search for: outdial

...nbound] type=friend context=ivr host=dynamic [sip_outbound] username=XXXX fromuser=XXXX secret=XXXX fromdomain=X.X.X.X type=friend host=X.X.X.X canreinvite=no disallow=all allow=ilbc allow=g729 allow=gsm allow=g723 allow=ulaw call-limit=8 extensions.conf: [outdial] exten => dial,1,NoOp(Dial String = ${DIAL_STRING}) exten => dial,n,NoOp(Dial Timeout = ${DIAL_TIMEOUT}) exten => dial,n,Dial(${DIAL_STRING}, ${DIAL_TIMEOUT}) exten => dial,n,NoOp(Dial Status = ${DIALSTATUS}) exten => dial,n,Agi(agi://localhost/Outdial.agi) exten => di...

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 => dial,n,Agi(agi://localhost/Outdial.agi, $...

...9;Zap/14-1' -- Starting simple switch on 'Zap/62-1' -- Accepting overlap call from '595457' to '034491****' on channel 31, span 2 -- Executing SetVar("Zap/62-1", "Out=Zap/g1/") in new stack -- Executing Goto("Zap/62-1", "OutDial-LCR|BYEXTENSION|1") in new stack -- Goto (OutDial-LCR,034491***,1) -- Executing SetVar("Zap/62-1", "LCR=01081") in new stack -- Executing Goto("Zap/62-1", "OutDial-Dial|BYEXTENSION|1") in new stack -- Goto (OutDial-Dial,034491****,1)...

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 dia...

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 sc...

...;,"SIP/00918885268942 at voipon,60,r"); ?> And CLI> == Using SIP RTP CoS mark 5 -- Executing [101 at outbound:1] Answer("SIP/2209-000026d3", "") in new stack -- Executing [101 at outbound:2] AGI("SIP/2209-000026d3", "/home/virendra.bhati/outdial.php") in new stack -- Launched AGI Script /home/virendra.bhati/outdial.php <SIP/2209-000026d3>AGI Tx >> agi_request: /home/virendra.bhati/outdial.php <SIP/2209-000026d3>AGI Tx >> agi_channel: SIP/2209-000026d3 <SIP/2209-000026d3>AGI Tx >> agi_language: e...

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 su...

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

...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 using: [outdialer] exten => 100,1,Dial(ZAP/4/1234567) exten => 100,n,DeadAgi(notdeadyet.py) exten => 100,n,Hangup [dialerplan] exten => s,1,AGI(showstatus.py|${DIALSTATUS}) exten => s,n,Hangup Here are the manager commands I am using: Action: login...

...= 0x101 Info = 0x0 == received CONNECT_CONF PLCI = 0x101 INFO = 0 -- DISCONNECT_IND ID=003 #0x0002 LEN=0014 Controller/PLCI/NCCI = 0x101 Reason = 0x3302 == DISCONNECT_IND PLCI=0x101 REASON=0x3302 == Spawn extension (OutDial-Dial, 01723168212, 2) exited non-zero on 'SIP/ePfd-7515' -- data = @89930:01079h -- capi request omsn = @89930 == found capi with omsn = 89930 == CAPI Call CAPI[contr1/89930]/3 == CAPI Call CAPI[contr1/89930]/3 -- CONNECT_CONF ID=003 #0x000e LEN=0014 Controller/PLCI/NCCI...

...ing 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 duration Outdial senderized: Yes [CO Trunk specific order] DID/DNIS/Phone number group directed to the T1 [Line Hunting] On in-bound calls the T1 must be configured to do a linear (not circular) hunt for trunks from 0 to 23. (Try 0 first, then hunt up to 1,2,3,etc) The voicemail server hunts from trunk 23 down to...

...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 unrestrict outdial again, but this is the problem with free.. there is always someone that will abuse it. If anybody has any ideas on how to keep the abuse down let me know. The best ideas I have now is to only allow a certain amount of calling per month, add velocity checking, and somehow put some accountabilit...

...onsole/dsp ; Console interface for demo IAXINFO=guest ; IAXtel username/password TRUNK=DAHDI/G0 ; Trunk interface TRUNKMSD=1 ; MSD digits to strip (usually 1 or 0) [outdial] exten => s,1,Set(TIMEOUT(response)=3) exten => s,n,WaitForSilence(4000,1,16) exten => s,n,Agi(agi://localhost/Outdial.agi) exten => s,n,Hangup() exten => failed,1,Verbose(Outdial failed) exten => failed,n,Verbose(Reason= ${REASON}) exten => failed,n,Hangup() **** Thanks in a...

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 crashes (murphy -will- visit you some day)? 4) What are the best redundancy techniques to use? 5) With respect to Digium's E1 card, what...

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 (ast_dsp_process): Unable to detect process 2 frames WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_process): Unable to detect process 2 frames WARNING[1209214400]: File dsp.c, Line 1198 (ast_dsp_pro...

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 => s,102,Playback(noservice) exten => s,103,Congestion

...l 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 something. This is confusing voor in inbound call, therefore I want to let the inbound call to here a dial tone. I already tried the function Ringing() and Playtones(). (before the inbound call is answered) Can anybody give me a hint or solution t...

...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.