similar to: Best email client for Asterfax

Trying to install asterfax-1.1-freeb2.i386.rpm, I get following error. How can I get rid of it. Installing jre Installing libtiff Installing ghostscript Installing Xvfb Installing openoffice.org Installing spandsp Installing spandsp0.0.3 Spandsp did not install correctly. error: %post(asterfax-1.1-freeb2.i386) scriptlet failed, exit status 1 -------------- next part -------------- An HTML

Hello, Have any one had any success installing asterfax. Have installed and it seems like its working, it accepts the fax that was sent via email, dail's out but then just freezes. It then complains about rcfax that needs to be batched or it closes asterisk down. Any advice. Kind Regards, Kyle Vorster

Hi, Has anyone able to make asterfax work on asterisk, specifically asterisk 1.4. reading the documentation it prefers asterisk 1.2, unfortunately i've already setup my asterisk and it's working ok at that version. TIA. regards, nha

Hi All, I am getting following error when i start AsterFax: Please help me to solve this issue: [root at prefroam asterfax]# ./asterfax.sh log4j: Threshold ="null". log4j: Retreiving an instance of org.apache.log4j.Logger. log4j: Setting [au.com.noojee.asterfax] additivity to [false]. log4j: Level value for au.com.noojee.asterfax is [DEBUG]. log4j:

Hi, I would like to customise an end user application like Centiles's callpad software (http://www.centile.com/solutions-applications-callpad.php ). Its purpose is to allow users to set or read various personal phone-related parameters (call history, voicemail settings, conference, ...) instead of using phone keys combinations. Are you aware of any software that could be used for this ?

hy Hope everyone is fine, I have one issue coming in asterisk , What i am doing is i am generating a callback if some one calls at a specif access number on asterisk, Asterisk sends a busy signal to the calling party that he received a request from party and then sends the call back to the person from where asterisk received a request but in From field as you can see below astrisk is sending the

Hi, Has anyone met any success, installing localized (ie non-english) menus within SIP firmware enabled Cisco 7941 ? Those phones seem to be trying to download localized menus from Cisco Call Manager but as they are managed by an Asterisk server, I'm looking for a workaround. Any advice ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

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

Hello, Beside Cisco 79x1-GE, I'm not aware of any Gigabit SIP Phone. Did I miss something ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070612/b9b701b3/attachment.htm

Hi, Is anyone aware of a SIP hardphone with Busy Lamp Fields supporting 2 colors (or more) ? This could be very useful to support extended presence, for instance. Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090519/0b8f1b62/attachment.htm

2009/3/17 Olivier <oza-4h07 at myamail.com> > > > 2009/3/16 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I'm rather new to this domain so I may be doing stupid things without >> being concious of that. >> >> I've got a Patton MATA I'm trying to setup as T.38 fax adapter. >> Whenever I connect a fax machine (Dell

Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090627/37b93684/attachment.htm

Hi, Let's say you have an Asterisk server running. Which parameters would you check to improve service continuity ? I was thinking of : - telco lines status (make sure every is up) - registered hardphones - config files backup (compare live and saved configuration files, if files differ, notifies the administration team) - systems variables (disk and CPU) - log files (trigger an alarm for

Hi, I feel I've read a thread about this previously but I couldn't find it. Is there way for an Asterisk server to check if a sip phone is forwarded without bothering phone's user ? I was thinking of some Alert-Info option that would let the phone reply with a 302 Moved Temporarily or 182 Queued message and not let the phone ring or display anything on its screen. So that, you could

Hello, LLDP is more and more available on various network elements (endpoint, switches, ...). It seems to ease network configuration. Do you have any experience with it ? How would you rate LLDP ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20091124/fce6307c/attachment.htm

2009/12/4 Olivier <oza-4h07 at myamail.com> > Hello, > > Has someone successfully used this QUEUE_VARIABLES() function (in > 1.6.2-rc7) ? > I tried to use it as I'm using SIPPEER() but without success. > > A previous question about it remainded unanswered ( > http://thread.gmane.org/gmane.comp.telephony.pbx.asterisk.user/224466). > > Regards > How can

2009/1/27 Olivier <oza-4h07 at myamail.com> > > 2009/1/27 Olivier <oza-4h07 at myamail.com> > > Hi, >> >> I carefully followed instructions in README file lasting with : >> /root/register >> ... blabla >> asterisk -r >> CLI> restart now >> >> Then asterisk -r fails with : >> # asterisk -r >> Asterisk

Hi, I need to replace digital handsets in offices where there cabling is appareantly not Ethernet-compliant. Today's usage is to press a key to toggle between private ou public line before issuing an outgoing call. Are you aware of a DECT handset (to overcome cabling limitations) that mimic this line-key behaviour ? For instance, acceptable behaviours would be to dial number string and press

2018-06-05 15:27 GMT+02:00 George Joseph <gjoseph at digium.com>: Thank you very much, George for replying. > > > On Tue, Jun 5, 2018 at 3:35 AM Olivier <oza.4h07 at gmail.com> wrote: > >> Hi, >> >> After a long discussion with a friend, I would like to ask here: >> >> 1.According SIP RFCs, is possible/recommended to have different values in

Hi, In exim4, I can see lines such as : mainlog.9:2008-03-12 08:53:28 1JZLmC-0000E7-0A <= root at foo.com U=root P=local S=43802 id=Asterisk-0-123413860-4174-2662 at ipbx-bs-60200 In my voicemail.conf, I see : ; If you need to have an external program, i.e. /usr/bin/myapp called when a ;externnotify=/usr/bin/myapp ; If you need to have an external program, i.e. /usr/bin/myapp called when a