Re: [asterisk-users] error receiving a fax ... but with a fax that was received without problems
Dear Joshua, I don't think this is it: first, this has been working in the past second, why would I get a message like the one below "exited non-zero" if everything is normal. In the past, all following lines were executed just fine (converting the tif to a pdf and sending it a gmail mailbox) I'm not convinced that everything is dropped after the ReceiveFax line, especially since the "Spawn extension" message applies to the ReceiveFax line itself (as if something goes wrong there) => any hints on where/how I can get a more detailed logging of this ? * == *Spawn extension (LocalSets, 502, 5) exited non-zero on 'DAHDI/4-1' On Sun, Sep 21, 2014 at 2:45 PM, Joshua Colp wrote: > Bart Remmerie wrote: > >> Dear all, >> > > Kia ora, > > When receiving a fax, the extension is "spawned", despite nothing but >> positive messages (see below) >> >> The sending fax considers it a success & the verbose output of asterisk >> gives a "FAX_SUCCESS" and a "NO_ERROR" error in the ReceiveFax command. >> >> The problem is that all the next steps (conversion of the fax to pdf & >> sending it to a mailbox) are never executed. When I do this manually, >> there seems nothing wrong with the received file >> >> Any hints ? >> > > After the fax is completed the call is hung up so subsequent dialplan > logic does not execute. You need to place the rest in the 'h' extension > which is executed upon hangup. > > Cheers, > > -- > Joshua Colp > Digium, Inc. | Senior Software Developer > 445 Jan Davis Drive NW - Huntsville, AL 35806 - US > Check us out at: www.digium.com & www.asterisk.org > > -- > _ > -- 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 > -- Bart Remmerie +32 (0477) 78.88.76 remme...@gmail.com -- _ -- 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
[asterisk-users] error receiving a fax ... but with a fax that was received without problems
Dear all, When receiving a fax, the extension is "spawned", despite nothing but positive messages (see below) The sending fax considers it a success & the verbose output of asterisk gives a "FAX_SUCCESS" and a "NO_ERROR" error in the ReceiveFax command. The problem is that all the next steps (conversion of the fax to pdf & sending it to a mailbox) are never executed. When I do this manually, there seems nothing wrong with the received file Any hints ? Asterisk output: *-- *Executing [502@LocalSets:5] *ReceiveFAX*("*DAHDI/4-1*", " */tmp/201409211416.tif*") in new stack *-- *Channel 'DAHDI/4-1' receiving FAX '/tmp/201409211416.tif' *-- *Channel 'DAHDI/4-1' FAX session '0' started *-- *FAX handle 0: [ 037.139057 ], entering CLOSING state *-- *FAX handle 0: [ 037.139097 ], entering CLOSING state *-- *Channel 'DAHDI/4-1' FAX session '0' is complete, result: 'SUCCESS' (FAX_SUCCESS), error: 'NO_ERROR', pages: 1, resolution: '204x196', transfer rate: '14400', remoteSID: '059323634' * == *Spawn extension (LocalSets, 502, 5) exited non-zero on 'DAHDI/4-1' Extensions-snippet exten => 502,1,Verbose(3,Incoming Fax) same => n,Set(FAXDEST=/tmp) same => n,Set(tempfax=${STRFTIME(,,%C%y%m%d%H%M)}) same => n,Verbose(3,- destination: ${FAXDEST}/${tempfax}.tif) same => n,ReceiveFax(${FAXDEST}/${tempfax}.tif) same => n,Verbose(3,- Fax receipt completed with status: ${FAXSTATUS}) same => n,Verbose(3,converting fax) same => n,System(/usr/bin/tiff2pdf ${FAXDEST}/${tempfax}.tif -o ${FAXDEST}/${tempfax}.pdf) same => n,Verbose(3,fax converted ... sending fax) same => n,System(echo "[TC] new fax" | mutt -s "TC:fax from ${CALLERID}" myn...@mymail.com -F /root/.muttrc -a ${FAXDEST}/${tempfax} .pdf) same => n,Hangup() -- _ -- 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
