I am unable to have hylafax pick up an incoming call. This is no doubt a simple configuration error. I would appreciate some direction as to which bits to set and how to set them.
Arch = x86_64
OS = FreePBX-2.11.0beta2.8 (CentOS-6.3) (freepbx)
asterisk = asterisk11-11.3.0-44_centos6 (freepbx)
hylafax = hylafax+.x86_64-5.5.3-1.el6 (epel)
iaxmodem = iaxmodem.x86_64-1.2.0-1.el6 (self packaged)
We automated our fax system in February of 2007 using external modems together with hylafax and avantfax. We are in the process of moving this application, together with its archives and data, to a new server running asterix and using a Digium, Inc. Wildcard TDM410 4-port analog card (rev 11).
We have installed and configured asterisk to use devices and users rather than extensions. We have created several users and extensions for both softphones and a soma320. Using these devices we can place and receive calls via the PSTN through asterisk. The fax modem devices and users share the same four digit ids and the devices are configured to use the IP ports that correspond to their device ids (4570/4570/4570 and 4571/4571/4571).
We have installed and configured hylafax+ as required. It is running without problem insofar as we can determine.
We have installed and configured iaxmodem together with two software modems, ttyIAX0 and ttyIAX1 in /etc/iaxmodem. Our /etc/inittab file contains these entries:
Faxgetty is present:
Iaxmodem is running and the two modems are registered:
-- Registered IAX2 '4571' (AUTHENTICATED) at 127.0.0.1:4571 -- Registered IAX2 '4570' (AUTHENTICATED) at 127.0.0.1:4570 > Saved useragent "snom320/126.96.36.199" for peer 41710
However, when I call in on the POTS line I inevitably get this result in asterisk:
-- Starting simple switch on 'DAHDI/1-1' -- Executing [[email protected]:1] NoOp("DAHDI/1-1", "No DID or CID Match") in new stack -- Executing [[email protected]:2] Answer("DAHDI/1-1", "") in new stack -- Executing [[email protected]:3] Wait("DAHDI/1-1", "2") in new stack -- Executing [[email protected]:4] Playback("DAHDI/1-1", "ss-noservice") in new stack -- <DAHDI/1-1> Playing 'ss-noservice.ulaw' (language 'en')
As far as I know I am not matching on either CID or DID for anything and I am unsure as to how I should proceed to route incoming fax calls through asterisk so that hylafax gets them but still route incoming voice calls so that the ring-group gets those. The intent is that for now the PSTN lines will handle both incoming and outgoing faxes and voice traffic routed through asterisk. In production we plan to have two ports dedicated to fax traffic and six to voice traffic (which I realize will require an additional 4-port FXO card).
What must I do in freePBX to accomplish this and how do I do it?