite to show To: +1xxxxxxxxxx@ot.bandwidth.com

Right now, it only uses 10-digit dialing.

I can get bandwidth to accept 10-digit, but every time we
make a change at bandwidth, they revert to E.164 and we
have to call and have them change back to 10-digit.
If I can get the system to use E.164 it will make things
much easier!

Thanks,

-Jon

 

[intrigrant]

maybe you must enable "Use tel URI

 

[CarGoSki]

Matrix says this in regards to SIP and supported protocol:

Session Initiated Protocol (SIP)

• Rec. E.164 [2] - ITU-T Recommendation E.164: The international public telecommunication numbering plan
• RFC 2833 [7] - RTP Payload for DTMF Digits, Telephony Tones and Telephony Signals
• RFC 3261 [8] - SIP: Session Initiation Protocol
• RFC 3263 [10] - Session Initiation Protocol (SIP): Locating SIP Servers
• RFC 3264 [11] - An Offer/Answer Model with Session Description Protocol (SDP)
• RFC 3323 [14] - A Privacy Mechanism for the Session Initiation Protocol (SIP)
• RFC 3489 [18] - STUN - Simple Traversal of User Datagram Protocol (UDP) Through Network Address Translators (NATs)
• RFC 3824 [24] - Using E.164 numbers with the Session Initiation Protocol (SIP)
• RFC 1889 – RTP
• RFC 1890 - RTP Audio
• RFC 4566 – SDP
• RFC 3265 - Event Notification
• RFC 3515 - SIP Refer
• RFC 3842 - Message Waiting
• RFC 3310 – Authentification
• RFC 2976 – INFO
• RFC 3323 - Privacy for SIP (PAI) and draft-ietf-sip-privacy-04 (RPID)

 

[Bivman]

tel uri only puts the outgoing number in the invite.
doesn't add the @ot.bandwidth.com


sigh..


Thanks,
-Jon

 

[hairlessupportmonkey]

what have you got in your itsp domain name field?

ITSP Domain Name: Default = Blank.
This field is used to enter the domain part of the SIP URI provided by the ITSP. For example, in the SIP URI name@example.com, the domain part of the URI is example.com. For outgoing calls the user part of the SIP URI is determined in a number of ways:

· For the user making the call, the user part of the FROM SIP URI is determined by the settings of the SIP URI channel entry being used to route the call. This will use one of the following:

· a specific name entered in Local URI field of the channel entry.

· or specify using the primary or secondary authentication name set for the line below

· or specify using the SIP Name set for the user making the call (User | SIP | SIP Name).

· For the destination of the call, the user part of the TO SIP URI is determined by the dialing short codes of the form 9N / N"@example.com" where N is the user part of the SIP URI.

 

[Bivman]

I have the domain name in there... ot.bandwidth.com
my shortcode 9N goes to my SIP ARS

xxxxxxxxxxN; Dial 3K1 N"@ot.bandwidth.com"


Which generates this output on the system status trace:


12/11/09 11:08:29 AM-150ms Line = 20, Channel = 2, SIP Message = Response, Call Ref = 195, Direction = To Switch, From = 618xxxxxxx@ot.bandwidth.com, To = 618xxxxxxx@ot.bandwidth.com, Response = 200 Ok
12/11/09 11:08:29 AM-152ms Line = 20, Channel = 2, SIP Message = Ack, Call Ref = 195, Direction = From Switch, From = 618xxxxxxx@ot.bandwidth.com, To = 618xxxxxxx@ot.bandwidth.com
12/11/09 11:08:29 AM-156ms Call Ref = 195, Originator State = Connected, Type = User, Destination State = Connected, Type = Trunk
12/11/09 11:08:29 AM-156ms Call Ref = 195, Answered, Line = 20, Channel = 2

 

[hairlessupportmonkey]

you say you need to show +1xxxxxxxxxxx.....

cant they accept 001? then you can prefex your short code.

to be honest, the IPO doesnt like dealing with +1 - it likes real numbers.

 

[hairlessupportmonkey]

ok - talking out of my ass a little here.

try this:

http://support.avaya.com/css/P8/documents/100020140

 

[hairlessupportmonkey]

you can make your SIP trunks to dial with USER URI information.

you can cheat a little here and make some virtual extns.

i like to call then:

SIP:<your number> / <Extn#> / SIP field add +1<DDI>

make sense?

then use your incoming call route as normal