Re: [asterisk-users] issue installing voicemail imap support: imap_tk module missing
I'm using * ubuntu 14.04 LTS * asterisk 11.10 (from source) * imap-2007f (from source) Bart On Wed, Jun 11, 2014 at 6:36 PM, Tzafrir Cohen wrote: > On Tue, Jun 03, 2014 at 10:26:26PM +0200, Bart Remmerie wrote: > > Does anybody know where imap_tk is supposed to be / where it comes from ? > > Is it a part of asterisk / imap / linux / ... > > > > I can't seem to find any references other than related to asterisk, but > in asterisk I only can find it as a (unfortunately missing) dependency for > imap support for voicemail... > > What distribution is it? > > -- >Tzafrir Cohen > icq#16849755 jabber:tzafrir.co...@xorcom.com > +972-50-7952406 mailto:tzafrir.co...@xorcom.com > http://www.xorcom.com > > -- > _ > -- 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 > -- Bart Remmerie +32 (0477) 78.88.76 <#> remme...@gmail.com -- _ -- 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
Re: [asterisk-users] issue installing voicemail imap support: imap_tk module missing
Does anybody know where imap_tk is supposed to be / where it comes from ? Is it a part of asterisk / imap / linux / ... I can't seem to find any references other than related to asterisk, but in asterisk I only can find it as a (unfortunately missing) dependency for imap support for voicemail... Bart -- _ -- 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
Re: [asterisk-users] dahdi "hungup" after each ring
OMG, I simplified the Dial application arguments to avoid too many phones ringing while I was testing what went wrong ... Thanks for the help, I'll go back hitting my head against the wall now. Bart On Tue, May 27, 2014 at 4:16 PM, Richard Mudgett wrote: > > > > On Mon, May 26, 2014 at 3:12 PM, Bart Remmerie wrote: > >> Hi, >> >> I guess something's wrong with my chan_dahdi configuration, ... but I >> can't seem to get it. >> When I test incoming calls on a DAHDI-channel (incoming from pstn), >> asterisk seems to interpret it as a caller hangup after each ring. >> >> Any ideas. >> >> OUTPUT: >> >> -- Starting simple switch on 'DAHDI/5-1' >> >> -- Executing [s@from-pstn:1] *Verbose*("*DAHDI/5-1*", "*2,Incoming >> call from 059332640*") in new stack >> >> == Incoming call from 059332640 >> >> -- Executing [s@from-pstn:2] *Dial*("*DAHDI/5-1*", "") in new stack >> >> == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/5-1' >> >> -- Hanging up on 'DAHDI/5-1' >> >> -- Hungup 'DAHDI/5-1' >> >> -- Starting simple switch on 'DAHDI/5-1' >> >> -- Executing [s@from-pstn:1] *Verbose*("*DAHDI/5-1*", "*2,Incoming >> call from *") in new stack >> >> == Incoming call from >> >> -- Executing [s@from-pstn:2] *Dial*("*DAHDI/5-1*", "") in new stack >> >> == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/5-1' >> >> -- Hanging up on 'DAHDI/5-1' >> >> -- Hungup 'DAHDI/5-1' >> > > It is your dialplan that has the problem. You are not giving anything > to the Dial application so it doesn't know what you want to do. > Read the "core show application dial" documentation. > > Richard > > > -- > _ > -- 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 > -- Bart Remmerie +32 (0477) 78.88.76 remme...@gmail.com -- _ -- 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
[asterisk-users] dahdi "hungup" after each ring
