similar to: SIP Outdial Not Detecting Ringing Line

I am originating a call to a Local channel using an Originate Action: Action: Originate Channel: Local/dial at outdial Context: outdial Exten: answer Priority: 1 Timeout: 45000 ActionID: some_id In my dialplan, I have this: [outdial] exten => dial,1,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten =>

Using Asterisk 1.6.1.14 and dahdi 2.2.0.2+2.2.0. We're placing outbound calls over an analog line. Some of these calls are going to cell phones that play music rather than providing a standard ring. As a result, the Dial command sometimes returns a DIALSTATUS of CHANUNAVAIL and sometimes it returns BUSY. The problem is that this is happening on calls that are being answered. Has anyone else

I've set up an outbound .call system for customer callbacks and the like. Calls are going out over analog lines and I'm trying to use the WaitForSilence routine to make sure the phone has stopped ringing before starting message playback. The problem is that if I set the first argument of WaitForSilence to anything other than 1, WaitForSilence never exits. Some general info on my setup:

I would like to use * as a standalone voicemail system. As such I need it to be able to outdial a certain extension for MWI-ON and another extension for MWI-OFF Is there anyway to get * to automatically dial an extension when a voicemail is left and another extension when the mailbox is cleared? Thanks -------------- next part -------------- An HTML attachment was

Hello, I am working on an outdial project and the Asterisk box is connected behind a PBX via SIP. When a call from the Asterisk box is routed out over the PRI attached to the PBX I am not getting proper call progress. The PBX is indicating that the call is answered while it is still ringing at the far end. Does anyone have any suggestions on how I should go about waiting for a variable number

Hi all, Just thought I'd pass along some pointers when outdialing with Voicetronix's OpenLine4 card. I was having a tough time dialing out from *, it probably has something to do with chan_vpb.c not waiting to hear the dialtone before telling the card to dial. A quick fix was to insert a "," in the dialstring telling the card to pause before dialing. However when the

Hi all, Does anyone have an application/script or extensions.conf file which will do the following? "When a new VoiceMail is left for a user, the asterisk system will place a call to a cellphone/pstn number(via some provider). When the user answers his cell/home phone, comedian mail will ask for his password and he can check his Asterisk VM?" Anyone have any examples of it

I have successfully built and made asterisk talk SIP extension to SIP extension, read all the docs, and about 1000 emails from the archive. The trunk side of Asterisk, from the docs perspective, is a smidgin TDM-centric, Analogue, T1, zaptel.conf etc..... Asterisk cares not about the externally presented digits as the telco KNOWS which time-slot or analogue line the call came from I live in an

Hello, we have a PRI (E1) to a carrier and a second one to a legacy PBX: DTAG ---pri---- * ------ Hicmo (PSTN) | | Sip and more Many normal inbound calls are direcly routed to the hicom. Outbound calls from the Hicom go through LCR and then to PSTN. Inbound faxes are working, but outbound faxes from hicom to pstn are

I am trying to implement a Centralized Call Waiting System. I have red some document about asterisk group features to manage group and category of a sip channel. I have done a lot of test about it but always it doesn't work correctly if I transfer the call. This is the macro code I use for inbound calls. [macro-test] ; ${ARG1} - technology something like SIP ; ${ARG2} - resource.

Hi All, I'm trying to configure a Digium T100P to talk to a legacy voicemail system. I have the signaling specs verbatim from the original manufacturer documentation as follows: [T1 Signaling] Service Type: T1,D4 format, AMI(Super Fram) Signaling: Four wire, terminated, E&M (Robbed bit) Start Protocol: Wink start; 250msec duration Dial Tone: Enabled Digits: DTMF, 4-digits DTMF: 50msec

Is there a way to limit the number of simultaneous outbound SIP calls made by Asterisk? We've tried using the 'Asterisk sip call-limit' parameter but that doesn't seem to be working and one of our engineers says that parameter has been depreciated. Thanks, Deric.Page at nisc.coop -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm using the Dial application in the extensions file with the G option to execute an AGI script after the Dial (I need to track the call status) as follows: exten => _X.,3, Dial({DIAL_STRING},,G(_X.^4)) exten => _X.,4, Answer() exten => _X.,5,AGI,agiScript.php The problem is that testing between two internal phones (with two ATA) I loose the audio when I include the G option in the

Hi, I am using phpagi for agi scripting. and want to update callerid number but didn't get any success. please help me how to update PHPAGI is new for me. Below is the code which I write. #!/usr/bin/php -q <?php set_time_limit(30); //require(.phpagi.php.); include("phpagi.php"); $agi = new AGI(); //answer the call $agi-> answer();

Is it possible to place multiple phone numbers in a single outbound .call file? If I try doing this, only the last phone number in the file is called. However, if I use 1 file per phone number, then Asterisk attempts to process all generated CALL files at once, incrementing the retry count for each that cannot be called because the designated channel is busy. For example if I have a list of 10

Hi all, * was running ... I have a WT405P and an AVM C4 with chan_capi 0.3.4a Today chan_capi stopped working, without any changings at the system. It seems, that not * is the reason, because isdn-log also shows no calls. If I try to call * from outside via capi, I only get a busy. That is the try from inside to outside: stern01*CLI> -- data = @89930:0107901723168212 -- capi

Hi, I'm developping some AGI in perl (5.8.6) on i386 using Asterisk 1.0.5. I want to get some variables such as DIALSTATUS and ANSWEREDTIME after a $AGI->exec("Dial", "dial_string"); but here is what i get actually: DIALSTATUS= DIALEDTIME=ANSWER ANSWEREDTIME=18 I searched the archives and saw that $AGI->verbose could mess the access to variables, but I don't use

Sorry if this hit the list twice.. but I didn't see it come through: Hey guys, Just for archival purposes, I have setup the Intellitouch ITC-3002 (2006) SIP phones for use with asterisk (1.2.7.1). After a few "gotcha's", I was able to do transfer's, moh's, push a button to check voicemail, callerID, etc.. One big gotacha was the dial timeout (by default!) is set to

hi, i have connected a SNOM 200 to the asterisk. here are my settings, Codecs ------- Default codec - g.711u/g.711a Packet size - 20ms Negotiation - Interoperable Type - 160 DTMF ---- Inband - negotiate Outband - negotiate Payload Type - 101 when a call comes to the SNOM or when making an outdial, following warning messages are coming on asteisk, WARNING[1209214400]: File dsp.c, Line 1198

I've noticed I have an issue with my Dialplan ... apparently instead of a busy signal when the caller is busy it falls through and gets a Congestion... What's the proper syntax for this, reorder tone when there is a reorder and busy when there is a busy... SBC is a T1/PRI. [macro-sbc-outdial] exten => s,1,Dial(${ARG1}/${ARG2}) exten => s,2,Congestion exten =>