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Hi, I have Avaya Aura 10.1 and SIP phone J139 and J189 with latest firmware since I have programmed bridge appreance and busy indicator for boss secratary feature. I am able to program buttons and it shows on display but it do not indicate while off hook or calling. Enabled bridge appreance in...

thread690-1751211 Hey How do i generate this CMS Investigation.zip file

Hey, I tried the same but it is not working. I tryed almost all the short codes. Call is getting connected after redial but not working when i am dialing.

Hi, I am experiencing an issue with dialing the toll-free number 1800XXXXXXX from an H.323 extension. Strangely, the call connects successfully when I redial the same number. However, the toll-free number works well when dialed from a SIP extension. Here are the details of the problem: I am...

Ok thanks for help...will update you once network changes are made.

Hi, Ya actually client has provided the range. I find only some phones have the above issue i can say 4 to 5 out of 230 phone have issue were it gets resolved when i change the IP of that phone. I had set the default route as 172.35.0.0 255.255.0.0 and 172.35.83.1. where everything is working...

Hi Guys, I have an setup of IP Office 11.1. and Grandstream open SIP Phones with latest firmware. I am facing issue with one way audio intercom and SIP trunk. When i call from Phone 1 - 172.35.83.77 to Phone 2 - 172.35.85.203 or Phone 1 to Mobile No i am not able to hear on Phone 2 or Mobile...

Hi, I got server edition with resiliency and I have to provide RCA for the issue. As unfortunately services gone down unable to specify the reason as i took some logs and here are they OP ipoffice Config File: Header Info: crc32=3643865663, length=134423, ver(11.0.4.3.0 build 3), unit_type=0 OP...

Hello, need some help as i have IP Office 10 SP2 with 1608L h.323 phone. When i receive call and after picking up call... opposite party disconncts the call still my phone does not gives engage tone for 15 sec and after 15sec it get ideal. It happens with all my H.323 extensions. Still i dont...

Hi..I have IP Office 10.2 and 1608 IP Phone works on H323. I have issue when the called party drops the call which was incoming for 1608...the call does not disconnect automatically till we keep the receiver.

Hi..I have J129 deskphone on which i have enabled web interface. Can any one tell Login ID and Password for web interface for J129. I tryed extension and password in supervisory setting also tryed admin but any of it not working. Can any 1 help.

It is, please find recent log from switch. https://files.engineering.com/getfile.aspx?folder=27811ad4-1da2-4613-bc92-45789fbbec46&file=157422710865679611974.jpg

Not working even after above settings done. https://files.engineering.com/getfile.aspx?folder=8ab00cec-0f3d-419d-8655-f63bf82308d9&file=1574051829184352371415.jpg

IP office with 1608 phones is enabled with forward on No answer and Busy. When the call comes to 1608 and forwarded to other extension after No answer. 1608 does not show as missed call for forwarded call and the user does not come to know whose call missed which was forwarded

IP office 10 with IP 1608 phones which are connected directly to switch port and PC LAN is connected to phone PC port as port security is enabled on switch port with 2 MAC allowed on the port. What happens is when user ON the PC in morning the switch reads error MAC of PC as eh-gd-id-00-00-00...

What if i do not end call manually. In my system it search of primary ip and so on but do not get register on secondary server.

Hi, I have server edition R10 with2 servers primary and secondary, 35 H.323 phones 1608,9608,1616 as well as SIP phones E129. Well SIP works well but when H323 phone is busy on call and primary server goes down, call do not get terminited till some time after that which ever H323 phone was busy...

Nitrigrant@ yes it works well if i change ip of that phone. It automatically choses random port which is not defined in server. For incoming it is taking port 5060 but for outgoing voice it is taking some random port.

How? What kind of changes has to be done in switch can u elobrate.

Hello I have IPO 10.0 I am facing issue with unable to hear voice from callie and call to number. There is no firewall between server, PoE switch and phone(sip e129 and IP 1608) If i change IP of that phone all works well. I tried changing port, reconfig, reboot, device change but non of that...