similar to: context of a transfer

Hi, Running Asterisk CVS-Head latest on a Dual P3 800 1Gb Ram. For some odd reason now that I have the asterisk box almost to the stage I want it, I hit a problem. I have a te405p in the system, Zap/g1 is connected to the telco as an ISDN 30, Zap/g4 is connected by ISDN Primary Rate to an Ericcson BP250 phone system. When calls come in on g1 they go straight through instantaneously to the

Vinicius Fontes wrote: > Hey guys, I don't know if this is the right place to ask this. I was > thinking about reporting a bug, but maybe it's better to sort out if > this is really a bug or just me being lame. > > I want to record *every* call in my Asterisk box, so I use the > MixMonitor() application like this is my extensions.conf: > > exten =>

my biggest concern about defaulting the context to anything at all besides [default] is that you then have to remember to configure the voicemail.conf with the corresponding contexts. as it stands, you have the ability to do just that, but you don't have to. if you have several hundred extensions broken out by dozens of contexts, it might not make sense to force the voicemail.conf to follow

Yo all, As there has been some intrest, here's my updated version: I post it to "-dev" as well as "-users", as it may be of intrest to both. Inspired by the example in the tips & tricks-section of "http://www.junghanns.net/asterisk/", I built a more elaborate set of features. Currently, my implementation supports call- forward unconditional, on no answer

Does anyone kmow what these errors mean or how they can be fixed. I'm using asterisk on a Fedora Core 2 box with a TDM400P with 2 fxo and 2 fxs ports. Dec 29 17:17:52 WARNING[6019]: chan_zap.c:5469 ss_thread: Channel Zap/4-1 in prering state, but I have nothing to do. Terminating simple switch, should be restarted by the actual ring. -- Hungup 'Zap/4-1' == Starting post

I am using version: CVS-v1-0-12/13/04-18:46:23 with a TDM400p (2 fxo, 2 fxs ports) and I keep getting errors along with phantom calls: Dec 21 16:02:07 NOTICE[5872]: chan_zap.c:5363 ss_thread: Got event 17 (Polarity Reversal)... Dec 21 16:02:14 WARNING[5872]: chan_zap.c:5434 ss_thread: CallerID returned with error on channel 'Zap/4-1' my analog phone reads caller ID info fine when

I am attempting to implement the new features added recently where you can have "Goto(s-DIALSTATUS)" in the dial plan. My extensions.conf looks like this: exten => s,1,Dial(${ARG2},20,r) exten => s,2,Goto(s-${DIALSTATUS}) exten => s-NOANSWER,1,Voicemail(u${ARG1}) exten => s-CHANUNAVAIL,1,Voicemail(b${ARG1}) exten => s-BUSY,1,Voicemail(b${ARG1}) exten =>

When trasferring a call, how is the context determined? When using a zap device, and the DTMF code for blind or attended transfer is entered, does the tranfer originate at the context the zap device is set to be in, or does it originate from where the outside call being transferred originated in, or the context the current call is in? I ask because I am seeing strange behavior when trying to

Guess I got where your confusiion lies.... you DON'T set up the ALL contexts the users will have acces to in sip.conf, in this file you will only set 1 single context. All other contexts you want the users to have acces to, you will have to add them in the extensions.conf So what you have to do is the following: -user 2092, set it the createmenu context in sip .conf - in extensions.conf

Hi, I'm trying to use reshape but I cannot quite understand how it works. Could somebody help me on this? Example, my data is something like: mydat <- data.frame(tree= 1:10, serra=rep(1:2, c(5,5)), bt01= 101:110, bt02= 201:210, bt03= 301:310, mm01= 9101:9110, mm02= 9201:9210, mm03= 9301:9310) > mydat tree serra bt01 bt02 bt03 mm01 mm02 mm03 1 1 1 101 201 301 9101 9201

Hi All, I have a senario where we have multiple locations and all have the ability to call using 1NXXXXXXXXX pattern, so we have created multiple contexts so the outbound goes fine, but while transfer occurs (after picking the inbound call and transfer), it uses the first 1Nxxxxxxxxx priority patterned context, like if the 3rd location is making a transfer, but 1st location have the priority

> > As a result, I just go back to put all users in [default] in > voicemail.conf. > > Am I missing anything? >> What do those contexts mean in your setup (beside being arbitrary >> groups)? I just want to group the mailboxes by say department rather than putting them all under [default]. So, I could use CLI to "voicemail show users for sales" or

Hi there, I have setup Asterisk with a couple of Sipura SPA-841's and Grandstream ATA's. The problem is that with both of these devices the Unattended call transfer process seems to be just like Attended but instead you hang up as soon as you have dialled the number of the party your are transferring to. The call transfer all works fine BUT as you complete your side of the transfer

Wrong list. asterisk-dev is for changing the C source code of Asterisk. I don't think AGI's "count" or are considered for inclusion into the subversion repository as stated by one of your conditions for payment. On Wed, 29 Apr 2009, Alistair Cunningham wrote: > I'd like to offer a bounty for a feature for Asterisk where an AGI > program can park and retrieve calls

I am running asterisk 1.0.1 with OH323 compiled in. We have a 323 trunk to CallManager with a mgcp controlled pri router. When using sip phones (directly registered with asterisk) to call out the 323 trunnk to PSTN, calls timeout after 3 rings. If I answer b4 3 rings - no problem, otherwise I get "no one is available to answer at this time" on the consoel and it redirects to an

Hi If I have ser sending calls to asterisk, is there a way to get a different block called in sip.conf for each call (based on some variable, NOT username, From:), if not and they all hit one block which has contect=abc, then when that context is called/matched in extensions.conf, how can I have diff features for various groups of users. EG lets say I have a large company with 4 departments

I have a dial plan that includes a company phone directory as a main menu option. If they just sit at the main menu, after 20 seconds, they are transferred to the operator. If the user picks an extension from the directory, they are transferred to the proper extension. If the called number is not available, they are transferred into VoiceMailMain. They leave a message, and hang up. The hang

Hi! If anyone is interested, these are my first tiny steps with RSpec on Rails: https://www.heise.de:444/svn/ctvdr/ctvdrwebadmin/trunk/ The "ctvdrwebadmin" application should become a web administration tool for a Debian based distribution that is specialized to VDR [1] - the Linux video disc recoder by Klaus Schmidinger. It shall provide an interface for the installation /

Hi all, is there an easy way to get a 'copy' of a type living in another context into the local context? Background: when calling a function residing in a different module (context2) from a module (context1), we first need to introduce a function declaration of the function with empty body. However, in order to do so, we need the function type. pFuncInContext2->getType gives us the

Hello All, I'm having an issue where asterisk doesn't hear any audio after transferring to voicemail. Here is the dial plan and console output. DIAL PLAN [voicepulse-in] exten => _14259491337,1,NoOp(Incoming call from VoicePulse) exten => _14259491337,2,Ringing exten => _14259491337,3,Wait(1) exten => _14259491337,4,Dial(SIP/1337,20) exten => _14259491337,5,VoiceMail(1337)