similar to: suggested hardware especially sound cards

Hi, It is possible to create a dummy account (SIP or IAX type) in order to be used in a "dummy" extension? I want to be able to use it as a normal extension (as an IP phone connected to it), but without the need to answer or call from that extension. I want that when I call that extension to hear the ring, and after the defined period of time to enter in the Voicemail system. I

To whom it might concern, The Gesko Ikarus 1200S analog telephone has advanced callerid capabilities. When used with an ATA186, it show the username and the phonenumber of the caller. (or whatever you let * tell it) http://www.gesko.be/idgg004.htm Price is 77 euro something and available with Telec. (NL) Met vriendelijke groet, Pauline Middelink -- GPG Key fingerprint = 2D5B 87A7

hi, i have to do a demo with asterisk, unfortunately i don't have yet an x100p card, so i need to use a 56k voice modem on my motherboard... could someone tell me how i can configure asterisk to use this modem to call? thanks a lot for the help!!! Angelo

Has anyone solved the problem on the ADSI phones that when you hit one of the soft keys, the Number Pad stops working?

I've seen this happen a few times and I think it's when the system that Asterisk is running on crashes due to a power failure (or for some other reason that causes a non-planned shutdown). While Linux comes up fine, Asterisk won't start because the drivers are loading in the wrong order. fixed by: 1) sh /usr/src/fix-asterisk-modules.sh 2) sh /etc/init.d/asterisk start Is this a

Hi List, I am trying to locate some detailed documentation and sample configs. I downloaded and compiled Asterisk, and I haven't been able to find much detailed docs on the config files. The distribution I compiled and installed doesn't have any config files, and the handbook is good but doesn't cover all of the configs. Here's my specific problem, when launching Asterisk for the

I have ADSI working on my Aastra (Vista/Nortel) 350 phone and everything is working fine. However, I want the asterisk.adsi to load into the 'self-load' slot but can't figure out what the correct FDN for doing this is. Does anyone know the right FDN for the SL slot on these phones? Also, does anyone have any cool/interesting ADSI scripts they wouldn't mind sharing? I'm

I have an Aastra 390 ADSI phone. It's not locked. I can call ADSIProg without a problem and it programs my phone. Calling Voicemail2 also programs my phone. However, in order for the VMail option to appear on the screen I have to go into the Services menu, pick Asterisk PBX and pick Select. Then the VMail softbutton appears on the screen, but any time I make a call it goes back to the

Hello ! In file app_meetme.c we can read A ZAPTEL INTERFACE MUST BE\n" "INSTALLED FOR CONFERENCING FUNCTIONALITY.\n" I receive message, when I try conference WARNING[28686]: File app_meetme.c, Line 151 (build_conf): Unable to open pseudo channel -- Playing 'conf-invalid' Does it means that I cannot establish conference without any hardware zaptel interface ??? What

In this message I try to summarize what I have learned in these last two weeks. My primary sources of informations were the * list archives and linux ISDN docs. I ain't no * master, so don't trust too hard. Relevant messages from the * list for the current discussion are: 009177.html 009268.html 0498.html 0849.html My setup is an Eicon Diva (HiSax: Eicon.Diehl Diva driver Rev. 1.1.4.2)

Hi All, I recently purchased the Asterisk Developer's Kit (TDM) to try out Asterisk. After following the directions in the Digium's FAQ topic entitled "Q. How do I configure my TDM40B and X100P?", I'm receiving the following error: WARNING[1074428608]: File chan_zap.c, Line 6748 (load_module): Ignoring rxwink ERROR[1074428608]: File chan_zap.c, Line 6692 (load_module):

Hi all, I can't seem to find any info on this anywhere on the web, except that BT caller ID doesnt use the standard bellcore system in use in the US. So, if anyone here in the UK is onlist and using the x100p successfully, please let me know. TIA, Dave

I just received an unlocked ADSI phone and I am playing with the ADSI script. I was wondering how I can include Voicemail functions (Check new messages, Delete message) into the soft buttons. I checked in app_voicemail.c and it looks like these functions have already been programmed. Is there a voicemail.adsi script somewhere? If not, then how do I get the functions I want onto my phone? Thank

Hi All, I have an equipment loaded with 4 X100P (numbered 1-4)) and one T400P (numbered 5-8). Everything works fine except that I cannot use one of the FXS ports (number 5). If I configure zapata.conf to recognize it, the whole system voice quality suffers. I've tried already to switch PCI slots, with no results. Below is a snapshot of my /proc/interrupts, maybe this can shed some light on

I am working with the telantek.adsi file, and I was wondering how I would create a softkey for Transfer. I tried making a key definition and using SENDDTMF "#", but that didn't work. Is there another way I could do this? Also, does anybody have any ADSI scripts for use with Asterisk that they would like to share? Thank you for your time. __________________________________ Do you

Hello, * newbie here, I'm designing a setup that is to eventually be used in a production virtual PBX/VoIP service. Customers need to be able to change their setups over the web - I want them to be able to do simple things like setting up call forwarding, as well as more intricate stuff that will require me to re-generate their dialplans. Administration of the service is to be

I have posted soo many times in the past but never recieved even a single reply . seem like you people are ignoring me or either way too busy .. never mind this is my last try . How can record a conversation with asterisk ? I tried to use Record() but dint work for me .. here is what i tried . exten => s,1,Wait,1 ; Wait a second, just for fun exten => s,2,Answer

Hello, We just had some new T1s turned up today to replace others that our contract has run out on and we are now getting more echo on the new T1 lines than we had on the old ones. The T1 type is 24-channel, D4/AMI SF Robbed-bit(the same as the T1s they replaced) The problem is that we are getting echo on about 10% of the calls in and out placed on these new T1s compared to less than 1% with

I am planning to replace our cisco 7200 core router with Linux. We currently serve around 1500 (3/4 DSL - different router) customers with probably half of them being concurrent at any given time. We have a fiber network and customers currently aren''t managed as far as how much bandwidth they can use at anytime. Therefore I have constructed a working tc qdisc Linux router as a test. It

I am having trouble with the AgenCallBackLogin app. I can't seem to define a context for the queue. Here is the relevant configs: queues.conf: [general] [default] [q_lo_1] music = default strategy = ringall context = c_in_1 timeout = 15 retry = 2 maxlen = 0 member => Agent/@3 agents.conf: [agents] autologoff=10 wrapuptime=15000 group=1 agent => 1001,1234,Agent1 agent =>