Hi, I guess something's wrong with my chan_dahdi configuration, ... but I can't seem to get it. When I test incoming calls on a DAHDI-channel (incoming from pstn), asterisk seems to interpret it as a caller hangup after each ring. Any ideas. OUTPUT: -- Starting simple switch on 'DAHDI/5-1' -- Executing [s@from-pstn:1] *Verbose*("*DAHDI/5-1*", "*2,Incoming call from 059332640*") in new stack == Incoming call from 059332640 -- Executing [s@from-pstn:2] *Dial*("*DAHDI/5-1*", "") in new stack == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/5-1' -- Hanging up on 'DAHDI/5-1' -- Hungup 'DAHDI/5-1' -- Starting simple switch on 'DAHDI/5-1' -- Executing [s@from-pstn:1] *Verbose*("*DAHDI/5-1*", "*2,Incoming call from *") in new stack == Incoming call from -- Executing [s@from-pstn:2] *Dial*("*DAHDI/5-1*", "") in new stack == Spawn extension (from-pstn, s, 2) exited non-zero on 'DAHDI/5-1' -- Hanging up on 'DAHDI/5-1' -- Hungup 'DAHDI/5-1' *relevant part from chan_dahdi.conf* [outside-line](!) ; A template to hold common options for all phones connected outside signalling = fxs_ks ; tried ls and gs without any success callerid = asreceived ; dialtone_detect=always ringtimeout=8000 faxdetect=incoming context = from-pstn Bart Remmerie -- _ -- 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
Re: [asterisk-users] issue installing voicemail imap support: imap_tk module missing
Dear Chad, the folder is correctly named thirdparty Using "third party" was no more than a typo in the email I sent to the list, sorry for that. At the same time ... it's often these tiny mistakes that are the problem. Any other suggestions ? Bart Date: Thu, 22 May 2014 15:18:52 -0700 From: Chad Wallace To: asterisk-users@lists.digium.com Subject: Re: [asterisk-users] issue installing voicemail imap support: imap_tk module missing Message-ID: <20140522151852.0f501...@ws78.int.tlc> Content-Type: text/plain; charset=US-ASCII On Wed, 21 May 2014 23:09:28 +0200 Bart Remmerie wrote: > configure: *** The IMAP_TK installation appears to be missing or > broken. [...] > These are the steps I followed: > > sudo apt-get install libssl-dev libpam0g-dev > cd ~/src/asterisk-complete > mkdir third party > cd third party > wget ftp://ftp.cac.washington.edu/mail/imap.tar.Z $ tar zxvf > imap.tar.Z cd imap-2007e > make lnp EXTRACFLAGS="-fPIC -I/usr/include/openssl" IP6=4 > cd ~/src/asterisk-complete/asterisk/11 > ./configure --with-imap=~/src/asterisk-complete/thirdparty/imap-2007e/ In your mkdir and cd lines, you have a space between third and party. That would make two directories, and then cd into 'third' (probably). Then the path you passed to configure didn't exist. -- C. Chad Wallace, B.Sc. The Lodging Company http://www.lodgingcompany.com/ OpenPGP Public Key ID: 0x262208A0 -- _ -- 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
[asterisk-users] issue installing voicemail imap support: imap_tk module missing
Hi, I'm trying to install voicemail-imap support but there seems to be a missing module: imap_tk checking for mandatory modules: IMAP_TK... fail configure: *** configure: *** The IMAP_TK installation appears to be missing or broken. configure: *** Either correct the installation, or run configure configure: *** including --without-imap. My configuration Ubuntu 14.04 LTS Asterisk SVN-branch-11-r413595 These are the steps I followed: sudo apt-get install libssl-dev libpam0g-dev cd ~/src/asterisk-complete mkdir third party cd third party wget ftp://ftp.cac.washington.edu/mail/imap.tar.Z $ tar zxvf imap.tar.Z cd imap-2007e make lnp EXTRACFLAGS="-fPIC -I/usr/include/openssl" IP6=4 cd ~/src/asterisk-complete/asterisk/11 ./configure --with-imap=~/src/asterisk-complete/thirdparty/imap-2007e/ This is where I get the error message about the missing imap_tk module. in menuselect, imap_tk is mentioned as a dependency, but I can't seem to find it anywhere or activate it somewhere. Any similar experiences, any hints, any solutions on how to find the imap_tk module, or can the dependency be removed ?? Thanks & regards, Bart -- _ -- 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