smaunsell

Hi George, I worked for a distributor of the IP Office in the UK for about 18 mths and have recently left. We were supporting dealers, like yourselves, and reported any problems that we couldn't fix directly to Avaya technical support.

SIP is not planned to be released on the IP Office until v3.2. This will be SIP trunks ie connection to Internet providers or other systems. Support for SIP endpoints ie phones will not be supported until v4.0.

I think v3.2 is planned for release at the end of 2006. And v4.0 will be mid-2007 I think.

Your distributor should be able to give you more details - which one is it? I'm biased of course (but I don't work there anymore), but the one I worked for had a very good reputation for providing the best technical support in the UK for IP Office. There are some distributors that are very bad indeed.

Let me know if there's any more info that you need. I still have contacts at thhis distributor so can find out any info you need.

Cheers

vervestr

As far as I know SIP-trunking will be supported from IP Office rel 4.0 (feb'07).

SIP Endpoints are projected for IP Office 4.1 and beyond...

Cheers, Ronald

ZebrAYA

SIP is with us
anyone has already implemented it?
SIP trunks for example...

ZebrAYA

SIP IS WITH US! - LET'S GET SIPPIN'


After passing the system upgrade to 4.0 version (learning on the fly)
I’m making pilot tests with a SIP provider, known for many people
Voipcheap.co.uk
I found the parameters for sip configuration on its own page
voipcheap.co.uk/en/sipp.html

nonetheless there some other parameters that AVAYA IPO manager requests, and not very clear, maybe they sound new to me:

I also ask if the "Network Configuration" section could affect it!

Remember our IPO’s have some firewall features that few people use. Although default state is “all allowed” at times there can be issues.

I couldn’t specify the option in Firewall options, because TCP code is 6, but UDP is?


Line | SIP Line

·Line Number: Default = Automatically assigned.
By default a value is assigned by the IP Office. This value can be changed but it must be unique.
·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 mysip@itsp.com, the domain part of the URI is itsp.com.
·The user part of the 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 below
·or specify using the SIP Name set for the user making the call (User | SIP | SIP Name).
·ITSP IP Address: Default = 0.0.0.0
This value is provided by the SIP ITSP. This address must not be shared by any other IP, IP DECT or SIP line in the IP Office system configuration.
·Primary Authentication Name: Default = Blank.
This value is provided by the SIP ITSP. Depending on the settings on the Local URI tab associated with the SIP call it may also be used as the user part of the SIP URI.
·If the From field on the Local URI being used for the call is set to Use Authentication Name and the Registration is set to Primary, this value is used as the user part of the SIP URI for calls.
·Primary Authentication Password: Default = Blank.
This value is provided by the SIP ITSP.
·Primary Registration Expiry: Default = 3600 minutes.
This setting defines how often registration with the SIP ITSP is required following any previous registration.
·Secondary Authentication Name: Default = Blank.
This value is provided by the SIP ITSP. Depending on the settings on the Local URI tab associated with the SIP call it may also be used as the user part of the SIP URI.
·If the From field on the Local URI being used for the call is set to Use Authentication Name and the Registration is set to Secondary, this value is used as the user part of the SIP URI for calls.
·Secondary Authentication Password: Default = Blank.
This value is provided by the SIP ITSP.
·Secondary Registration Expiry: Default = 3600 minutes.
This setting defines how often registration with the SIP ITSP is required following any previous registration.
·Registration Required: Default = Off.
If selected, the SIP trunk will register with the ITSP using the value in the ITSP Domain Name field.
·In Service: Default = On.
When this field is not selected, the SIP trunk is unregistered and not available to incoming and outgoing calls.
·Use Tel URI: Default = Off.
Use Tel URI format (for example TEL: +1-425-555-4567) rather than SIP URI format (eg. mysip@itsp.com).
·VoIP Silence Suppression: Default = Off.
When selected, this option will detect periods of silence on any call over the line and will not send any data during those silent periods.
·Out of Band DTMF: Default = Off.
This field is greyed out and cannot be changed. When on, DTMF is sent as a separate signal rather than as part of the encoded voice stream ("In Band"). This is recommended for low bit-rate compression modes such as G.729 and G.723 where DTMF in the voice stream can become distorted.
·Local Tones: Default = On.
This field is greyed out and cannot be changed. When on, call tones are generated by the local IP Office system to which the phone is registered.
·Fax T38: Default = Off.
This field is greyed out and cannot be changed.
·RE-INVITE Supported: Default = Off.
·Voice Packet Size
This is the length of time represented by each VoIP packet in milliseconds. This is automatically defaulted to match the Compression Mode selected. Values are G.711/G.729 = 20ms, G.723=30ms.
·Compression Mode: Default = Automatic
This defines the compression method to be used for this line.
Network Configuration
·Layer 4 Protocol: Default = UDP
This field sets whether the line uses UDP SIP or TCP SIP.
·Use Network Topology Info: Default = LAN1
This field associates the SIP line with the System | LAN1 | Network Topology settings
·Send Port: Default = 5060
This field sets the port to which IP Office send outgoing SIP calls.
·Listen Port: Default = 5060
This field sets the port on which the IP Office listens for incoming SIP calls.

bnecom

Works great once you sort out the settings. Need VCM and SIP trunk licenses.

HTH

Jason Wienert
Central City Communications