similar to: Application possibilities

Hi, as far as I know my problem is not a bug but wanted behaviour. Let's assume the following dialplan: exten => 123,1,Answer exten => 123,n,Dial(DAHDI/g0/00492112233,20,g) [...] exten => 123,n,Hangup I do an dial-in with my SIP-Client (or phone). The Dial-Application starts the outdial and I get connected with my partner. After hangup, only 1 CDR is generated. But I(!) need 2

Bicom Systems releases outCALL, an Asterisk open source Outlook integration LONDON, UK (11th April 2007) - Bicom Systems announced today it has released outCALL, an open source desktop application allowing integration Microsoft Outlook. OutCALL allows users an easy way for placing and receiving phone calls integrated with users Outlook contacts. "The open source PBX market needed

So, I'm trying to get the Asterisk vxml (from i6net) working. Having no luck with it. My dial plan has: exten => _X.,1,Answer() exten => _X.,n,Wait(1) exten => _X.,n,Vxml(file:///tmp/menu.vxml) The /tmp/menu.vxml file has: <?xml version="1.0"?> <vxml version="1.0"> <form> <block><audio

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

Hi OutCALL 1.40 is released. It is available in two flavours: - Without extension authentication - With extension authentication Changelog: OutCALL 1.40 (2007-06-29): - Multi-language support (French-Canada is included in the setup, while the English PO file is distributed with OutCALL setup which can be translated and added into OutCALL in run-time) Please use http://www.poedit.net/ for

I recently upgraded to asterisk 11 from 1.8. I had VXML working via AGI in 1.8 - from extensions.conf: [VXML] exten => s,1,Answer exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})}) exten => s,n,AGI(agi://localhost/url=${ENCODED}) exten => s,n,Hangup Using asterisk 11 on the same host with the same config in extensions.conf: -- Executing [s at VXML:1]

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 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 =>

Hi Everyone, I am using OutCall 1.6 (latest) with Asterisk 1.6 and Windows Vista. I can originate calls see the program login nicely but when a call comes in it only shows the Name portion of the CLID and not the number hence it pulls up a new contact on Outlook. The new contact only show name and last name and no CLID Number again. So, this repeats every-time I call even if I manually enter a

The first release of Voiceglue is now available. Voiceglue provides a VXML interpreter using Asterisk telephony and the OpenVXI VXML parsing suite. It is released under the GPL, and thus compatible with Asterisk and OpenVXI licensing. The first release is available at the project website: http://www.voiceglue.org There is also a mailing list for those interested in continued evolution of

How can I match wildcards inside a GoToIf? I have something like this, but it doesn't work: [default] exten => _2XXXXXXXX,1,Macro(outcall,${EXTEN}) [macro-outcall] exten => s,1,GotoIf($["${ARG1}" = "220408XXX"]?2:3) exten => s,2,Hangup Any ideas? Regards, Ricardo.

Our company is at the point now where we're almost ready to switch over to an Asterisk server for a number of telephony applications. There is one final application I've been trying hard to find to replace something we already use with another provider. It's kind of an advanced "FollowMe" application. (freedomvoice.com) It works as follows: 1. An outside caller calls into

As some of you know I've been trying to facilitate an involvement with www.tellme.com <http://www.tellme.com/> speech recognition tools and Asterisk. See www.studio.tellme.com <http://www.studio.tellme.com/> There have been a number of people who are already integrating the two and utilizing Tellme as an ASP to deliver speech recognition to their asterisk applications.

Hi everyone, Does anyone have the documentation for OutCall? http://code.google.com/p/outcall/ The link isn't working. Thanks Dan ________________________________ IT Maintenance Contract Clients can now access our Instant Chat Service to receive immediate remote IT support. Click the chat link below for support. For more information on receiving IT support from ?150

All, I have a weird situation here and haven't been able to turn up any useful information in searches, so I thought I'd post to the list. Essentially, I have a customer who wants us to forward some of their calls to various cell phones. Normally, I'd use FollowMe() for this (that's how most of our customers are set up.) However, this particular customer has some requirements

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

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'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:

Hello, Does asterisk supports VXML? Couldn't find much resource on that on google and wiki. Thanks Foong

Hello, We're attempting to use Asterisk for distributing calls via SIP in a large-scale speech recognition/VXML environment. We currently use DNS SRV with weights and priorities to instruct VoIP gateways (not Asterisk) to route calls to pools of servers. This works extremely well and provides for load balancing, fail-over, and by setting the TTL low (several minutes) we can easily take