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Hi; I don't know what these things are called so that I can look for an OS solution. I want to install an email server interface like Hotmail where people can check their email TTW. What is this called? Any recommendations? TIA, Susan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachments/20100115/f4362274/attachment-0001.html>

Hi; I have an email form that worked fine until now. For some reason, if I send an email to an email address at a domain that I control, I can receive the email TTW no problem. However, if I try and push it to, for example, this gmail account, I never get it. It's not even in the spam filter. What could this be? TIA, Susan -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.centos.org/pipermail/centos/attachment...

...|SIP/413") in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Set("Local/413@default-0edc,2", "DYNAMIC_FEATURES=automon") in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Executing Dial("Local/413@default-0edc,2", "SIP/413|20|Ttw") in new stack Apr 5 12:38:24 VERBOSE[22757] logger.c: -- Called 413 Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Macro("Local/510@default-6b6c,2", "stdexten|510|SIP/510") in new stack Apr 5 12:38:24 VERBOSE[22758] logger.c: -- Executing Set("L...

...|SIP/413") in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Set("Local/413@default-2959,2", "DYNAMIC_FEATURES=automon") in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Executing Dial("Local/413@default-2959,2", "SIP/413|20|Ttw") in new stack Apr 5 14:02:42 VERBOSE[23365] logger.c: -- Called 413 Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Macro("Local/510@default-25e1,2", "stdexten|510|SIP/510") in new stack Apr 5 14:02:42 VERBOSE[23366] logger.c: -- Executing Set("L...

...xx agent => 3002,3002,Exxxx agent => 3003,3003,Axxxxs agent => 3004,3004,Arxx agent => 3005,3005,Lxxx agent => 3006,3006,Loxxx *************************** **********modules.incl************ [macro-stdexten] exten => s,1,Set(DYNAMIC_FEATURES=automon) exten => s,2,Dial(${ARG2},20,Ttw) exten => s,3,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}@default) exten => s-NOANSWER,2,Playback(thank-you-for-calling&goodbye) exten => s-NOANSWER,3,Hangup exten => s-BUSY,1,Voicemail(b${ARG1}@default) exten => s-BUSY,2,Playback(thank-you-for-calling&amp...

...lude extensions-home.incl #include menu.incl [globals] OUTBOUNDTRUNK=Zap/g1 PSTN1=Zap/1 PSTN2=Zap/2 PSTN3=Zap/5 PSTN4=Zap/6 PHONE1=Zap/3 PHONE2=Zap/4 *********************** macros.incl *************** [macro-stdexten] exten => s,1,Set(DYNAMIC_FEATURES=automon) exten => s,2,Dial(${ARG2},20,Ttw) exten => s,3,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(u${ARG1}@default) exten => s-NOANSWER,2,Playback(thank-you-for-calling&goodbye) exten => s-NOANSWER,3,Hangup exten => s-BUSY,1,Voicemail(b${ARG1}@default) exten => s-BUSY,2,Playback(thank-you-for-calling&amp...

...terisk Dial command.) It seems however that either Asterisk or Alcatel drop the call prematurely (it's more likely to be on the Asterisk side). What could I try? The Asterisk log displays (* ext is 4053; Alcatel ext is 5900): -- Executing Dial("SIP/4053-08311988","Zap/g1/5900||tTW") in new stack -- Requested transfer capability: 0x00 - SPEECH -- Called g1/5900 -- Zap/2-1 is proceeding passing it to SIP/4053-08311988 -- Zap/2-1 is ringing -- Zap/2-1 is busy -- Hungup 'Zap/2-1' == Everyone is busy/congested at this time (1:1/0/0) -- Executing Hangup("SIP/4053...

...Bom, agora vamos ao pulo do gato, esse passo ? muito importante pois ? ele quem verifica se existe ou n?o o siga-me para o ramal. Vamos ao contexto: [disca] exten => _3XXX,1,Noop(CF/${EXTEN}) exten => _3XXX,2,Set(siga=${DB(CF/${EXTEN})}) exten => _3XXX,3,Dial(SIP/${siga},30,Ttw) exten => _3XXX,4,Dial(SIP/${EXTEN}) ; Unconditional forward exten => _3XXX,5,Hangup -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110610/662f280b/attachment.htm>

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue

...switched to kernel 2.6.23 with the MG2 echo canceller (nothing else changed). Each time I try to establish a call on the PRI line I get a congestion signal. in /var/log/asterisk/full: Feb 20 08:09:43 VERBOSE[10657] logger.c: -- Executing Dial("SIP/4445-082e8340", "Zap/g1/3666||tTW") in new stack Feb 20 08:09:43 NOTICE[10657] app_dial.c: Unable to create channel of type 'Zap' (cause 34 - Circuit/channel congestion) Feb 20 08:09:43 VERBOSE[10657] logger.c: == Everyone is busy/congested at this time (1:0/1/0) At first I thought that the new kernel version could...

...before Please also find a detailed description of the package including example code here: https://journals.plos.org/plosone/article?id=10.1371/journal.pone.0239237 Any comments, suggestions or bug reports are welcome on https://github.com/MiKatt/openSTARS All the best, Mira -- Dr. Mira Kattwinkel Quantitative Landscape Ecology iES Landau, Institute for Environmental Sciences University of Koblenz-Landau Fortstra?e 7 76829 Landau Germany Phone: + 49 6341 280-31553 Office: Building I, Room 2.02 _______________________________________________ R-packages mailing list R-packages at r-proje...