Try using an IAX client namely Dante's IAX SOftphone. IAX will handle the ADSL modem and their NAT's and Firewalls much better than a SIP client.
Let me know how it works.
Remember guys he wants to give them connectivity but at an affordable price. You can configure Asterisk to talk to the other switch through SIP and also have the ability to support PSTN lines at that location. Its the most affordable option and its a solid solution.
Thanks for the feed back guys. The route number was not changed. The RLI were all rebuilt anyway just to make sure they were programmed correctly. Remember the digits go where they are suppose to go. But when the digits reach their destination the call drops. And I get the message call is not...
Hey Guys,
I just recently changed my ITG card in my Option11C from the double slot to the single slot Media Card. The vendor has programmed everything but this is the situation as it stands. My extension to extension dialing between my two Option 11C switches is working, so I can call 42XX and...
Thanks Option11Newby, I was looking in the wrong place. I did not explore the table with the active DN's. I was looking down the bottom at the line access information. As Mark said, it was right there infront of me. I was just not looking in the right place.
Thanks alot guys you always help me...
MARKHILLMAN, There is really no need to insult anyones intelligence here. This is a forum for discussion and a way to help others. I am a certified Meridian technician and I have worked on many Options, BCM 200, 400. I am new to the 50 so forgive me if I am not seeing the options where they...
I took a look and I dont see anythiong in regards to call forward under the line access tab. I am not sure why. Do I have to turn on Call forwarding anywhere first?
I have a BCM 50 that I am trying to configure call forward of unanswered calls to voicemail. But under the active sets I dont see any fields dealing with call forward. Can anyone assist me in programming call forward on a set on a BCM 50.
Please.
Thanks,
I have two lines connected to my asterisk box. I however only want one line to be answered by the auto attendant. The other line will be used for outgoing calls only. I tried changing the context of the line. The original context was from-pstn. I changed that and it worked for a while but then...
I checked on the TGAR and RPL's and everything looks good. The TGAR and TARG do not match and the route access code was added to the RPL. Any other advice?
Good Day,
I have a situation where I have a VOIP gateway connected to my Option 11C used for International calls. I wanted to be able to offer my users access to this service from remote locations. I do not have the DISA package so I thought about it and what I did is assign a DID number for...
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