similar to: lookup a anonymous internal caller

We have a small number of phones, when a call comes in we want all the phones that aren't in use to ring. Is there a simple way to test and see what phones are in use then ring the other phones? I tried some code like this: [zap] exten => s,1,Answer exten => s,2,ChanIsAvail(${DERRICK}) exten => s,3,SetVar,"EVERYONE=${DERRICK}" exten => s,4,ChanIsAvail(${DON}) exten

I have in my sip.conf the following [jon.moore] type=friend mailbox=8100,8150 In voicemail.conf, both mailboxes are defined. On my Aastra 480i phone, I only see the first mailbox listed. I've verified this, by changing mailbox= to reverse the order, and I then see 8150 when I go to Services > Voicemail on the phone. I also only get MWI events for whichever mailbox is listed

Hi everybody, How can we add new contexts in asterisk realtime module? All I could figure out after googling is that a new context HAS to be declared in extensions.conf with 'switch => Realtime/@<databasetable>' under the context name declaration. This works fine as long as we are adding extensions only to this one context, but doesn't give the freedom to add new contexts for

Hi I have two small question, if you can help me ;=) Problems with Answer+Music my extension: [Cal-In] exten => _811XXXX20,1,Goto(C-Internal,100,1) exten => _811XXXX21,1,Goto(C-Internal,200,1) [C-Phibee] exten => 100,1,Ringing exten => 100,2,Wait,1 exten => 100,3,Answer exten => 100,4,Dial(SIP/201&SIP/200,30) exten => 100,5,Hangup exten =>

I am trying to implement a follow-me script (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) but I am having a brain fart as I haven't a clue where to get started with what to do with this. From my main menu, I want the extension 300 to execute the script as follows: exten => 300,1,dial(sip/200,20) exten => 300,2,playback(pls-wait-connect-call) exten =>

Hi all, I am having difficulty getting incoming PSTN calls working. I have set up an account with a third party provider. In my system, the user register with SER and use Asterisk for PSTN access, voicemail etc My provider told me to change my sip.conf as follows register => username:password@sip.blueface.ie/2093 ; To receive incoming calls specify this block and

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First of all great job on diax. I downloaded it and tried it, could not connect, got an authentication rejected,but I have not had a chance to figure out why yet - tried with a working gnophone setup in the configuration files. Is there any way to pass command line arguements to the program ? Where I see a real niche for a lightweight softphone is being able to serve the thing from a

Hi Guys, Running Asterisk v1.4.26 (Elastix flavour) and have Aastra 9480i, 6757i, and 6730i, but none of them indicate the voic-email. Where should I look for trouble to find the root issue for MWI? Thanks, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20100714/d268d1f8/attachment.htm

I need to know how to setup the data side of the T1 on my Linux Box. I have found information about configuring a PRI and HDLC but nothing about the Frame-Relay type setup for data. The following is information from our T1 provider. Network T1: Framing = ESF Line code = B8ZS Build out = 0-133ft(DSX)/0dB(CSU) Clock = network Pulse-density-enforce = off alarm-option = on alarm-delay = 15

Hi, Yes InternalExtension is the context and 2093 the extension. Just to explain something odd that?s happening (and I?m very stumped with this) .I think my contexts are definately the reason that I can?t interrupt the menu for incoming pstn calls to choose a submenu: My users register with my sip proxy (SER). Therefore when I create an entry for them in sip.conf I set only one context. Also to

I'm trying to connect my Asterisk server via sip using my vonage soft phone account. Has any anyone successfully got to work? I get error from asterisk saying: == Parsing '/etc/asterisk/sip.conf': == Parsing '/etc/asterisk/sip.conf': Found Aug 24 11:01:11 WARNING[1125329600]: acl.c:146 ast_get_ip: Unable to lookup '216.115.25.199:5061' when trying to register with

Hi, How do I disabled asterisk to use database and storage voicemail in directory. Im getting the below error [Dec 3 19:08:53] WARNING[4934]: app_voicemail.c:3430 inboxcount: Failed to obtain database object for 'asterisk'! [Dec 3 19:08:55] WARNING[5129]: app_voicemail.c:2353 last_message_index: Failed to obtain database object for 'asterisk'! [Dec 3 19:09:04]

On June 18, 2019 12:34:35 PM PDT, Mathieu Chouquet-Stringer <m+syslinux at thi.eu.com> wrote: > Hello Hans Peter, > >Any idea on how I could debug this problem? Basically lpxelinux 6.03 >reboots while loading the initrd if its size is above a certain >threshold. It only happens on certain servers and there's not output >when it happens. I can trigger it reliably on

I'm simply trying to forward calls to users who have the call forwarding feature enabled (FWD Status and FWD Ph Number kept in the astDB). The problem is that I want users to be able to forward calls to numbers that they would normally be allowed to dial within their own context. (I don't want a local call only person forwarding to a long dist number, for example.) I'm able to

Dear, I want to make an application to calculate quantile within a For() I tried the following without success: ej. date p_val <- matrix(sample(10, 1000, replace=TRUE), 200,5) test 1 rr <- paste("p_val$",names(p_val[1]), sep="") quant <- quantile(rr, probs = c(0, 10, 20, 30, 40, 50, 60, 70, 80, 90, 100)/100, na.rm=FALSE, type=1) test 2 rr <-

Hi all; I need to create ASCII files as output from R and I'm using sink(), cat(), and paste() for this. My problem is that the ASCII files hace several columns, and I would like to know if intermediate columns (the second one for example) could be alineated to the right. My values are integer ranging from 1 to 1000. Thanks a lot. Best regards, Javier

When I booted with anonymous enablings, I saw ... NOTICE: enabling probe 8 (syscall:::return) NOTICE: enabling probe 9 (syscall:::return) ... Then when I claimed the data, dtrace said dtrace: error on enabled probe ID 607 (ID 148: syscall::lwp_park:return): invalid address (0xfe67a000) in action #1 dtrace: error on enabled probe ID 607 (ID 148: syscall::lwp_park:return): invalid

Hi, I am running 2.2.5 and I would like to know if the "restrict anonymous" as been implemented correctly, as it was supposed to behave from the start, in order to deny ALL anonymous connections as stated in the man : "When restrict anonymous is yes, all anonymous connections are denied no matter what they are for." Ive been reading some dev mailing lists and someone said

I'm trying to set up anonymous ftp using vsftp. I want to be able to allow uploads. The anonymous id is userid "ftp". The ~ftp/ directory is actually owned by root, but ~ftp/pub is owned by ftp. Here's the vsftpd.conf: anonymous_enable=YES anon_upload_enable=YES local_enable=NO write_enable=NO local_umask=022 dirmessage_enable=YES