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Dear all, Recently we have seen end of NEC telecom business following with Panasonic, Siemens.next who ?Don't know. But it's clear,Big players are existing this market. Do any one have idea what will be the faith of IPPBX, VOICE segment.

Is there any idea why big players are exiting this business? ISDN circuit are about to die.in my country PSTN and E1 lines are phase out and no support from telco .Only SIP trunk or single sip account,GSM options are available.Technolgy is changing drastically than our expectations keep updated...

yes the news is true NEC is ending its business like Panasonic done.

Thanks to reply client has grandstream video phone and synched his android phone contacts with grandstream.in contacts if there is +91XXXXXX number then customer want +91XXXXXX must be dialed by single touch but there is no option in sip server to eliminate + which is not dtmf.

Dear friends, We have using grandstream GRP2615 how i can strip + from oudial,specifically from phone .

Yes from PC,ping is working. Gateways also confirmed with IT person they are right.

Dear All, We have two locations connected over VPN where two voip devices are working one network is 192.168.1.0 and other 192.168.2.0 voip 1 -192.168.1.202 voip 2-192.168.2.202 both IPs are able to ping from PC in both network but when I try to ping each other through their telnet they are...

Hi I have installed sl2100 ver 4.1 with sip trunk and with active ip trunk license after few days when system restart ip trunk license disappears and need to load again remaining programing stays as it is

Hi friends, I am telecom vendor and has very less interaction with these firewalls. Recently I had installed VOIP IPPBX behind firewall old one IPPBX is behind fortigate firewall. new IPPBX IP is 192.168.3.101(Sonicwall network-192.168.1.1) and fortigate network is 192.168.2.1. Both side...

Hi friends, I installe matrix comsec IPPBX with grandstream GRP2601P opensip endpoint,When I calls another SIP extension the RBT is of dual frequency means I can hear two different ringing tones.But my fault finding is like this 1.xlite soft phone working well with same pbx in same network...

Dear Team, Avaya Sip J129 Phone regesterd with Avaya IP500, customer is having 2 network for Data and Voice, when the new phone is connected to network phone taking the nework Domain name automatically, and also the Auto Provisioning Service is Activated, i want to deactivate the Auto...

You can assign multiple pots for landing for single sip trunk,hunt groups are available in multi port ATA.for two ports ATA hunt group is not available.If you have multiple sip accounts from telco then you can map one to one.

Hi all there is problem in our network setup clients are on DHCP when one client get IP address same time other client also gets same IP cant able to understand what is going on there is firewall and layer three switches is there any policy to be changed

Dear snowman50, Thanks for the Reply The Voicemail Pro Service is Working in the Services, application started. Porblem is only in VM PRO Server, VMPRO is not finding the Server, at the same network if i change the PC for VMPRo its Working Properlly Checked with Firewall / Antivirus Disabled,

Dear Team, I Have Installed Avaya Voicemail Pro client in Windows Server 2016 with IP Office The Settings has set in IP Office for Voicemail, also Servce working properlly, but voicemail not geting connected to the System, i tried to do setup in anather machine with same settings, it workes...

Dear Team, Iam Using Avaya J139 SIP Phones with Ip Office R 11 i want to programmed the Buttones, i have tride to programmed buttones from IP Office but not affected, from J139 web configuration is showing 04 buttones in the parameter but 03 is default for Apearance. please suggest.

give priority to voice traffic in firewall check codec settings in phone and pbx E.g g711 u or Gg711 a tcp/udp flood disable if there is one way audio use symmetric rtp. check network latency must be less than 150ms nat setting in firewall must be disabled

Analog lines doesn't have start stop bits when incoming call answered by DISA and caller disconnect call in middle then call keeps ringing until its ring timer.you cant do anything with analog lines in such case you need PRI. second check whether your lines are crossed with other lines some...

Yes,you need USA version.

It has inbuilt GUI connect laptop with LAN port on NS300 IP 192.168.0.101 put in browser. access machine from browser