search for: ag168v

...s through my FXO. After much tinkering I was able to get my gateway (wellgate 3701A) configured to a point where I have some success, but no real joy. I have configured the RTP Payload type (or RFC2833 Payload type) to 101. I don't have a clue what this means, but I took the 101 from my AG168V ATA's configuration screen, as I know that device seemed to work fine through the old HT-488 fxo(via rfc2833). I then changed my asterisk extensions for both the FXS and FXO on the wellgate to include dtmfmode=rfc2833. This has brought me to a point where both my hardphones (ATA's) and...

http://www.x100p.com/products_2.htm Anyone ever use this box? How's it compare with the Iaxy? I'd like to buy one or the other.. The Iaxy is appealing because to me, it seems less "no name", but this one says that it supports using hostnames, whereas apparently the iaxy only supports IP addresses?? That's appealing to the dynamic DNS guy in me! :-) Any

...tinkering I was able to get my gateway (wellgate 3701A) > configured to a point where I have some success, but no real joy. > > I have configured the RTP Payload type (or RFC2833 Payload type) to > 101. I don't have a clue what this means, but I took the 101 from > my > AG168V ATA's configuration screen, as I know that device seemed to > work > fine through the old HT-488 fxo(via rfc2833). > > I then changed my asterisk extensions for both the FXS and FXO on the > wellgate to include dtmfmode=rfc2833. > > This has brought me to a point where...

Hello All, I am looking for a IAX2 analog telephone adapter, just want to ask your views on which ones are bad, good and the best. Thanks in advance, Dinesh Birlasekaran Network Engineer, ComIT, Institute of Molecular and Cell Biology 61 Biopolis Drive, Singapore 138673 HP : 92962676 DID : 65869804 Fax : 67791117 Email : dinesh@imcb.a-star.edu.sg WWW: www.imcb.a-star.edu.sg

I find that DTMF does not work reliably if jitterbuffer=on for certain IAX providers. For instance, DTMF tones are missed entirely about half the time when I dial into an exgn.net account. However, it always works fine for an unlimitel.ca account. Someone else has seen this too: http://bugs.digium.com/view.php?id=6011 Can anyone suggest a workaround (other than jitterbuffer=off)? - Mike

Have a customer running a 3rd party PBX implementation based on Asterisk, not utilizing SIP inbound and outbound calls I believe are coming through a Digium TDM2402B. They are utilizing Polycom phones. They are experiencing frequent static on the line, and overall insufficient volume on conversations. They are in a bit of a rural area, I was curious if anyone thinks it could be an issue with