similar to: Help: How does one determine the length of an outbound/dialout MESSAGE to be delivered

Just a comment about the documentation for postfix integration when using receipient delimiter and delivery of user+foo at domain.ext to user at domain: http://wiki.dovecot.org/LDA/Postfix I had some issues to get this working, so thought I'd share my experience. Dovecot version 1.2.rc3. Postfix version postfix-2.7-20090511. I have no local (unix) mailboxes, everything is delivered to

Hello Everyone, Can someone point me to reliable links on how to tweak Asterisk AMD I am calling a number and have to two files to play depending if it is a real person or an answering machine. Most everytime Asterisk calls it thinks it is an Answering Machine and it starts playing the AMD message, instead of the delivering the "1st real message" Any hints? Thanks in advance, -C

Hello All. I am trying to use RAGI the ruby agi framework with trixbox. I am having a problem with the dialout part. The RAGI framework creates a file in the /var/spool/asterisk/outgoing directory and routes the call to an extension (I have listed the relevent portion of the file below). The problem is that the initial dial command does not execute properly in trixbox. I am hoping somebody who

Hi, I want to create a messaging system that recognizes threads of messages, not unlike gmail. So far I have these models: Conversation belongs_to :user has_many :messages Message belongs_to :conversation The problem I am running into is not only does a conversation belong to a user but the conversation also has a receipient user with his/her corresponding conversation. How would I

I try to track down an error that causes that Astcc just reports the time, but not the costs. I could narrow the problem down into this sub routine: sub calccost() { my ($adjconn, $adjcost, $answeredtime, $increment) = @_; eval { my $adjtime = int(($answeredtime + $increment - 1) / $increment) * $increment }; my $cost; print STDERR "Adjusted time is $adjtime, cost is $adjcost with

Hi, I noticed that when dial terminates it does not return to the dialplan, and therefore can not execute any entry after Dial(). Is there any trick to overcome this limitation ? How I am supposed to handle the returned vales DIALEDTIME, ANSWEREDTIME if I can not execute anything after Dial()? I made a workaround with DeadAGI (below) but it is unreliable: if 2 calls end

Hello, I saw that Asterisk don't calcultate fine the ANSWEREDTIME. I want that when ANSWEREDTIME =~ 5.6 become 6 and if =~10.3 become 10 because, now, if ANSEREDTIME =~ 15.9, it become 15! it isn't correct How can I have a rounded ANSWEREDTIME ? Where have I to manipulate the sources? thank you -- Francois -------------- next part -------------- A non-text attachment was scrubbed...

Dear all, I tried to make a call with extensions.conf. exten=> _00[1-9].,1,Dial(zap/g1/${EXTEN}) exten=> _00[1-9].,2,NoOP(ANSWEREDTIME=${ANSWEREDTIME}) exten=> _00[1-9].,102,Hangup But the 2 and 102 will not be executed. So I can get the correct answered time via 2. Is any idea about it? Is it the problem of my ZAP channel's configuration? My zapata.conf is as below:

Hi List, In one of my AGIs (using DeadAGI) I grab the answered time using: my $res = $agi->exec ("DIAL $dialstring"); my $answeredtime = $agi->get_variable ("ANSWEREDTIME"); However this information differs from what's written in the Master.csv file (which happens to be the correct value!) Any ideas why? I'm using asterisk 1.2.7.1 and the

No matter what I do, ${ANSWEREDTIME} is always 0, even on the most simplest dial plan such as: Using Asterisk 1.2.13 exten => 77,1,Answer exten => 77,2,Playback(custom/dax/S300) ; one minute file exten => 77,3,Noop(${ANSWEREDTIME}) exten => 77,4,Hangup -- Executing Answer("SIP/5402-b7b45f58", "") in new stack -- Executing

Hi, I have a dial plan where I need to notify an external system when a call was answered and when the call hung up. In both requests the start time needs to be the same. My Dialplan looks something like this: [outbound] Exten => _X.,1,Dial(SIP/${EXTEN}@1.1.1.1,,U(call-answer-from-carrier)) Exten => h,1,NoOp(ANSWERED_TIME: ${ANSWEREDTIME} >>> DIAL_TIME: ${DIALEDTIME}

If you set "dialout=fromvm" in your voicemail.conf, how do you then go about being able to dial back out? Is there a service feature code?

I've been reading the Linejack strikes again messages, and have another Newbie question is it possible to use a Voip Product as a Dialout line for * ? I have a Vegastream 100 Voip to PRI. box. With * can I use that as a Dialout / dialin box? The Vega100 does either sip or h.323. Thanks. Bradley Greep

Hi, I am having some trouble with ISDN Dialout. Using a Netjet-s PCI Card. When in Minicom, the only way I can dialout is if i issue ATS18=1 First. Otherwise I get a BUSY message. So thats fine. But when I dialout from asterisk, I get an immediate hangup, so my guess is that asterisk is not issuing ATS18=1 to the ttyI device. Here are my configs, any input would be greatly appriciated.

I am trying to find a way to allow dialout from voicemail when connected from an 'internal' extension context, but prevent dialout when connected from an 'external' extension context. As far as I can tell the dialout context that can be set in voicemail has no regard for the context from which the call to voicemail came in. Any ideas on this? Maybe a variable passed when

Hi! I have an automated dialout via a call file to a mobile. Can I detect when the call is not answered but forwarded to the mobile operator voicebox? I would like to stop the dialout if this is the case. TIA, Mike

Hi, When I try to dialout over chan_capi everything works fine when I settle for msn=* in my capi.conf and use the primary msn of my ISDN-line. But trying to configure a different MSN the chan_capi doesn't dial and comes with: No one is available to answer at this time What can be the prob? -- Thanks, Marc aka IzNoGood

Dear all I read on the list back in 2003 that * does not support IXJ LineJACK dialout yet is this still the case? Thanks

Hello my name is Ilija Poznic and I have a problem. My configuration is 1. Digium TDM4000P with one FXS. 2. AVM Fritz ISDN adapter (configured with capi). When I connect to my ISP and then start *. Asterisks is registering me to SIP provider iconnect. After that I can call international call trough VoIP. My problem is that I want to dialout to ISP only when I have a international call.

Dear All, I have "get full variable" AGI call to get the ANSWEREDTIME channel variable. I have originated the call to one extension, once answered I have called DeadAGI to control the call. I have problem that after hangup the call AGI "GET FULL VARIABLE" returns -1 for ANSWEREDTIME channel variable. What is the problem? Where I made wrong. Please suggest me..