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We are migrating a CM6 system, with AAM, to 11-digit E.164 dial plan. As part of that we need to convert the existing 5-digit mailboxes to 11-digit. Anyone know of a tool/method to complete that? I have tried experimenting with Provision and ASA tools but nothing so far. I did the same thing...

We are migrating a CM6 system, with AAM, to 11-digit E.164 dial plan. As part of that we need to convert the existing 5-digit mailboxes to 11-digit. Anyone know of a tool/method to complete that? I have tried experimenting with Provision and ASA tools but nothing so far. I did the same thing...

I just implemented a Session Manager R6.1 with CM6.1 for SIP integration with a Rightfax Server. Calls from the PBX to the Rightfax are fine but calls out of the Right fax are not. The call comes out of the Session Manager and htis the PBX but instead of routing out the PSTN via the ARS table it...

Does anyone happen to know if Expanded Meet Me Conferencing is supported in CM 6.1? We did an upgrade tonight from CM3 to CM6, and re-IP'd all hardware and now the SIP signaling group is stuck in "Far-End Bypass". I know it is end of support with Avaya but I can't find any documents that say...

I am having an issue with outcalling, notify me, from an Avaya Aura Messaging VM. It works fine to local extensions and to external cell phones but my problem is to a pager system. This is an internal pager system so there is a hunt group that users dial, then enter pager number, then enter VM...

I have a customer wanting to integrate an Avaya CM6 w/Session Manager to the Ascom i75 SIP handsets. I found an Avaya integration note but it mostly just tells how to program the Ascom and gives no specifics on the Avaya side, imagine that. Anyone have any documents with more specifics on this?

I have an MM R5.2 connected via DIRECT SIP trunk to the PBX, CM R6. All is working fine except "Call Me" to an outside number. I see the call come into the PBX but it is saying "User not found (adjunct origination)". I see the called number come in as"9xxx.xxxx@sipdomain.com". Is the switch not...

I have an S8300 with an embedded SES server that integrates to a MM w/exchange. WE are having some calls fail to particular numbers, all the time, while others work fine. I am trying to use the SES Trace Logger files but apparently not understanding what information they want in the filters...

I am having a problem with a SpectraLink OAI Gateway. I'm getting a lost heartbeat error. If the error isn't cleared in time the box will lock up and I can't access it. I have to power it down. Then it will work again. I have an Avaya 8800 switch. I am getting pages from a Rauland Responder 4...

I have an operator who cannot transfer directly to a voicemail box or enter a caller into a conference bridge. The Operator is using OSPC in Road Warrior mode. I am thinking that the DTMF isn't passing. For instance she tries to transfer a user into a specific voicemail box, once connected to...

All, I am trying to implement a SIP Trunk between an Avaya Communicaiton Manager System R5.2 and an IPCM SIP server. Calls come in the Avaya and route across the SIP trunk to the IPCM just fine. However, when a user tries to call outbound from the system on the other side of the IPCM I don't...

Does anyone know if I have to have an SES Server to create SIP Trunks between an IP Office and a Communication Manager Server. I am thinking I do but have not found any documentation as of yet either way. Thanks

A colleague has just mirgrated a G3si CM1.3 to an S8800 CM5.2. The customer insists they were getting Caller ID WITH name before the migration. At this time they are only getting the number. We have verified that the programming of the trunk and incoming call route is identical to the...

I am trying to load One-X Attendant on a PC running Windows 7 Professional. Specifically when I try to load WebLM I get the error "Summary Not available" "This may be a result of an invalid summary type request. e.g. requesting an install summary when an uninstall has occurred". Has anyone ran...

I just installed OSPC, for the first time, on a customer PC. The application seems to function ok, with two problems. The busy lamp field won't show when users are busy and the name is not showing up on the buttons either. I can click on the BLF for a user and it calls them ok just doesn't show...

I have several users complaining that when receiving a call to their IP Agent they click to answer the call but it is automatically being put on hold and then dropped. The agent sees the call on hold but can't retrieve it. These were working stations and then "all the sudden" the customer says...

Does anyone know where I can find a white paper on configuring an IP Trunk from an Avaya S8730 to a Cisco Call Manager? I have a customer with a new site that already had a Cisco switch that they would like to transfer calls from their Avaya S8730, via UDP and IP Trunk, to the Cisco Switch. Thanks

The S8300/G700 i just installed will not play music on hold to callers placed on hold via 4621SW. This includes internal and external calls. The music works fine analog/digital to analog/digital but will not work IP to IP nor will it work if one of the stations is IP and the other is...

I have just installed an S8300/G700 system using 4621SW telephones using CM 5.1.1. All firmware is up to date for system and phones. To Access paging you dial a three digit extension answered by a UPAM, wired as a station answer. Behind the UPAM is a nine zone paging module which expects to hear...

I have a customer experiencing and issue when calls are transferred to an extension and then covers to voicemail. ex.. I call customer, customer transfers me to extension that has coverage path to voicemail, customer sees cover on her display showing that the call is covering to voicemail but...