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Hi Guys, I last posted a thread in September last year regarding Music on hold or "your are the first person to join the conference" when using VMPro as a conference bridge. See post from last year - thread940-1770756 Would like to know if Release 10 was able to do what I require below. With...

Hi Westi, So my setup will not allow music to play until the 2nd person joins. I do not have any users active on the IPO its merely a bridge for meet me conference configured via Voicemail Pro. T

Janni78 , I have used voicemail pro to setup the conference bridge with the pin codes. Not sure what the difference is or if there is a difference between personal and the way Ive currently got it setup as.

Hi Guys, Need some guidance please. Ive installed ipo 9.1.5 on a virtual machine to be used as a conference bridge.. All working. The client has request that music needs to play until the second party joins - Is this possible? Setup: 1) user calls in and prompted to enter conference id 2)...

Hi Guys, Just a correction to the shortcode. 2 short codes of the ip office 72XXXXX and 71XXXX feature : conference meet me Tel Nr . (yep a dot) 72XXXXX for 5 digits 71XXXX for 4 digits DaIPOTek - Unfortunately these number were migrated from another system. it was 4 and 5 digits there.

Hi Guys , I'm running 9.1.5 server edition and the client has request that I setup and existing conference bridge on the ip office. They are using 4 and 5 digit extension and would like the conference bridge number to match there extensions. I got the 4 digits working but for some reason...

Amriddle01, Im assuming I need to set the local URI , contact etc in the SIP line to *** and then configure the ICR to that line group id with destination .. regards

Forgot to mention If the calling party is sending 10 digits then the SIP entry needs to contain the 10 digits even if the called party is configured with lesser digits i.e Calling party calls 011 555 555 Extension /user = 555 Sip entry need to be configured with 011555 555. This worked for

Try adding the full number to the SIP tab of the user. SIP Name and contact to be the extension number you are trying to call. Give it a go

I have no server licenses on the server edition. Also the IPO 500V2 is release 8.1.79. I'm only using the IP 500V2 for my outbound trunks - That is its only use. Im busy testing the following " http://www.tek-tips.com/viewthread.cfm?qid=1764093 " and I think I got it working just need to test...

Hi Guys, I have a IPO Server edition 9.1.2 with SIP trunks connected to an IPO 500 V2. I can call extension between the two. Im trying to call externally from the IPO Server edition via the bri on the IPO 500 V2. The call does reach the IPO 500v2 but complains that it cannot find user. I have...

thanks guys... Really appreciate all the assistance.

Thats exactly what i'm looking for if youre able to use the played dial tone as a normal dial tone. Can someone confirm, I don't have a lab to test, Thanks

So when the dial tone in VM is active could one dial externally or as per normal before it times out?

I think it is dead air messages. We have a few clients with analogue trunks as well. When calling an extension from external and then externally hang up the internal extension still rings. So when the no answer timer kicks in the external party making the call already hung up. One of my...

Hi Guys, Does the IP Office (any recent version) have the capabilities of calling out after 5 sec after the handset has been lifted. This is similar to hot dialing with a difference in that if you decide to dial a number before the 5 sec has elapsed then the feature gets disabled. The customer...

Hi Intrigrant, Are you referring to setting the SNTP_SERVER pool.ntp.org to point to the IP Office. If so then I already done that as well as the local NTP server onsite. Does the NTP setting for the 1120e SIP get obtained from the device script above?

Hi All, I have a server edition IP office 9.1.3 using 1120e phones that were decommissioned from a CS1K. All is working well except the time on the phones needs to be manually set. I have provisioned a FTFP server to allow for firmware updates from UniStim to SIP - All working Also I included...

Hi Janni78, Ive done the test on 9.0.7 , 9.1.3 and the latest 8.1. Same result. Why would it not work on SIP phones? is it not the system sending a message to the SIP phone in the same way a normal message would activate the MWI.

In the process of getting the IPOSS registered. If possible can someone test the Short Code above and confirm that it works on both H323 and SIP phones (1120e or E129)