similar to: Which Codec(s) to use..?

Hey Everyone, I am using NuFone for 866 inbound service and I am trying to figure out the callerid part of it. Any call into my * system just shows "Toll Free Call" and will not give me the calling party's caller ID info. Is this just something I have to live with using NuFOne, or did I miss some type of config in * that will grab the callerID other than the inbound 866 number...?

Hi Everyone, I read through the list on the issues with the ztdummy driver which I need for MeetMe, but I seem to have come across a problem that I cannot seem to find an answer for. I am running Gentoo 2.6.10 on an Intel box. I have read the the wiki entries on the ztdummy and followed the instructions as they relate to the 2.6 kernel. Everything compiled great, but a modprobe ztdummy

Hi Everyone - I have webvmail up and running, I can see the messages, forward them, pretty much everything but listen to them. Here is what I see in my logs: 192.168.0.1 - - [25/Feb/2005:08:15:40 -0800] "GET /vmail/vmail.cgi?action=audio&folder=INBOX&mailbox=2377&context=default &password=000012&msgid=0000&format=gsm&dontcasheme=4624.gsm HTTP/1.1" 200 9438

Hey Everyone, I downloaded and installed the X-Lite softphone the other day (the lite version) and cannot seem to get it to work well. Don't get me wrong, it registers with my asterisk server and everything seems to work well, except the call quality really is horrible. I thought it may be the place I was trying it at (DSL) so I took it to the office and tried it right next to the asterisk

Hi Everyone - I just got my hands on a Cisco 7970 and was told that I should do a factory reset before trying to configure it to work with Asterisk. After the factory reset, it will not boot at all, instead sits with the line button lights flashing one at a time in sequence. I have had no luck trying to figure it out - anyone run into the same problem that can lend a hand..? Thanks

Hey Everyone - I was going to create a visio diagram outlining how my extensions will flow out. I was just wondering if anyone on the list may have an example they have already done up so I can get some ideas. Thanks ****************************************** Richard J. Sears Vice President American Internet Services

Hey Everyone, This is not an operational issue, and to my knowledge only effects the look of the command, but when I issue a "sip reload" then a "sip show peers" I see all of the actual usernames I have assigned in my sip.conf. However, five minutes later I reissue the sip show peers and all of the usernames have disappeared and are replaced by the sip ID. The only way to get

I have a [start] context that all my inbound and '0' calls are routed into. Because of the way I want to set my system up, I want to prompt the user to enter a 1 if they know the extension, or a 2 for a directory and nothing else. It works, however there is a 5 to 10 second delay after enter the 1 or 2 before the system responds. I have read over the wiki on how asterisk handles digit

Have Asterisk us at running fine, but have run into a small snag. It's not emailing the voicemails to the users. I have attach=yes set, sendmail is configured and works from from the commandline (sent an email to myself). Unless I'm wrong, or missing something, asterisk is configured by default to send an email when a users receives a voicemail, correct? Thanx A

*This message was transferred with a trial version of CommuniGate(tm) Pro* In setting up Asterisk, I'm looking to dump my current phone system (Nortel Venture). I presently have three POTS lines. I would use a VOIP provider, but now are presently available in the Toronto, ON, CANADA area that support user owned hardware/software. I need a 416/647 area code number. In looking at FXO cards

Does the voicetronix card work with Asterisk? -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of woody+asterisk@solutionsfirst.com.au Sent: Monday, February 02, 2004 11:06 PM To: asterisk-users@lists.digium.com Subject: RE: [Asterisk-Users] Pictures of new multiport FXO/FXS from digum > -----Original Message-----

Richard, I feel a little stupid now. Our spam filter (GWAVA) was blocking the emails because I had WAV files in the block list. One of those things that doesn't occur to you until you've had a little bit of sleep. Thanx for the help! A *-------------------------------- Date: Fri, 25 Mar 2005 06:03:42 -0800 From: "Richard J. Sears" <rsears@americanIS.net> Subject: Re:

Hi, I would like to use a digum card to call an external number through my PSTN. I think that I have a problem in the configuration. Asterisk returns me app_dial.c:764 dial_exec: Unable to create channel of type 'Zap' I use Fedora core 3. I installed libpri, zaptel and asterisk. I plugged my line on the FXS module (green part). I make modprobe zaptel && modprobe wctdm without

hello, I will do a server to do a lots of conferences (MeetMe). I want to know that if I dont use a digum card, the limit of simultaneous calls is harder without a card than with a card ?if, yes, how harder is the limit? thank you Cordialement, BERGANZ Fran?ois P Pensez ? l'Environnement, n'imprimez ce mail que si n?cessaire. -------------- next part --------------

Hello, Avaya has a nice feature that allows you to a) ring both a cellphone and a desktop phone at the same time b) transfer calls (and access other PBX features) from the cellphone that recieved the call, as long as the call is bridged through the PBX c) while talking on the cellphone, pick up the handset on your desktop phone and the call is automatically moved there, hanging up the cellphone

Hi all, i'm trying to make AMD, Answer Machine Detection, to work on my outbound context but i can't get it to work, just on inbound context like whe i use the application Answer before AMD, but i need to make AMD to do the detection on an outbound predictive dialer integration. Follow are the inbound and outbound examples. My current environment is Asterisk 1.4beta3 and a Digum

I am trying to do a wire level tap on T1 equipment using digum equipment. So far most call monitoring hardware for call centers try to stay on the analog side requiring a lot of rewiring. I have already posted to the list about T1 "bridging" using DAC's support in the zaptel drivers. I still don't know if I can spy on channel information since I don't have any digium

Hey all, I'm having problems where there is significant static when making SIP -> PSTN calls. SIP -> SIP and SIP -> VM calls are totally clear and fine. Here's the setup: Polycom 601,501, and ten 301s. Digum 2400 TDM card w/echo cancelling, 12 FXO ports. The TDM card is on IRQ 5 with nothing else on it. Server Specs: Asus P4P800E Deluxe P4 3.0 Ghz 1 GB Ram 80 GB SATA HD -

hi all, we do have some troubles with zaptel timing source - we have a setup with 3 telco PRI lines, connected to asterisk digium card 0 - asterisk does some handling - calls are leaving on digium card 1 - going to a siemens hipath - there is some call handling - some of the calls then are going from the hipath over a qsig line to a bosch integral PBX - handling the rest of the calls. To be able

I have about a dozen SIP phones up and working, now I want to connect the asterisk box to our Fujitsu 9600 PBX. I currently have two dial-up servers conencted to the Fujitsu PBX that I built with mgetty/pppd and have the lines provisioned the same way as those dial-up server, ESF, B8ZS, and E&M wink start, so I have confidence in the guys who set up the PBX. I've built a loop back plug