similar to: VoiceMail Outdial?

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

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

Hello, I'm trying to supervise an existing Voicemail box with a BLF button on Debian's asterisk 13.14.1 system. I mostly found this [1] document. I added in a context a line like: exten = *7000,hint,MWI:31 at default With "core show hints", I can read this: *7000 at subs : MWI:31 at default State:Unavailable Presence:not_set Watchers 1 My questions

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

> -----Original Message----- > From: JR Richardson [mailto:jmr.richardson@gmail.com] > Sent: Saturday, June 17, 2006 2:30 PM > To: asterisk-users@lists.digium.com; Douglas Garstang > Subject: Voicemail with NFS (working, I think) > > I'm using a stand-alone VM server and exporting the VM files ro for > MWI function only. All my registration servers mount the remote

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, 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, 1. How do you then, synced then unread message presence with custom device status ? From an external program ? When a user leaves VoiceMailMan application ? Using externnotify ? 2. What is MWI:101 at default expression for (see [2] ? Cheers [2] https://wiki.freepbx.org/display/FPG/Subscribe+a+BLF+button+to+Monitor+a+Voicemail+Box 2017-11-21 17:58 GMT+01:00 John Kiniston <johnkiniston at

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have Asterisk sitting between the PSTN and a legacy PBX. Asterisk is doing some IVR work prior to forwarding calls to the PBX and it also acts as the voice mail server for the PBX, with Asterisk configured for IMAP storage. When a call comes in and the caller leaves a voice mail, the VoiceMail application calls the program configured in

Dear List, I'm having a blinking MWI light on the snom 320 even when there's no message waiting in Asterisk. We've managed to make the voicemail button work using fromdomain=192.168.0.1 in sip.conf vmexten=2500 (our VoicemailMain application extension in extensions.conf). We also added notifymimetype=application/simple-message-summary also in sip.conf to allow SIP simple MWI

I have the voicemail setup working in that I get the MWI and it emails the message correctly. When I pressed the MWI button on my SNOM 200, it dials into the voicemail system and prompts me for a mailbox and password. I know there is a way to automatically connect directly into the mailbox via the extension.conf file, but I can not find the documentation I am looking for in reference to variables

I am shopping around for an IAX provider that provides both outbound minutes and an inbound DID for a flat fee. I have been looking at TELIAX. http://teliax.com/ Does anyone have any experience with them? What codecs do they support? Inband or 2833 dtmf? Voice quality and reliability? Any comments and/or other recommendations would be appreciated. Thanks

I am pretty sure that there are no IAX providers that offer CallerID name but wanted to double check with the list in case something has changed recently. Is anyone aware of an IAX provider that offers incoming CallerID name? Is there a technical limitation within IAX which is preventing IAX providers from offering CallerID Name? Why is no one offering this? -------------- next part

Can i register a cisco 7902G with asterisk????? what i have to do??? thanks in advance. R -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20041130/12bf3c06/attachment.htm

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

Any more ideas on my below mail? If a user is registered with SER and leaves a voicemail message with asterisk (by using rewritehostport etc in ser.cfg), then how is the user supposed to listen to the message afterwards? Is there any other way other than the MWI method?? Thnaksm Aisling. ---- Original Message ---- From: ashling.odriscoll@cit.ie To: asterisk-users@lists.digium.com Subject: FW:

Howdy I have a question regarding support for picking up when the telco sends a MWI message. My client's setup is a small office with three incoming lines on a TDM400P with iaxy's and a Grandstream as extensions. I am using CVS Head from yesterday. (I was resolving a different issue.) Since they only have two voice lines, with the third as a fax, I am using voicemail from the telco.

Hello ppl, Am using realtime odbc storage for voicemail, sip users/peers, static for extensions and so on. My issue is I am not getting MWI for any fones, even tho I've got rtcachefriends=yes in sip.conf WIth tcpdump, I always see the NOTIFY going as Messages-Waiting:.no Voice-Message:.0/0.(0/0) even tho there are legitimate voicemails in the INBOX path for that particular users in the