No subject

to a fax (modulation and demodulation) over VoIP.
Try again, the fact that you succeeded twice was pure luck, and as far
as I understand that didn't even work out.
Switch back to TDM. Your dial up modems want that magic thing called
timing and no jitter that only TDM will give you.


On Mon, Aug 29, 2011 at 2:56 PM, Aaron Krohn <akrohn at ewebforce.net> wrote:
> We have an Ascend Max router that has a PRI plugged into it, providing ou=
r
> current dialup users with web access. The PRI is no longer cost effective=
,
> so I've been tasked with converting it to something cheaper.
>
> We added a DID to our (existing) asterisk system (we have a couple dozen
> voip customers). We added an Adtran 908 to convert the VoIP signal into a
> virtual PRI for our MAX router to handle dialup calls.
>
> When dialing into the number with a modem, MAX sees the call, picks up an=
d
> apparently tries to negotiate it, but eventually disconnects. It HAS,
> however, twice, successfully connected the call for a short time, but no
> browsing was possible. I've done some debugging output on the Adtran whic=
h
> seems to indicate that an RTP BYE command is received: TM.T01 01
> SipTM_Connected =A0 =A0 =A0rcvd SIP call-leg request: BYE ... This is the=
 first
> difference between a debug output where the call connected and one that d=
oes
> not work. This is the one that doesn't work.
>
> TL;DR - Is it possible to do dial-up through Asterisk? Or is it like t38
> faxing that is 'possible' to get working, but not if you have other job
> responsibilities?
>
> Thanks!
>
> --
> Aaron Krohn
> Web Force Systems
>
> Business Office:
> 131 Dillmont Drive, Suite 201
> Columbus, OH 43235
> Direct: =A0614-384-0019 =A0 =A0Fax: =A0614-785-0871
> Tech Support / Help Desk Direct: =A0614-384-0020
>
>
> --
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