similar to: Asterisk 1.6.1.x SMDI MWI w/Fujitsu F9600 Problem

We're in the process of replacing an ancient Centigram-based, but Fujitsu-labelled, voicemail system with an Asterisk solution. The system will interface with a Fujitsu F9600 switch and use the SMDI module in Asterisk 1.6.1.x to communicate the calling information needed to make the interplay between the two systems work smoothly. This portion of the project is configured and working fine, as

I am the one that placed the bounty. After it being there for 2 months and getting no takers (and very few if any people asking about it), we are almost finished writing it in house. I'll keep the bounty up untill we do finish our product so if anyone beats us to getting it working they'll get paid... W. Kevin Hunt CCIE #11841 MCSE, Linux+ SME www.huntbrothers.com >

Hello: I've been wrestling with an integration issue for which there also seems to be one piece missing. I'm hoping someone on this list can help. Due to a complete lack of cooperation from our current voice provider we are in need of an alternative way to access our Octel 350 voicemail system from our SIP (SER) proxy. I can redirect and relay calls to numerous destinations via

Hi, I'm a brand newbie to asterisk trying to set it up for the first time and I can't get a softphone to connect, the connection times out. I had a trixbox pro install working, but I need more control and would like to learn to do it with asterisk. In /var/log/asterisk/messages I see: WARNING[17401] res_smdi.c: No SMDI interfaces are available to listen on, not starting SMDI

Hello! I'm integrating an Asterisk-based voicemail system with an old switch, and I want the call history from SMDI. My understanding is that the terminal number in the SMDI message matches the channel's trunk number. >From within an Asterisk app, how do I get the trunk number? Thanks! Nate

Last question for the day, I promise. On voip-info.org and trixbox.org, I found some old threads on MWI via SMDI. Has this been rolled into Tbox or has anyone successfully rolled it in after the fact. As part of our longterm plan, I'd like to move the legacy PBX to Tbox and pass MWI back to it via SMDI, like the current system. As we drop the extensions on the legacy side and move to all IP

Does anyone know if Asterisk can function as a voicemail system for a Nortel Option 11 PBX? We will be connecting Asterisk to act as an IVR before sending calls to the Nortel and as a Voicemail system in case the user does not answer. That part is trivial, the only problem we have is that the customer wants the Voicemail light on the Nortel phones to light up when a user gets a new message.

An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040331/c2abf19f/attachment.htm -------------- next part -------------- Hello, Is there any work in progress for supporting SMDI in Asterisk ? if Not, could anyone tell how to get started implementing it for Asterisk. Regards, Tony

Greetings. I am attempting to configure a system based on Asterisk 1.2.3 to be used as a backup should our aging voice mail/auto attendant system fail, which seems increasingly likely given its advanced years. The first part of this task is getting the auto attendant feature to work correctly, which I would have figured to be relatively easy. I have successfully built a menu structure, but cannot

Does anyone know what would be involved in making Asterisk work as a voicemail system in a Centrex environment? We have a Centrigram voicemail system that belongs in the Smithsonian. There are analog lines coming into the box and a 56KB data feed from the phone company's switch. Peter

We have a customer with an obsolete Centigram voicemail system who would like to replace it with Asterisk. Any one with experience doing this or information on the signalling and trunking used to connect the Mitel SX-2000 to the Centigram server? -- George Pajari (dCAP), netVOICE communications 604 484 VOIP(8647) x102 www.netvoice.ca www.ip-centrex.ca www.digium.ca

This is what I would like to do... exten =>7299,1,Voicemail(u8896) exten =>7299,2,Dial(zap/g1/#641299) exten =>7299,3,Hangup What I would expect to happen is... Incoming call is answered by voicemail... Voicemail app finishes and the next priority starts. This is where the problem lies... After the intial call is ended the process is ended and the voicemail notification string

Steve Blair writes > I can redirect and relay calls to numerous destinations via >SER but because the Octel needs an SMDI interface for mailbox >identification I am stuck, none of the solutions thus far support >SMDI-SIP munging. > > I just started thinking about the possibility of using Asterisk >with a few FXS cards to provide the gateway between SIP and >the Octel.

Greetings, The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of this beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug fixes, the following new features have been added since beta4: * The SMDI interface in Asterisk has been reworked to fix a number of issues as well as add some new features. SMDI message information is now accessed in the

Greetings, The Asterisk.org development team has released Asterisk 1.6.0-beta5. As of this beta of 1.6.0, 1.6.0 is now feature frozen. In addition to a number of bug fixes, the following new features have been added since beta4: * The SMDI interface in Asterisk has been reworked to fix a number of issues as well as add some new features. SMDI message information is now accessed in the

Hi: I deleted old modules in /usr/lib/asterisk/modules before make install. I built zaptel and libpri before asterisk. Modprobe zaptel and modprobe -v wctdm executed witiout complaint. Also, ztcfg shows 4 channels configured, as it should. Starting asterisk produced the output below with several warnings and a failure. Can someone help, please. I double-spaced the warnings in the text below

Hi, Anybody seen this before? (using a pre-compiled asterisk from the OBS on a sles11sp1) (I mean, i did the same with a 1.6 without any problem, but i need 1.8) after starting: kc3004:~ # /usr/sbin/safe_asterisk: line 145: 16133 Segmentation fault (core dumped) nice -n $PRIORITY ${ASTSBINDIR}/asterisk -f ${CLIARGS} ${ASTARGS} > /dev/${TTY} 2>&1 < /dev/${TTY} Asterisk ended with

i belong to india. i am making pbx using sangoma fxo card. i want that when ever call comes to my PSTN line i should see the no from where call is coming. so i have to configures chan_dahdi.conf according to my region. i checked dahdi.conf and in that they have mentioned for india

I'm testing an upgrade of an i686 production machine running 1.4.24.1 to 1.6.0.9. I've installed dahdi-linux-2.1.0.4. But: asterisk -cvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvvv Asterisk 1.6.0.9, Copyright (C) 1999 - 2008 Digium, Inc. and others. Created by Mark Spencer <markster at digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details.

Hello Friends: I've just installed Asterisk 11 on my Linux (debian) server but it is not starting up when trying with "asterisk -vvvvvvvvvvc" and "service asterisk start". Starting process just stop and shows: "Illegal instruction" as final output. Looking at logs I fouind at /var/log/asterisk/messages : [Nov 25 11:09:26] Asterisk 11.6.0 built by root @