First of all, thanks for the response :) Noticed are ok, I've added maxregexpire=300 in iax.conf, as shown below...
We use ubuntu 12.04, asterisk was installed from its repository: srv-faxserver*CLI> core show version Asterisk 1.8.10.1~dfsg-1ubuntu1 built by buildd @ yellow on a x86_64 running Linux on 2012-04-24 12:47:04 UTC Here configuration files: :: iax.conf :: [general] disallow=all allow=alaw bindport=4569 bindaddr=127.0.0.1 language=it srvlookup=yes maxregexpire=300 [modem1] type=friend context=fax disallow=all allow=alaw username=modem1 secret=************ qualify=yes notransfer=yes host=dynamic callerid="Fax" <XXXXXXXXXX> requirecalltoken=no [modem2] type=friend context=fax disallow=all allow=alaw username=modem2 secret=******* qualify=yes notransfer=yes host=dynamic callerid="Fax" <XXXXXXXXXX> requirecalltoken=no [modem3] type=friend context=fax disallow=all allow=alaw username=modem3 secret=****** qualify=yes notransfer=yes host=dynamic callerid="Fax" <XXXXXXXXXX> requirecalltoken=no [modem4] type=friend context=fax disallow=all allow=alaw username=modem4 secret=*********** qualify=yes notransfer=yes host=dynamic callerid="Fax" <XXXXXXXXXX> requirecalltoken=no [modem5] type=friend context=fax disallow=all allow=alaw username=modem5 secret=************* qualify=yes notransfer=yes host=dynamic callerid="Fax" <XXXXXXXXXX> requirecalltoken=no [modem99] type=friend context=fax disallow=all allow=alaw username=modem99 secret=*********** qualify=yes notransfer=yes host=dynamic callerid="Fax" <XXXXXXXXXX> requirecalltoken=no :: sip.conf :: [centralino] secret=***************** dtmfmode=rfc2833 canreinvite=no context=default host=dynamic type=friend nat=yes port=5060 qualify=yes disallow=all allow=alaw ----- Messaggio originale ----- > Da: "James Cloos" <[email protected]> > A: "Gianni Fioretta" <[email protected]> > Cc: "Asterisk-users" <[email protected]> > Inviato: Giovedì, 4 luglio 2013 23:16:11 > Oggetto: Re: [asterisk-users] Asterisk + iaxmodem + hylafax makes sometimes > wedged for hylafax > > >>>>> "GF" == Gianni Fioretta <[email protected]> writes: > > GF> -- Executing [0224300258@fax:1] Dial("IAX2/modem2-3460", > "SIP/centralino/0224300258") in new stack > GF> == Using SIP RTP CoS mark 5 > GF> -- Called SIP/centralino/0224300258 > GF> -- SIP/centralino-00000284 is making progress passing it to > IAX2/modem2-3460 > GF> -- SIP/centralino-00000284 is ringing > GF> -- SIP/centralino-00000284 is making progress passing it to > IAX2/modem2-3460 > GF> -- SIP/centralino-00000283 is making progress passing it to > IAX2/modem4-8449 > GF> -- SIP/centralino-00000283 is ringing > GF> -- SIP/centralino-00000283 is making progress passing it to > IAX2/modem4-8449 > GF> -- SIP/centralino-00000284 answered IAX2/modem2-3460 > GF> [Jul 4 16:49:55] WARNING[22988]: chan_sip.c:9123 process_sdp: Failing > due to no acceptable offer found > > That last line above shows that an outgoing fax attempt failed because > the sip end wasn't able to negotaiate a codec for that part of the call. > > It looks like it was modem2's call which failed; modem4's call seems not > yet to have been answered. > > I don't know whether that is what triggers the wedge, but the failure to > negotiate a codec for the sip/rtp leg probably is a configuration bug. > > Which version of asterisk? Self compiled or a distribution's version? > > The sip.conf and iax.conf might help debug it. (Elide passwords, of course.) > > If you run the conf files through something like: > > :; egrep -v '^[[:blank:]]*;' iax.conf|egrep -v '^$' >/tmp/short-iax.conf > > before editing the password lines it will be easier to read them. > > -JimC > -- > James Cloos <[email protected]> OpenPGP: 1024D/ED7DAEA6 > -- Gianni Fioretta - [email protected] YetOpen S.r.l. - http://www.yetopen.it/ via Carlo Torri Tarelli 19 - 23900 Lecco - ITALY - Tel 0341 220 205 - Fax 178 6070 222 -------- D.Lgs. 196/2003 -------- Si avverte che tutte le informazioni contenute in questo messaggio sono riservate ed a uso esclusivo del destinatario. Nel caso in cui questo messaggio Le fosse pervenuto per errore, La invitiamo ad eliminarlo senza copiarlo, a non inoltrarlo a terzi e ad avvertirci non appena possibile. Grazie. -- _____________________________________________________________________ -- 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
