similar to: If voice mail not found dialplan

I've got a setup where we have 100 DID's. Our default dialplan has one line that calls a macro: exten => _22XX,1,Macro(STDEXT,${EXTEN}) The macro is pretty basic: [macro-STDEXT] exten => s,1,NoOp exten => s,2,Dial(SIP/${ARG1},15,Tt) exten => s,3,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Voicemail(${ARG1}|u) exten => s-NOANSWER,n,Hangup exten =>

Running Asterisk 1.2.8 kernel 2.6.13.4-1. Everything in my dialplan seems to be working well except for one problem. When calls are blind transferred to an invalid extension I would like the call to go to the operator on ext 1000? What is the best way to do this? Thanks in advance Here's a snippet of my extensions.conf [default] exten=>_10XX,1,Macro(stdexten,${EXTEN},SIP/${EXTEN})

hi ppl, i'm using asterisk 1.2 because i'm making use of voiceone, but before i was using asterisk 1.4 and had the same problem, it concerns an italian voip/sip provider called eutelia/skypho, my problem is the following one: when i start my pbx my skypho account is working fine, meaning that e.g. incoming calls are shown in the asterisk CLI and caller and callee can hear each other when

Hi folks, i'm trying to configure my extensions.conf as small as posible and for that reason i'm using macros. The problem is that maybe I have a misunderstood the concept for the directive "mailbox" in sip.conf. Under my knowledge configuring the mailbox directive to the mailbox I want would be enought to leave an retreive messages in that voicemail box. Of course it seems to

Hey all I have been beating on this all weekend long. Any feed back would be appreciated. We stood up a 11.6 system. We tested everything we could think of. We moved over to it and all seemed to be working good than a customer told us that they were not hearing our vociemail greetings. When we call into the system and it drops to voicemail we just get a beep no greeting played. We checked

When I set CONNECTEDLINE() info for an incoming ISDN call, the calling party sees only CONNECTEDLINE(num) and the name does not get displayed. Some time ago I called a number, where I did get back a name and a number and everything was displayed correctly. So I think the calling site should basically be able to handle all connected line info. Looking at a pcap trace of the D-channel data, I

Hello, Is there any equivalent of the CONNECTEDLINE function which can be called from an application using the AMI? Thanks for any ideas. -- David Cunningham, Voisonics http://voisonics.com/ USA: +1 213 221 1092 Australia: +61 (0) 2 8063 9019 -------------- next part -------------- An HTML attachment was scrubbed... URL:

>>> But if you've already got the caller on the phone, then you might consider the CONNECTEDLINE function in Asterisk... And that we don't. It's the third party that would like the notification the the destination phone is currently busy with another call. CONNECTEDLINE only functions after a channel has been answered. I was successful with using CONNECTEDLINE when issuing

Hi, Previously i was using asterisk 1.4 with freepbx installation. To try the 1.6 version i installd anc configured everything.. Just one thing didnt work so far.. I am using grandstream 2000 and it has a line busy indicator for chef secretary phones. But now, this feature does not work. I can see the line is online..with a green steady light.. But when the line is busy or DND, it wont change to

Hello, I recently gave CONNECTEDLINE function a try on an Asterisk 11 setup with a couple of SIP phones. When a SIP phone dials an other one, with a CONNECTEDLINE statement in its dialplan, I noticed that Asterisk update caller's information using a Remote-Party-ID header in 180 Ringing message. For instance: Alice ----------------> Asterisk ------------------->Bob ------- INVITE

Gentlemen, Did tryout "CONNECTEDLINE" function, was exactly what I have been looking for. But there are at least one thing I cant figure out. Did a very simple and "stupid" extension 0317998955 and ran a test. My phone (0317998975) dials 955, the display on my phone changes from "955" to "Connected Line 955" when my call is answered, shouldn't the

Everybody, I've had a request from my manager that I figure out how to get our Asterisk 13.x system using chan_sip to be able to display on the Polycom VVX series phone display (firmware 5.9.5), when an extension is called and the person on the other end is on the phone. He said, "Our old Analog phone system could do it, how hard can it be?" I've gone down the path of trying

First I wanted to thank both Marc Schwartz Greg Snow and for their reply. Then I needed to add a level of complexity to the problem. I would be able to create the biggest possible matrix. In other way does it exist a method to ask smthing like the following : max number of rows for a matrix if column=x? Thank you ------------------------------------------------------ Passa a Infostrada.

Hello! Just read http://www.voip-info.org/wiki/view/Asterisk+func+CONNECTEDLINE tried on dahdi, it works, i.e. if I call asterisk user from my pbx connected phone I see what I set in Set(CONNECTEDLINE(name)= But if I call the same user over h323 ( no matter is it asterisk with ooh323 or cisco gateway) I don't see this. Could you tell me is it possible? Thank you!

I have been racking my brain attempting to get the remote callerid information for calls made to extensions on another Asterisk system connected via IAX2 but nothing has worked. To clarify, I would like to display the number AND name on the calling phone when calling extensions on another Asterisk system. I seem to be able to 'send' all the information I want to the system I am calling but

Hi All, Does anyone knows what doest this output means? [root@serverxen ~]# xm list Name                                      ID Mem(MiB) VCPUs State   Time(s) Domain-0                                   0     3202     8 r-----   5220.1 vm1                                  3     4095     2 -b----   3529.2 vm2                                  5     8191     4 -b----    399.0 [root@serverxen ~]# xm

Hello, I have installed an samba ad1 and an samba ad2 with replication. On the dc1 "samba-tool drs showrepl" say "... was successful" On dc2 there is the same. When dc1 goes down I get an Error (on my client): Active Directory Users and Computers error: "server is not operational" What is wrong here? For my understand that is no Replication, when one goes down and

Hi, i have the same problem. Trying to use the dialplan function CONNECTEDLINE() this way Set(CONNECTEDLINE(name)=${SIPPEER(${EXTEN},callerid_name)}) Set(CONNECTEDLINE(num)=${EXTEN}) ends with [Jul 1 16:30:50] ERROR[3954]: pbx.c:2902 ast_func_write: Function CONNECTEDLINE not registered Same happens trying function CALLEDID. I am using Asterisk 1.6.1.20. What do i

HI, My name is tarun and iam new to this list.. Can any one please help me ?? How can i kill a rsync.. Suppose if iam in middle of rsync of a whole 10gb of data. Can i kill rsync in the middle using kill "rsync process id" and resume it later, wouldnt it result in any errors if we kill rsync suddenly. Or is there any signal to send to rsync to terminate it cleanly..Doest rsync have

On our tests using asterisk, some calls have been terminated abruptely with exact 185 seconds. This is happening with all our incoming calls from a trunk from 1 of my DID providers ( other providers or trunks are fine) and I could reproduce it by calling a queue from my Wengophone Softphone and letting the MoH play for 185 secs. If I make the same call from my WRTP54G on the same place,