Hi! I'm new to this list. I tried to search the list archive for a solution on my current setup, but couldn't find any. We have an asterisk connected directly to the PSTN with 2 E1 lines through a Sangoma A102d interface. We also have a regular FAX machine. My question is how to get the fax service handled by asterisk? I want to cancel the analog line I have for the FAX machine today. What would be the best solution? Fax machine and asterisk is on the same LAN, not much load, with high end switches etc. Can I expect good results with using our existing FAX machine, connected to asterisk through an ATA box? Best Regards, Erik
Hi Erik, we use an ATA device connected to the fax machine. If you want to receive faxes, since Asterisk fax detection is not reliable, use one DID to link it directly to the ATA: you lose a number but you gain a fully-working fax! Giorgio Incantalupo. Erik Haider Forsen wrote:> Hi! > > I'm new to this list. I tried to search the list archive for a > solution on my current setup, but couldn't find any. > > We have an asterisk connected directly to the PSTN with 2 E1 lines > through a Sangoma A102d interface. We also have a regular FAX machine. > > My question is how to get the fax service handled by asterisk? I want > to cancel the analog line I have for the FAX machine today. > > What would be the best solution? Fax machine and asterisk is on the > same LAN, not much load, with high end switches etc. Can I expect good > results with using our existing FAX machine, connected to asterisk > through an ATA box? > > Best Regards, > > Erik > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > >
I have an TDM800P+ata+fax and work fine. This setup take 5 min. The best solution must be hylax fax + asterisk. But you need an asterisk specialist to make the setup and take more time. With this solution you can send fax and receive fax in your inbox and reduce toner/papper costs. Regards, Luis Morales On Wed, Sep 24, 2008 at 4:21 AM, Erik Haider Forsen <erikf at opera.com> wrote:> Hi! > > I'm new to this list. I tried to search the list archive for a > solution on my current setup, but couldn't find any. > > We have an asterisk connected directly to the PSTN with 2 E1 lines > through a Sangoma A102d interface. We also have a regular FAX machine. > > My question is how to get the fax service handled by asterisk? I want > to cancel the analog line I have for the FAX machine today. > > What would be the best solution? Fax machine and asterisk is on the > same LAN, not much load, with high end switches etc. Can I expect good > results with using our existing FAX machine, connected to asterisk > through an ATA box? > > Best Regards, > > Erik > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- --------------------------------------------------------------------------------- Luis Morales Consultor de Tecnologia Cel: +(58)416-4242091 --------------------------------------------------------------------------------- "Empieza por hacer lo necesario, luego lo que es posible... y de pronto estar?s haciendo lo imposible" Leonardo Da'Vinci ---------------------------------------------------------------------------------
Hey Erik, You can also check out pika technologies which supply chan_pika. This comes with a fax application that will let you do your faxes in asterisk (even using non-pika boards). Works pretty good... pikatechnologies.com mattm On Tue, Sep 23, 2008 at 4:51 AM, Erik Haider Forsen <erikf at opera.com> wrote:> Hi! > > I'm new to this list. I tried to search the list archive for a > solution on my current setup, but couldn't find any. > > We have an asterisk connected directly to the PSTN with 2 E1 lines > through a Sangoma A102d interface. We also have a regular FAX machine. > > My question is how to get the fax service handled by asterisk? I want > to cancel the analog line I have for the FAX machine today. > > What would be the best solution? Fax machine and asterisk is on the > same LAN, not much load, with high end switches etc. Can I expect good > results with using our existing FAX machine, connected to asterisk > through an ATA box? > > Best Regards, > > Erik > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080923/607d6321/attachment.htm
Hi all, Sorry to interrupt. I need some help regarding fax passthru mode. We are trying to configure fax passthru mode in asterisk using sip. For out of network calls/fax we use trunk configuration. i am using asterisk 1.4.2.The user has to use fax machine connected to their ata and dial the callee number, the call is originated just like a regular voice call. have not defined any special context for sending faxes. Have enabled t38 and canreinvite in peer/user and trunk configuration. But the fax is not going thru. Our service provider does support fax passthru. Following is the trunk and user/peer configuration: TRUNK CONF [TRUNK-OUT] type=peer host=XXX port=5060 context=default country=us dtmfmode=rfc2833 restrictcid=no canreinvite=yes insecure=no disallow=all allow=ulaw allow=alaw allow=g729 allow=gsm promiscredir=yes t38_udptl=yes USER/PEER [abc] username=abc type=friend secret=123 qualify=25000 nat=yes mailbox=12129339037 insecure=port,invite incominglimit=2 outgoinglimit=2 intl_trunk=TRUNK-OUT local_trunk=TRUNK-OUT host=dynamic dtmfmode=inband context=uscan canreinvite=yes callerid="Rizwan Qureshi" <12222222222> accountcode=1:0:abc amaflags=default disallow=all allow=ulaw allow=alaw allow=gsm t38_udptl=yes Any solutions? On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen <joakimsen at gmail.com>wrote:> On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro > <stotaro at totarotechnologies.com> wrote: > > ATAs work OK I guess, just make sure to use a loss less codec such as > ULAW. > > Since the OP stated he is using E1 lines then he should probably be > using alaw instead. > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Best Regards Rizwan Hisham -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080924/663949ae/attachment.htm
