If you desire SIP in Avaya, you have to add a SES (SIP Enablement Server) to your Avaya setup. They are cheap. You only have to pay for the box and the maintenance percentage. You don't need to buy user ports or any of that garbage as long as you setup your extensions using Optum, which is a free Avaya feature. The SES maintains a registry and a dial plan. SIP phones attached to SES send media directly to medpros and the SES does a protocol conversion between SIP and H.323 to bridge a connection between the SIP phone and the CLAN cards.
The voicemail issue you describe with the MWI is because Avaya's systems use qsig trunks to connect to voicemail servers. Asterisk is not connected int hat manner, so of course you won't be able to support Avaya MWI's. However, you can deposit a script on your asterisk that would send the standard notifies to the Avaya phones to manipulate the MWI's directly. However, you will need to statically address the phones and keep track of them because you cannot poll an SES server for their SIP URI's. Hell, Avaya won't even give you root on any of their servers. You cant audit the box and you can't poll them unless you pay them money to join their partner program and get their SDK. If you already have Avaya, you should just buy Message Networking or a Mitel voicemail server if you want seamless voicemail with Avaya. However, you should know that using Avaya is probably a bad idea to begin with. Until February 07, the majority Avaya's soft switch products were running on Redhat 9, which was unsupported since 2003. Avaya was only managing a dozen packages and they've always left it up to the customer to know when they need an update, requiring the customer to request a field load. It has to be the worst update model in the industry when it comes to infrastructure monitoring and patching. By using Avaya, you are blindly trusting them to properly maintain a Linux appliance. This is something they are not capable of and you can't even audit them. Avaya is what happens to organizations when they have ignorant telecom infrastructure engineers deciding what products to buy. Avaya focuses sales on those engineers because they k now their products won't pass certification by network, systems, or security engineers. Telecom engineers only look for features and usually get their asses handed to them after they put Avaya VoIP into their infrastructure. -------------------------------------------------- Salvatore Giudice [EMAIL PROTECTED] VoIP Security Training, LLC http://VoIPSecurityTraining.com 848 N. Rainbow Blvd. #1676 Las Vegas, NV 89107 Phone: (617) 959-7625 Fax: (214) 279-2906 -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Jim Houser Sent: Friday, November 30, 2007 9:54 AM To: [EMAIL PROTECTED]; 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: Re: [asterisk-users] Off-Topic: Avaya This is both a hardware and software licensing issue. Avaya offers a SIP server separate from their main VoIP gateway. The core platform uses H.323. Either SIP or H.323 has a license cost per registered device. We have an Avaya S8300 Communications Manager providing H.323 and have this tied to an Asterisk deployment on a Sun Microsystems server. The connection between the two systems are handled by both T1, (PRI using Qsig), and H.323. The BIG issue we have is we cannot light the message waiting light on the Avaya 46XX phones registered to the Avaya server but using Asterisk voice mail. If anyone can help we would pay to solve this. Our Asterisk is 1.2.xx. Thanks. -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Alejandro Cabrera Obed Sent: Friday, November 30, 2007 7:30 AM To: Asterisk Users Mailing List Subject: [asterisk-users] Off-Topic: Avaya Dear all, sorry for my OT but I need to know if Avaya voip server uses SIP or H.323 ??? Anybody can't tell me this...so I'm here for thei reason. Thanks a lot _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users