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We have connected 2 magix systems using INA cards at both ends channels 1-12 DATA 13-23 Voice 24 D-channel we are doing centralized voicemail can udp calls to and from both sites but are blocked from calling through to the other side be it copper 70 or Flexpath (891) FROM SITE A ars table 2...

We have a call center setup where afterhours and on weekends it calls (route to xxxx ..an x-ported extension with cover remote to offsite ...it works fine with Land lines but its BLOCKED when trying to use a cell numbers in the cover remote field ...has anyone seen something like this ????? I...

we are installing a t1 card into a system setup as follows exsisting Setup: slot 1 slot 2 slot 3 slot 4 slot 5 308 partner 308 400 308 messaging procesor 6 port converted Setup: slot 1 slot 2 slot 3...

We have just upgraded our system to an S8300 rel 11 from a prologix rel 10 ....we are now using EAS with Agent Id rather then bcms login id's we can list bcms agent xxx and see the data from the switch but can not get any data from BCMS vu server if we log on as client on the desktop or right...

We are switching from a release 10 Prologix to an S8300 release 11 we had been using c-lan connection for BCMS Server and ASA from a workstation using node-names ip , ip-int ip-services ...data module (ethernet) my question is : since we do not have a c-lan or an ipsi card card in the s8300...

WE ARE HAVING AN ISSUE PUTTING A (AVAYA)LKA10 HEADSET ON A 6424D+ PHONE OFF OF IP OFFICE REL 1.3.37 ABLE TO RECEIVE CALLS ON HEADSET ....BUT NOT ABLE TO BREAK DIALTONE FROM DIALPAD ON CORDLESS HEADSET .....SO YOU CAN ANSWER CALLS BUT NOT TRANSFER THEM WE HAVE TRIED 2 DIFFERENT...

Is it possible to have an extension (voicemail) call out to more then one number ...using source number within the user as this case is its a SERVICE afterhours mailbox source number P*601 Shortcode *601 tel # 1212xxxxxx,,,,,100## (BEEPER) grp 0 feature dial works...

We have an issue where the Audix picks up and you hear 3 - 4 seconds of silence ....NO Alarms on Ip600 , Audix busy board a12 test board a12 (Pass) release board a12 test board a12 long clear (Pass - resolved) sa trans and did a reset system 4 Audix still picks up and silence...

I have installed a NEC Electra 48 with a voicemail Elite in slot 07 for a friend that wanted to reuse it from another site ( I mostly work on Avaya ) I worked on them awhile ago but can get around logging into x100 or x101 ...also using cosession for the voicemail *my question is this we have...

*Just wondering if anyone out there knows the default Craft (password) for the IPSI board *logging into the IPSI board and trying to set the static address for the first time *with crossover cable telnet to the address and I get the [IPSI] prompt *type ispilogin returns with login ...

*We have 2 Merlin Magix systems using a point to point with a 100dcd at both ends slot 06 in site A (centralized voicemail) slot 02 in site B *we are breaking off the first 18 channels as Data and channels 19-23 voice and 24 as the d-channel *we have programmed the channels as data (R) my...

Is it possible to get SMDR output for an call accounting system Ip offce IP406 with a T1 also using 30 port Digital and 30 port Analog modules can it be done through LAN connection if Ip office is on the Network running 1.3.37 Pabell

We are trying to connect through the RS232 connection on the Merlin Messaging (2.0) .....set for 2400 8-N-1 and using the 355a adaptor (HyperTerminal)...no luck .....able to get through admin port on Processor with same 355a using winspm .....are we doing something wrong We just want to get a...

What is involved with setting up Softphone on your desktop ...we just upgraded to release 11 and are using 4612 IP phones (They work like a charm) off of the our Network .... set up clan (filesvc) svc port 1719 tftp is running on a standalone PC on the Network Thanks Pabell

*We have an issue with trying to transfer callers to extensions (some forwarded off-site , most are not) * If you come in over the T1 (PRI actually) you can depress "Transfer" dial the extension # or use the DSS button on the 4450 console (Programed as USER "Jim")and hang...

We have IP 406 with IP 400 Analog, and 2 IP 400 Digital Modules We are having an issue with Message Lights staying even though there are NO messages (new) or even saved in some cases .....in the Definty, Merlin Legend/Magix and even the Partner system you can dial a code to remove a MSG light...

we have a Ip office with 2 sites site 1 has 403 with 8 digital phones site 2 has 406 with (1)16 port analog tunk module (1) 16 port Digital (1) 30 port Digital using 40 digital sets We tried using an open analog port in site 1 and define it as Paging speaker in EXTENSIONS field under...

*Using an Ip office with 406 + 16 port analog co card with 2 330 switches (24 PORTS each)using Ip 4606 + 4612 Ip phones (DHCP connection) The Ip 406 Module has a PRI card (Centrex over the PRI) we are able to dial out over PRI (channels 1-23 are outgoing -incoming group 0) by inserting a 9...

*We are upgrading to a MAP40 platform and changing the platform .... *Current settings in Audix are Switch Interface Admin> Call Data Interface admin >DCIU *New platform is C-Lan......... ...all the docs say to remove C-lan Switch Interface (remove software) and install DCIU ,to move MO...

We are trying to setup a couple of IP 4606 phones off site using ptp T1 with Cisco routers between ... able to find the following :for my Ip 4606 settings phone = my local Ip address CallSv = not able to find in the Ip 600 where do we find this .... callSv Port = 1719 (default)says not to...