On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham <rizwanhasham at gmail.com> wrote:> Hi all, > Sorry to interrupt. I need some help regarding fax passthru mode. > > We are trying to configure fax passthru mode in asterisk using sip. For out > of network calls/fax we use trunk configuration. i am using asterisk 1.4.2. > The user has to use fax machine connected to their ata and dial the callee > number, the call is originated just like a regular voice call. have not > defined any special context for sending faxes. Have enabled t38 and > canreinvite in peer/user and trunk configuration. But the fax is not going > thru. Our service provider does support fax passthru. Following is the trunk > and user/peer configuration:They support passthru, and the originating send fax is what? PSTN? or VoIP ATA with t38 support? There has to one that does the t38, if the point where it gets converted to VoIP does not support t38 then passthru will not help you.> > TRUNK CONF > [TRUNK-OUT] > type=peer > host=XXX > port=5060 > context=default > country=us > dtmfmode=rfc2833 > restrictcid=no > canreinvite=yes > insecure=no > disallow=all > allow=ulaw > allow=alaw > allow=g729 > allow=gsm > promiscredir=yes > t38_udptl=yes > > USER/PEER > > [abc] > username=abc > type=friend > secret=123 > qualify=25000 > nat=yes > mailbox=12129339037 > insecure=port,invite > incominglimit=2 > outgoinglimit=2 > intl_trunk=TRUNK-OUT > local_trunk=TRUNK-OUT > host=dynamic > dtmfmode=inband > context=uscan > canreinvite=yes > callerid="Rizwan Qureshi" <12222222222> > accountcode=1:0:abc > amaflags=default > disallow=all > allow=ulaw > allow=alaw > allow=gsm > t38_udptl=yes > > > Any solutions? > > On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen <joakimsen at gmail.com> > wrote: >> >> On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro >> <stotaro at totarotechnologies.com> wrote: >> > ATAs work OK I guess, just make sure to use a loss less codec such as >> > ULAW. >> >> Since the OP stated he is using E1 lines then he should probably be >> using alaw instead. >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > -- > Best Regards > Rizwan Hisham > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
The fax is originated from a fax machine connected to an ata which supports t38. On Wed, Sep 24, 2008 at 11:54 PM, C F <shmaltz at gmail.com> wrote:> On Wed, Sep 24, 2008 at 5:43 AM, Rizwan Hisham <rizwanhasham at gmail.com> > wrote: > > Hi all, > > Sorry to interrupt. I need some help regarding fax passthru mode. > > > > We are trying to configure fax passthru mode in asterisk using sip. For > out > > of network calls/fax we use trunk configuration. i am using asterisk > 1.4.2. > > The user has to use fax machine connected to their ata and dial the > callee > > number, the call is originated just like a regular voice call. have not > > defined any special context for sending faxes. Have enabled t38 and > > canreinvite in peer/user and trunk configuration. But the fax is not > going > > thru. Our service provider does support fax passthru. Following is the > trunk > > and user/peer configuration: > > They support passthru, and the originating send fax is what? PSTN? or > VoIP ATA with t38 support? > There has to one that does the t38, if the point where it gets > converted to VoIP does not support t38 then passthru will not help > you. > > > > > TRUNK CONF > > [TRUNK-OUT] > > type=peer > > host=XXX > > port=5060 > > context=default > > country=us > > dtmfmode=rfc2833 > > restrictcid=no > > canreinvite=yes > > insecure=no > > disallow=all > > allow=ulaw > > allow=alaw > > allow=g729 > > allow=gsm > > promiscredir=yes > > t38_udptl=yes > > > > USER/PEER > > > > [abc] > > username=abc > > type=friend > > secret=123 > > qualify=25000 > > nat=yes > > mailbox=12129339037 > > insecure=port,invite > > incominglimit=2 > > outgoinglimit=2 > > intl_trunk=TRUNK-OUT > > local_trunk=TRUNK-OUT > > host=dynamic > > dtmfmode=inband > > context=uscan > > canreinvite=yes > > callerid="Rizwan Qureshi" <12222222222> > > accountcode=1:0:abc > > amaflags=default > > disallow=all > > allow=ulaw > > allow=alaw > > allow=gsm > > t38_udptl=yes > > > > > > Any solutions? > > > > On Tue, Sep 23, 2008 at 10:21 PM, Andrew Joakimsen <joakimsen at gmail.com> > > wrote: > >> > >> On Tue, Sep 23, 2008 at 10:02 AM, Steve Totaro > >> <stotaro at totarotechnologies.com> wrote: > >> > ATAs work OK I guess, just make sure to use a loss less codec such as > >> > ULAW. > >> > >> Since the OP stated he is using E1 lines then he should probably be > >> using alaw instead. > >> > >> _______________________________________________ > >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >> > >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona > >> Register Now: http://www.astricon.net > >> > >> asterisk-users mailing list > >> To UNSUBSCRIBE or update options visit: > >> http://lists.digium.com/mailman/listinfo/asterisk-users > > > > > > > > -- > > Best Regards > > Rizwan Hisham > > > > > > _______________________________________________ > > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > > Register Now: http://www.astricon.net > > > > 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 -- > > AstriCon 2008 - September 22 - 25 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Best Regards Rizwan Hisham -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080925/7845197a/attachment.htm