Sorry pushed send too fast
On 2/08/2012, at 5:59 AM, Eric Wieling <ewiel...@nyigc.com> wrote: > Yup, there is your problem. Tell hylafax to extend the amount of time before > it times out. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz > Sent: Wednesday, August 01, 2012 1:53 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem > > This is /etc/dahdi/system.conf > > fxsks=1-4 > fxsks=5-8 > echocanceller=mg2,1-8 > loadzone = us > defaultzone=us > > > And /etc/asterisk/chan_dahdi.conf > language=en > context=fax-out > signalling=fxs_ks > faxdetect=both Why both here? Its going to listen for a fax tone on outbound. Can you change to inbound > echocancel=no > cancallforward=yes > canpark=yes > transfer=yes > echocancelwhenbridged=yes > group=3 > callgroup=3 > channel => 4-4 > > Is the actual hardware that I should change? > > Thanks, > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Eric Wieling > Sent: Wednesday, August 01, 2012 10:15 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem > > Sounds like DAHDI/4 is a FXO port. FXO ports are considered answered when > dialing is completed. > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of motty.cruz > Sent: Wednesday, August 01, 2012 1:11 PM > To: 'Asterisk Users Mailing List - Non-Commercial Discussion' > Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem > > that is correct! The reason I think is because when Dahdi "answered" > iaxmodem thinks is the remote fax machine that answered, but it reality it > keeps ringing, if the remote fax machine answer within the first ring then > iaxmodem connects but if not it does not detect a fax. > > Here is a sip extension dialing throught the same context. > > dxxx*CLI> > == Using SIP RTP CoS mark 5 > == Using UDPTL CoS mark 5 > -- Executing [xxx1463@fax-out:1] Dial("SIP/507-00000000", > "dahdi/g3/wwxxx1463") in new stack > -- Called g3/wwxxx1463 > -- DAHDI/4-1 answered SIP/507-00000000 > -- Hungup 'DAHDI/4-1' > > Dahdi answered, but after dahdi answer it rang for 4 rings before remote fax > answered, if I would be the iaxmodem I would had given up by then, > > Do you see my problem? Anybody else experienced same issue? > > Thanks, > > > > -----Original Message----- > From: asterisk-users-boun...@lists.digium.com > [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Tim Nelson > Sent: Wednesday, August 01, 2012 9:46 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] Asterisk Dahdi 1.6.2.23 Iaxmodem > > ----- Original Message ----- >> Thanks Tim, >> I tried your suggestion below the logs: >> >> -- Accepting AUTHENTICATED call from xxx.xx.xx.xx: >>> requested format = ulaw, >>> requested prefs = (), >>> actual format = ulaw, >>> host prefs = (ulaw), >>> priority = mine >> -- Executing [xxx1463@fax-out:1] Dial("IAX2/503-7761", >> "dahdi/g3/wwxxx1463") in new stack >> -- Called g3/wwxxx1463 >> -- DAHDI/4-1 answered IAX2/503-7761 >> -- Registered IAX2 '503' (AUTHENTICATED) at xxx.xx.xx.xx:4570 [Aug >> 1 09:04:59] NOTICE[3392]: chan_iax2.c:8486 update_registry: >> Restricting registration for peer '503' to 300 seconds (requested 60) >> -- Registered IAX2 '503' (AUTHENTICATED) at xxx.xx.xx.xx:44145 >> [Aug 1 09:05:03] NOTICE[3391]: chan_iax2.c:8486 update_registry: >> Restricting registration for peer '503' to 300 seconds (requested 60) >> -- Hungup 'DAHDI/4-1' >> == Spawn extension (fax-out, xxx1463, 1) exited non-zero on >> 'IAX2/503-7761' >> -- Hungup 'IAX2/503-7761' >> >> >> [root@drew home]# faxstat -s >> HylaFAX scheduler on host.xxxxx.com: Running Modem ttyIAX0 >> (+1.xxx.8626): Running and idle >> >> JID Pri S Owner Number Pages Dials TTS Status >> 9 126 S root xxx1463 0:1 1:12 16:10 No carrier >> detected >> > > Your setup looks correct. Can you connect a normal analog phone to the POTS > line and dial that fax number directly? I just want to see if that is > successful or not, indicating if the problem is PSTN related (need to dial > 10 digits, or 1+10 for example in the US). > > The interesting thing is the result within Hylafax is 'No Carrier' which > means the call was indeed answered, but fax was not present on the other side. > > --Tim > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to > Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
