search for: outdialed

First off, I also posted this on the digium forums so if anyone here also reads those, sorry for the cross-post. When I place an outbound call using SIP to my cell phone, asterisk immediately starts processing the dialplan without waiting for the call to be answered. We could handle this on DAHDI using callprogress, but I don't know of a similar setting for SIP. Here is the contents of

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 =>

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

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

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

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();

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

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

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 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

I have been trying to get a DIALSTATUS output from a call started with originate. I searched a fair bit and have found several references to using local channels to do this. However, I could not find enough of the specifics to get it working myself. What I need to do is dial a zap channel and run various scripts if the channel is answered, busy, no-answer,etc. Here is the dial plan I am

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 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

GoIAX, the Asterisk community's free IAX provider, is offering free US dids now. I loaded about 175 dids in and put up a very beta sign in page. Unfortunately I had to restrict the free us/canada outbound calling back down to toll-free only. There was a lot of war dialing and prank calling going on. I'm working on some stuff to hopefully curb that kind of stuff down so I can

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:

Hi all, I've got some questions to post in regard to running asterisk in a production-grade environment, specifically targeting high-density IVR applications. No VoIP involved, just straight PSTN -> * and perhaps the occasional outdials or agent-based predictive dialing. 1) Which user would you run * under? 2) What other security-related issues do you have to resolve? 3) How do you handle

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 =>

Hi, I'm looking fore a way to play a dial tone before our IVR platform answered the phone line. I want to use for the following reason: When a caller calls our Voice Platform, the call will direct dial out to a number. I want to dial out before the inbound call is answered. But now the inbound call here's nothing. When the outdial call is picked the inbound call will here

...a panasonic globalrange phone) which always sends a fully qualified phone number. That is, for a local Canadian number, even if I key in 6135551212 it actually sends to asterisk 01116135551212. This means of course, along with "normal" phones I end up having twice as many extensions for outdialed numbers. Is there any way I could canonicalize this down to the more normal NXXNXXXXXX format before I process through all of my dialplan rules? Effectively it means being able to alter ${EXTEN}. Is this doable in any way? b.