similar to: vm notification no longer contains calling party

Hello everybody. Is there a way how to setup asterisk to notify caller's phone? Example: I have some numbers and names in asterisk database ( cidname, cidnum), and I want to display the name of person on my phone ( which has no addressbook, but can display chars ) which I am calling to be sure that I have dialed the right number. Thank you for any answer. Andrej

I am using Sipura 841 phones and Asterisk CVS-v1-0-06/14/05. Whenever a call comes in with blank CIDName or CIDNum the phone reports the respective variable as "asterisk". I can manually set the variables to whatever I want: CIDName (alpha-numeric) & CIDNum (Numeric). But if I try to make them blank, or null, or maybe throw some alpha characters into CIDNum, they get reported

Hello, Sometimes, when a call comes in from the PSTN through our VoIP gateway, the information that is sent to our web page that logs calls includes the original CID name instead of the one that is we expect to be rewritten on the fly using Asterisk's LookupCIDName: ================= ;extensions.conf [internal] exten => group,1,LookupCIDName exten =>

Hello, In case there are other users of the AsteriskWin32 port... I haven't really used the AGI feature of Asterisk to run an application from extensions.conf. *Win32 supports Perl, which I don't know. Apparently, it's also possible to write AGI applications as EXE's (there's a eagi-test.exe file installed by default). => When a call comes in, I'd like an AGI

Hello When a call comes in, I'd like to fork a Python script that broadcasts a message so that users see the CID name + number pop up on their computer screen, and simultaneously ring their phones. The following script doesn't work as planned: It waits until the script ends before moving on to the next step, which is Dial(): =========== exten =>

Hi, I am attempting to connect to the blacklist database using PHPAgi and it always seems to hang. The code snippet I am trying is: $r = $agi->get_variable("CALLERID(num)"); $cidnum = $r["data"]; if ($cidnum < 1000000000) # No valid callerid. { exit(0); } $r = $agi->database_get("blacklist", "$cidnum"); if ($r["result"] ==

All; I am running Asterisk 11.6-cert16 and I have voicemail setup so voicemail messages are sent as email attachments. That works fine. However, the body of the email contains the CallerID(name), but is missing the CallerID(num). For example, the email body looks like this: Just wanted to let you know you were just left a 0:21 long message (number 13) in mailbox 101 from WIRELESS

Greetings, 2 nights ago I upgraded one of my remote servers to the latest CVS Stable, Asterisk CVS-v1-0-02/07/05-16:13:48, and my inbound and outbound caller ID stopped working. At first I thought it was my carrier, so I've been yelling at them for the past day or so. I then upgraded my core network (about 8 asterisk servers) to the latest stable as of 3am last night, Asterisk

Thank you very much. For some reason "emailsubject" was not included in my example config. Well, it's working great now. Last question, I promise :P. Is it possible to change the date format? I want it in Norwegian. -----Opprinnelig melding----- Fra: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] P? vegne av Barry Flanagan Sendt: 22.

Hi! I'm having a ZyXEL P2000W that I'm using together with my Asterisk box (CVS from some week ago). When I get a call directly to the *-box (lars@hostaname.domain.se) I see on the console that Asterisk get the calling users name as CIDName and his SIP-address as CIDNum - but at the P2000W i only get "asterisk" as CLIP? And I can see in the P2000W that "asterisk" is

I have found that I can make the phones display any one word on this second line by adding a fromuser=<word> in sip.conf. This really isn't good enough though. When you look at the received calls or missed calls directory, each item has two lines, the first is the CID name, and the 2nd is supposed to be the CID number. However, if it is asterisk, or some other word, when you hit the

> > Now I have two phones connect to my hardware PBX,and want to Make two calls from within Asterisk and switch them together. I now have the two numbers and the other parameter should how to set. for example: the value of data, type and format ,I set the type "Local" and type AST_FORMAT_SLINEAR but I don't know it is write. and the data is don't know how to set. struct

Why Asterisk showing (on SCCP and H323 phones) different CID related to type of Incoming channel: If incoming channel is SIP, on phone is displayed CALLERIDNUM If incoming channel is ZAP, on phone is displayes CALLERIDNAME It vas very frustrating! I lost couple hours of my time to find that my dialplan is not faulty, but asterisk is!

/**************************************************************************** * * * Always Be Conferencing (ABC) * * * * Creator: chris @ Penguin PBX Solutions * *

I am looking for the most efficient way to replace all occurrences of NaN in a data frame with NA. I can do this with a double loop, but it seems that there should be a higher level and more efficient way. With is.na, I could use ifelse, but if.nan seems not to have similar capabilities. TIA, Jon Peck Jon K. Peck jkpeck@aya.yale.edu [[alternative HTML version deleted]]

Full_Name: Robert Peck Version: 2.8.1 OS: XP Home Submission from: (NULL) (71.168.114.196) There are some packages that I cannot install on my PC. Example: LearnBayes The following output is sent back: > utils:::menuInstallPkgs() trying URL 'http://www.ibiblio.org/pub/languages/R/CRAN/bin/windows/contrib/2.8/LearnBayes_2.0.zip' Content type 'application/zip' length 379068

Hello, I don't know how to put this on words, so maybe an example is better: On a PDC: zoolook at venkman:~$ smbclient -L localhost -U% Domain=[BENSA] OS=[Unix] Server=[Samba 3.4.7] Sharename Type Comment --------- ---- ------- print$ Disk Printer Drivers IPC$ IPC IPC Service (venkman server (Samba,

I've been working on ELF object support for the MicroBlaze backend and found that ELFObjectWriter assumes the x86/x86-64 architecture. Attached is a patch that makes the 16-bit e_machine value in the ELF header configurable by the target backend. Right now the target backend simply passes the 16-bit value that it would like to use in the ELF header. I have considered a second approach where

On Fri, Nov 12, 2010 at 6:01 PM, Wesley Peck <peckw at wesleypeck.com> wrote: > Will this poking include converting the JIT to use the MC framework? That is the motivation for me poking at things, yes. > I've added an MC based asm parser, disassembler, and code emitter recently to the MBlaze backend and it would be nice to get JIT support automatically :) I agree, that would be

Is there a bug in Firefly (3rdparty) wherein it does not show caller ID? I am using SetCIDNum(12345) before I dial my firefly (IAX2) phone... no caller ID. CallerID is passed properly to other clients. -A.