similar to: Voicemail Question...help

Hi all: Does somebody know how many meters of cable is supported between Asterisk Digium-FXO cards and the analog telephone ? Thanks, Angel

Hi Matt, I have verified with ztmonitor the audio level and it was too low, then with this the fax machine report "Not Response". I modified the audio level in zapata.conf and after that the fax machine report "Commnunication Error". Do you an idea what could be ? Thanks, Angel. > Message: 3 > Date: Sat, 18 Sep 2004 00:48:23 +1200 > From:

Hi, Is there a way to send and receive SMS over SIP protocol with Asterisk ? I mean, between two SIP phones like below... SIP_phone "A" (sending sms) ====> Asterisk ========>SIP_phone "B" (receiving sms) ... Is it possible ? If so, how could I do it ? Thanks, Angel. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list, Time to time, my asterisk goes down.Verifying with TOP, I see the swap memory of the computer get used totally but, I don't see what the process is using it. Hereis a copy wath I see doing top. Does somebody have an idea ? My asterisk version is ====>>> Asterisk CVS-HEAD-08/18/04-22:30:24 Thanks Angel. 08:49:19 up 5:23, 1 user, load average: 0.50, 0.70, 0.64 35

I ran in to an almost impossible ability to run MeetMe on our poweredge. Our PE is a 4 proc with OHCI and we had no need to buy any PCI cards other than for running MeetMe. If you know how to recompile your kernel, recompile it and make RTC a module (its compiled in by default). Then you can use zaprtc as your timing source. Be sure to `rmmod rtc` before you `insmod zaprtc` because zaprtc is a

A better explanation can be found here... http://www.digium.com/index.php?menu=faq#TDM%20&%20Analog_0 > -----Original Message----- > From: Benjamin on Asterisk Mailing Lists > [mailto:benjk.on.asterisk.ml@gmail.com] > Sent: Monday, September 27, 2004 11:37 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [Asterisk-Users] FXO question

Hi, I have to connect 30 phone lines to my asterisk server, can somebody help on how I have to do it ? I have a TDM405P and one TDM400P with 4 FXO ports. Do I have to use 8 TDM400P ? Or, is there another way to do it ? Thanks, Angel.

I've spent the past several weeks reading up and playing around with Asterisk while I've been waiting for an ISDN card I got on ebay to arrive so I can really get to business. I'd just like to run my project ideaa by some of you to hopefully get a little feedback. I aplogize if this ends up being a somewhat long message. In the Marine Corps we've somewhat recently started using

Hi! I am trying to use hping to chek the latency of our network. Somehow things are not going to plan and I thought someone might be able to shed some light on the subject. Here is the setup: (the IP addresses gvien here are fake, but they do represent the correct state of the networking setup) vlan interface IP mask V2 eth0 192.168.20.20 32

hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg -vvvv and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf ------------------ fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf ---------------- context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in

Does anyone know the mechanism by which companies like YouMail (and MNO's using their own voicemail system) are able to redirect ALL calls from a ALL subscribers to *just one* voicemail DID, yet determine WHICH subscriber did the redirection? I had always assumed this was simply done using RDNIS. In other words, the original calling party's CallerID is passed with the redirected

Hi, I am a new user of Wine, and I am having trouble running the programme 'Accu-Chek Compass'. It is a programme for diabetics, that downloads data from a meter, and then graphs it in all sorts of ways. The Programme installs correctly, but when I try to run it, it just flashes on the screen for a split second, and exits. Any help would be nice.... Plod.

I'm having a problem redirecting to voicemail. This may be an asterisk bug I'm not sure, can somebody confirm? Network layout GATEWAY - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h connected to a PRI line. (Additionally patched with http://bugs.digium.com/view.php?id=2687) PROXY - Ser version: ser 0.9.3 (i386/freebsd) FEATURE - Asterisk 1.0.8-BRIstuffed-0.2.0-RC8h handling voicemail.

Hello, I'm having a problem with significant audio dropouts occurring in voicemail messages left via an ISDN-BRI trunk. Dropout durations are as short as 15ms and as long as 200-300ms. The audio that is recorded, appears to be otherwise complete, just with frequent holes punched in it. The same trunk has no problems with audio files played toward it from voicemail, nor interacting with

Hi all, In order to do loadbalancing between my two *, i wanted to stock all things concerning voicemail on a NFS partition... I see that the voicemail system put his files onto two differents directories : /var/spool/asterisk/voicemail/mycontext etc. and /var/lib/asterisk/voicemail/mycontext etc. I've two questions : Why ? and how can i do to centralize the destination of the messages AND

Hello list. The VoiceMailMain()-application has an advanced menu. Can I get a Voicemail-application that has less functionality ? I only want the user to listen to new voicemail-messages (and delete them), not the functionality with the folders and redirecting messages to other mailboxes... I've looked at the code in /usr/src/asterisk-1.4.30/apps/app_voicemail.c but it seems complicated

Hi, whenever I include a "Ringing" Line in some Voicemail Extension I get an error when a call from the outside (via ISDN) comes in, but it works when an internal (SIP-phone) calls the extension. Here is my configuration for testing: ------------extensions.conf------------ [isdnext] ; strep external "101", our number and leave only extension exten =>

I'm having a bit of a problem here: I have a * box with a fritz isdn card (running capi 2.0 and chan_capi) and a x100p card for testing purposes. As a proof of concept, I wanted to be able to dial into the * using the isdn line, listen to a message, and enter a 3 digit extension number. If this happens, I wanted the * box to dial out using the x100p card, into our PBX (Nortel Meridian). If

Hello, I wonder if it is configurable possible to add a new menu demo to run within voicemail context dialogue! I want to run am interactive menu before or within the normal voicemail dialogue to run a script based on the subscriber selection. My point is to get use from the authenticated password that provided by user to go though his voicemail to access this new feature as well! If not, is there

chek this out how can i fix this pple? http://channels.debian.net/paste/2475 http://channels.debian.net/paste/2476 pls i need help -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://dovecot.org/pipermail/dovecot/attachments/20060503/266fdab0/attachment-0001.html>