similar to: Say who is using the PSTN?

Hi all, I'm playing with Asterisk and I've already configured all needed .conf files. It works quite well, but now I need your help to tune the system: when I place a call from a softphone to the PSTN, I can't hear directly Telco's tones and I can't use its services, e.g. a mobile's answering machine. I don't know if I have to modify the dialplan or if it depends on my

hi ppl :D my configuration is as follows, i run (let's call it machine 2) debian etch 4.0 and asterisk 1.2, i use voiceone (www.voiceone.it) as an interface to manage asterisk, I have a grandstream/handytone 486 as a sip device, no PSTN line or anything like that all SIP only. I have a machine (machine 1), which functions as my router and machine 2 and sip device are behind it, grandstream box

Hi, I have some really disturbing problems with Asterisk 1.4.1 and my dialplan for outgoing calls. First of all i switched some weeks ago from * 1.2 (bristuffed version ) to this version and in my opinion a lot more troubles arose.... For outgoing calls I use a Digium B410P with chan_misdn (before a Junghanns QuadBRI with zap). 1) So first thing is, that a user reports to me (highly

I am using VoiceOne http://voiceone.it/ as my management interface. I am not 100% sure when it started, but my CDR is now full of "s" as the DST instead of the actual dialed number. As I understand it - it is because it is being recorded in the CDR while in a macro (as below). Is there any work around so that I can record the actual dialed number? [macro-dialout] exten =

I got two phones on an ATA-186 (601, 602) and two phones on the TDM22B (603, 604). I have two lines on the TDM22B. I cannot figure out some of the problems: 1. 601 dials via ZAP/3-1 to local phone number at PSTN: ringing pickup on PSTN (empty) still ringing in the phone set 601 2. call from PSTN back: 601 picks up ... everything works !!! No caller id shows up 3. For testing I have only one

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

I can dial from 601 to a public number. The public number rings. I pickup and hear nothing, while on 601 it keeps ringing. (BTW, is it right to say "ringing" on the active phone?) The *CLI> doesn't show me anything useful: Executing Macro("SIP/601-8238", "dial-pstn|88097880|") in new stack Executing SetGlobalVar("SIP/601-8238",

Hi. I'm dialling two numbers - one that's unobtainable, one that's busy. ${DIALSTATUS} is coming back ANSWER each time right before the channels hang up. Am using the following dialplan macro to dial out. [macro-advdial] exten => s,1,Dial(${ARG1},20,g) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status

SIP SECURITY WARNING Version: v1-0 (cvs today) Problem: sip context in general section ignored - goes to default - allowing unauthorized sip devices to place calls in default context Fix [workaround]: Remove or rename "default" context in extensions.conf Notes: I am not sure what other asterisk functionality may be affected by this - review your other config

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

I think you will find the functionality you are looking for is in * already. Here is an excerpt from the sample extensions.conf file that is included with the source: exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds maximum exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER) exten

Dear All I want to integrate sugarcrm and asterisk , so when customer call the call center the agent or extension which answers the call , before pickup the phone and talk to customer , view his/her information if it is available. I do this as described below : 1-Setup login username for sugarcrm for each extension 2-Extension Users will login to the sugarcrm. 3-Develop php script to handle new

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten =>

Hello I used to have an extension like this which worked fine with asterisk 1.0.7 I first dial to see if an IAX phone is present, if not I would try on SIP instead exten=s,1,Dial(IAX2/iax${ARG3},20,tr) ; 20sec timeout exten=s,2,Goto(s-${DIALSTATUS},1) ; Default action exten=s,200,DBget(temp=CFBS/${ARG1}) ; Get CFBS key, if not existing, goto 301

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 =>

SetUp: - Asterisk behind a NAT, - Red Hat 9.0 - Asterisk 1.2.14 My Asterisk box is behind a NAT and I have a DiD from an ITSP. I have my dial plan set up so that when outside callers dial the DiD, the call is answered by my auto-attendant. The caller can then select who they'd like to speak to and the call is transferred to the external line associated with that person (usually a mobile

What would it do if you exten => h,1,ResetCDR(w) exten => h,2,NoCDR() exten => h,3,DEADAGI(get-unqiueid.php) I have not tried it but in theory it should write the first CDR and then kill the write of the second NO ANSWER CDR. Let me know if it works for you as I may need to do it on some of my h exten code as well. Bryant ---------------------------------------- From:

Hello List! We have problems setting the right caller id on outgoing calls. The Asterisk Pbx is located in Bucarest(Romania), our Telco provider is rcs-rds.ro. We have the local telefon number 40787 00-99, associated to our PRI E1 Line. Where 00-99 are the DID numbers available. The telco is aspecting a 3 digit long Callerid from us, for example like "710", for the extension 10.

Hi All! Let me explain the problem. When using the Originate? command from the manager api, the dialstatus variable returns results? for whichever phone picks up first, and in this case it is the IAX/2? connection. It doesn't matter if Zap/G2/XXXXXXX is set as the channel,? or an extension either. What I am ultimately trying to do is get the? dialstatus of the Zap/X/XXXXXXX channel, i.e.,

I have recently built my first asterisk system and am very impressed with its capabilities. However, I have run into one problem that hopefully someone can help me with. I am trying to use the DIALSTATUS function to route incoming calls to the appropriate Voice Mail (busy or unavailable) or to an Unavailable Number recording if the number is not assigned. However, I find that DIALSTATUS