search for: hin87

Using a Polycom 550 and 650 phones on my Asterisk server for testing. I can't figure out why the volume is so low. How can I adjust the volume control on Asterisk? It's at max on the handset phones. Thanks! Hin

Hi, I have a strange problem with all of our Polycom 550 & 650 phones. I am running a TFTP server on my Asterisk server and option 66 Boot Host pointing to Asterisk on my DHCP server. The auto-provisioning is working because the phones are registering correctly with their extension. If I change the MAC.cfg file to another extension and reboot the phone, it will reflect the new ext. The

Is there a way to have Asterisk talk to a Avaya IP Office phone system? If it's possible, where can I find the instructions? ____________________________________________________________________________________ Never miss a thing. Make Yahoo your home page. http://www.yahoo.com/r/hs

I've installed AsteriskNow version 1.0.2 on a test machine and is trying to configure my Auto-Provision over http. I spent hours trying to figure out why it wasn't working and finally realize that my files are not displaying under the phoneprov directory. I tested by putting a test html file under /var/lib/asterisk/phoneprov/ and /var/lib/asterisk/phoneprov/configs/. Both directories I

I am using H.323 to create a trunk between Asterisk and Avaya IP Office system. Everything is working correctly, Asterisk can call Avaya and vise versa. Now I create a conference room with a user pin in Asterisk. Avaya can call into the conference room, but can enter the pin number. The error message they are getting is the "invalid pin number". However the pin number works if they

Using FreePBX, is there a way to play a beep sound when you are connected to a parked call? Right now, it's dead silence and we can't tell if the call has been connected. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100212/fbd18de3/attachment.htm

I have a question about the blind transfer using ##. This works great on our cordless phone, but there have been occasions that we can't transfer using ##. I was able to reproduce the issue by doing the following: 1) Call in from the outside line, 2) Ask the operator to transfer me to an extension using ##. 3) Get the voice mail greeting of the individual. 4) Hit 0 for the operator

I have discussed QoS with our ISP and in order to implement this, I need to make sure all VoIP packets are marked in the IP packet header (IPP bits?). Does Asterisk automatically marks the VoIP packets or do I need to do something in Asterisk? I need to do this for SIP and H323 protocols. Any information would be helpful. Thanks, Hin -------------- next part -------------- An

I have posted my problem on the link below, but didn't get any answer. I am hoping someone here can help me with this issue. Here's my problem: I am using H323 to talk between Asterisk and Avaya IP Office 500. For some strange reason, when we are talking on a VoIP call, we get disconnected after 10+ minutes. We have two other Elastix box, but none of them are getting disconnected. From

I am trying to set up a Grandstream GXV-3000 Video phone to Asterisk ver 1.2.21.1. The problem I'm having is that it can call other SIP phones, but not vice versa. Can someone tell me where is the problem? TIA! Here's part of my configurations: ---------- sip.conf ---------- ; 113 is the Grandstream phone [113] type=friend username=113 secret=secret context=default dtmfmode = rfc2833

I have quite a number of users complaining that when they are using handsfree to talk over a landline, the other end can't hear them. It's like the person is speaking 5 feet away and can barely hear their voice. However between internal SIP calls, it's fine. What could be the problem? -------------- next part -------------- An HTML attachment was scrubbed... URL:

Whenever I try to drag calls to the Parking Lot or On Hold, FOP would drop my calls. I have searched online and have found similar problem, such as the link below. I have tried their solution but still the FOP is not working correctly. I even installed the HUDLite server and is getting the same results. www.freepbx.org/forum/freepbx/users/flas...ot-transfering-calls Here is the log when I tried