Jean-Louis curty
2008-Jan-08 20:26 UTC
[asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?
Hi, I succesfully install spandsp chan_misdn and digium card. the rxfax works fine and I get the fax result by email. I would like to do the same using a Patton gw + zaptel but I can't receive fax anymore, the call comes in from ISDN in the Patton gw, patton sends it to asterisk, asterisk run a macro to make a tif file using rxfax, the tif file is correctly created but with a 0 size the call looks normal, 1 pages, 45 seconds and disconnect but the file is still 0, anyone succeeded in this ? Many (many) thanks! jean-louis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080108/7eb4091d/attachment.htm
Olivier
2008-Jan-09 06:48 UTC
[asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?
2008/1/8, Jean-Louis curty <jlcurty at gmail.com>:> > Hi, > > I succesfully install spandsp chan_misdn and digium card. the rxfax works > fine and I get the fax result by email. > I would like to do the same using a Patton gw + zaptel but I can't receive > fax anymore,which patton product do you use ? how are patton gw and asterisk connected to each other ? the call comes in from ISDN in the Patton gw, patton sends it to asterisk,> asterisk run a macro to make a tif file using rxfax, > the tif file is correctly created but with a 0 size the call looks normal, > 1 pages, 45 seconds and disconnect but the file is still 0, > > anyone succeeded in this ? > Many (many) thanks! > jean-louis > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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/20080109/117cba29/attachment.htm
Robert Moskowitz
2008-Jan-10 17:46 UTC
[asterisk-users] Is it possible to use spandsp and patton to do fax2mail ?
Olivier wrote:> So, I think in this case (Ethernet link), "standard spandsp" doesn't > help as it needs a TDM board.Nope not the case at all. I have been doing fax--ATA--lan--Asterisk->email for quite some time without ANY zaptel interfaces. Zaptel creats the pseudo interface and that does the job.> But, as spandsp has recently gained T.38 support, it could help to > build email2fax or fax2email but I have no experience myself in it.Oh????? I asked about this and did not get a good answer. I will dig more into it.> I would be very curious to know.Stay tune. Got to run an errand right now....> > Regards > > 2008/1/10, Jean-Louis curty <jlcurty at gmail.com > <mailto:jlcurty at gmail.com>>: > > ok sorry for the confusion created, > I mean isdn network , in other word tdm, > so tdm link connected to patton, patton connected in the lan via > ethernet speaking sip, > > jl > > 2008/1/10, Olivier < oza-4h07 at myamail.com > <mailto:oza-4h07 at myamail.com>>: > > > > 2008/1/10, Jean-Louis curty <jlcurty at gmail.com > <mailto:jlcurty at gmail.com>>: > > exactly > isdn patton -> eth/lan sip asterisk > > > so why is misdn installed for ? > it seems you don't have any ISDN card inside you Asterisk server. > > jl > > 2008/1/10, Olivier <oza-4h07 at myamail.com > <mailto:oza-4h07 at myamail.com>>: > > > > 2008/1/10, Jean-Louis curty <jlcurty at gmail.com > <mailto:jlcurty at gmail.com>>: > > I use 4960 and 4638 gw but it's applicable for any > patton gw ( analogue or isdn ) since it's always > the same way of configuring > - define ports > - define interface > - define services > > my config: > 1 asterisk > 1 patton ( let say 4638 ) > > the patton gw registers itself has a asterisk sip > peer, > inside the patton any sip call coming from > asterisk is routed to isdn ( outgoing calls ) > > > So patton and Asterisk are connected using Ethernet, > right ? > Or do you also have a TDM link between both boxes ? > > PSTN --- <TDM> --- Patton --- <Eth ? TDM ?> --- Asterisk > > in patton again , all isdn calls of any ports is > routed to the asterisk IP as a sip call, > > jl > > > > 2008/1/9, Olivier <oza-4h07 at myamail.com > <mailto:oza-4h07 at myamail.com>>: > > > > 2008/1/8, Jean-Louis curty <jlcurty at gmail.com > <mailto:jlcurty at gmail.com>>: > > Hi, > > I succesfully install spandsp chan_misdn > and digium card. the rxfax works fine and > I get the fax result by email. > I would like to do the same using a Patton > gw + zaptel but I can't receive fax anymore, > > > which patton product do you use ? > how are patton gw and asterisk connected to > each other ? > > the call comes in from ISDN in the Patton > gw, patton sends it to asterisk, asterisk > run a macro to make a tif file using rxfax, > the tif file is correctly created but with > a 0 size the call looks normal, 1 pages, > 45 seconds and disconnect but the file is > still 0, > > anyone succeeded in this ? > Many (many) thanks! > jean-louis > > _______________________________________________ > -- Bandwidth and Colocation Provided by > http://www.api-digital.com > <http://www.api-digital.com> -- > > 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 > <http://www.api-digital.com> -- > > 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 > <http://www.api-digital.com> -- > > 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 <http://www.api-digital.com> -- > > 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 <http://www.api-digital.com> -- > > 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 <http://www.api-digital.com> -- > > 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 > <http://www.api-digital.com> -- > > 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 -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Robert Moskowitz
2008-Jan-10 22:23 UTC
[asterisk-users] No NAT, but firewall mangles Register SDP
Nailed it! TCPdump on Trixbox 2.4 (Asterisk 1.4.17-1) going out and public side of firewall (Linksys WRT54G running Sveasoft) Firewall is configued NOT to NAT (public addressing on internal network. I stop asterisk (amportal stop). wait 30 min to insure timeout. Start both tcpdumps. Start Asterisk (amportal start). Get into Asterisk cli to insure registration was successful. Stop everything. Look at dumps with Wireshark. It very first SIP packet is a REGISTER coming from TB heading for Broadvoice (Only a SIP extension and Broadvoice SIP trunk defined). The UDP ports are SRC=5060 DST=5060. Length is different 5 bytes were added by the firewall, inside the SIP packet. From TB the Contact content is Phone#@IP#, while going out the firewall it is Phone#@IP#:5060 And this works. For calling from Broadvoice into TB. But if I run a firewall that does NOT mangle the SIP content it does NOT work. sip.broadvoice.com is really a Proxy server, and the INVITE coming from it has content that directs the RTP server over to a different Broadvoice server. That is when the Linksys box is there mangling the SIP content. With the regular firewall, TB gets an INVITE without the redirect content and tries to set up the RTP call with their proxy server which ICMP rejects the RTP packets. So..... What do I do so that without a mangling firewall this works? Is Broadvoice "broken" and can only work through a NAT? Will simply adding NAT=yes result in the Phone#@IP#:5060 in the first place? thank you all.