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Currently, the system terminates a call when the rtp hold timeout is reached At this moment i cannot change this option This is not a very graceful way of handling calls from a client perspective
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I believe the system should call back the extension that put the call on hold in the same way it does for calls that go to call park that are not answered. Allow dhcp option 42 to override ntp server Public group for the ipvideotalk category products
If you wish to ignore this product entirely leave the group
( you will not see any related boards / topics I worked with luis today from grandstream support who was very attentive and prompt to address this issue Problem solved in version 1.0.9.69!! My phone was at an earlier version so this is what needed to be done
Next upgrade to version 1.0.9.26 Then upgrade to version 1.0.9.69 Using the web ui of the gxp make these changes Worked on a mitel sx200 with a t1 (not a t1/e1), connecting a grandstream 4501 sip trunking, and were successfully making inbound/outbound calls
Did so by making the appropriate setting changes to both the 4501 and the pbx.
I plan for 40 grandstream phones connecting to a sip provider (voip.ms) direct This also requires paging ability where any phone can trigger paging with a button, speak into the phone and the audio is broadcast Some of the offices are very far apart with one being in a separate building The main building is 1000ft x 270ft with poe ports and wifi in all the spaces where phones will be
Hello everyone i have use ucm6202 firmware 1.0.19.27 When i call using webrtc using public ip it has no voice I using google chome version 77.0.3865.90 (official version) (64 bit) and the latest firefox but it’s the same Is that a problem of ucm
Dear grandstream customers, firmware 1.0.23.8 for gwn7602 is released as official
Here is the link to the release notes The firmware and release notes can be downloaded from here Please contact us should you … I would like to terminate the fxs line into a grandstream ata (most likely ht813) and then sip into the phone system
Is it possible to make the ht813 (or possibly a different ata model) appear as a “sip provider” or will it be necessary to place asterisk / freeswitch between the ata and the phone system? How to connect two 6108 pbx to direct call with internal numbers, how to do? Is there a possibility to change the option
