similar to: CAPI- 2 Cards

I have a E1 conected to asterisk all zap channels are ok, but when calls come into Asterisk caller don't hear none ring, the call goes straight into the menu, how can i simulate 2 or 3 rings? here it is my conf. exten => s,1,Answer exten => s,2,Wait,2 exten => s,3,NoOp(${CALLERID}) exten => s,4,ResponseTimeout,45 exten => s,5,DigitTimeout,3 exten =>

Somone in europe have had succes getting Callir ID showed on a phone screen conected to an Handytone 286 ? Adri? Vidal -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: text/enriched Size: 235 bytes Desc: not available Url : http://lists.digium.com/pipermail/asterisk-users/attachments/20041124/e5514052/attachment.bin

Someone have info about install an AVM fritz into FC3 ? I'm getting problems with kernelcapi, after succesfully installed the fcpci support. Thanks -- Adri? Vidal adriavidal@gmail.com Mail is better with 1Gmail

Someone can confirm the new release is out? Haven's seen any post about it!!!!! -- Adri? Vidal adriavidal@gmail.com

Hi, i'm trying to compile Asterisk under YDL 3.0.1, libpri, zaptel compile ok, but at make install in asterisk give me this error, have an idea because it can be? Thanks in advance. k\" -DASTLOGDIR=\"/var/log/asterisk\" -DASTCONFPATH=\"/etc/asterisk/asterisk.conf\" -DASTMODDIR=\"/usr/lib/asterisk/modules\" -DASTAGIDIR=\"/var/lib/asterisk/agi-bin\

Have someone running fine Alert-Info with a Snom 370 ( System Information: Phone Type: snom370-SIP MAC-Address: 0004132661BD IP-Address: 192.168.10.170 Firmware-Version: snom370-SIP 7.3.14 14961) i've tried exten => 200,1,SIPAddHeader(Alert-Info:<http://www.notused.com>\;info=alert-external) exten => 200,n,Dial(SIP/${EXTEN},30) Can see into the phone SIP trace is

I'm testing some of the new features of Asterisk 1.8, but seems an impossible mission to make the calendars run. cli show empty: **CLI> calendar show calendars Calendar Type Status -------- ---- ------ * And i can't see anything into the log, calendar.conf is ignored. Any sugestion? -- -- Adri? Vidal -------------- next part -------------- An HTML

Hi Everyone, I am running DD-WRT on a router that feeds about 30 Aastra 6753i. The phones occasionally go into "No Service" mode. The POE switch doesn't seem to be the problem as it's tested fine. I think the router sometimes gives up and comes back quickly. Or something of that nature. However, the connections are maintained if a call is going on because there are peer to peer

Hello! I'm having a bit of an issue with one of my installations that I cannot figure out. For some reason, when two people are in a call (both local to the * box, same subnet, pure SIP), the call will randomly be placed on hold and provide MOH to the other party. We're using Polycom IP430 handsets almost exclusively for this installation. Can anyone think of a reason why a call would

A brief update on a few things which should be interesting to anybody who's got a Mac running OSX ... 1) JPT, integrated desktop dialer now with Asterisk support (and X-Lite too) Jon Nathan, the author of Jon's Phone Tool (JPT) has just released version 2.0.4 which supports dialing through local and remote Asterisk servers. It also supports dialing through a locally installed X-lite.

I am trying to setup a simple pstn gateway using Asterisk and a X100p card. I have got everything installed using Redhat 9 and am able to load Asterisk. I also configured sip and I am able to connect to the asterisk gateway with Xlite on the windows side. I am able to dial 1000 and get the welcome message. What I am NOT able to do is dial a seven digit local or 10 digit long distance number and

Followed instructions from these old post, CVS updated my asterisk too, edites makefile... but ---------------------------------------------------------- Get oh323 from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/ openh323-Janus_patch4-src-tar.gz Get pwlib from http://www.inaccessnetworks.com/projects/asterisk-oh323/Libraries/ pwlib-Janus_patch4-src-tar.gz Get

Someone have had good luck compilig h323 into YDL? first thinked was a bug in code but "twisted" said it is " wierd - isn't that the recursive pthread lib? If so, do you have the kernel development headers/libs installed?" I've instaled kernel source, what more can i do? any help would be very wellcome, i've googled everywhere but can't find nothing about it.

Hello everybody, I'm having trouble with samba 3.5.11 and an access 97 database. I work in a business with 5 users that use a mdb database over samba, and for the first to make a query everything works fine, but if some other user tries to make a query when the first one is doing it, the database works really slow, like if the .mdb file is being blocked (that's what I suppose). I've

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

Si no puedes pulsar los enlaces, pulsa ??Mostrar el contenido?? e int?ntalo de nuevo. Correo-electr?nico enviado el 10/22/2010 4:53:03 PM, por Adri?n Furlan: Buenos d?as, ?Por fin un sistema inteligente y sencillo para administrar sus contactos! Con UNYK, he reunido todos mis contactos en una sola libreta de direcciones, que se actualiza autom?ticamente. Uno de mis contactos modifica sus

Hi, does the hint extension work together with the Snom phones in stable? I don't get an error in the dialplan, but it does not work either. On SIP/26 I want to monitor SIP/22. This is what I do right now: extension.conf [incoming] exten => 955,hint,SIP/22 exten => 955,1,Dial(SIP/22) sip.conf [26] ... subscribecontext=incoming ... Running Asterisk 1.0.2-BRIstuffed-0.2.0-RC2b TIA

Hi to everyone! I have succerfully instaled my new Asterisk 1.4 on my debian etch. I have my users in sip.conf like this: [200] type=peer host=dynamic context=home secret=200 callerid= 200 dtmfmode=rfc2833 nat=yes mailbox=200 at home disallow=all allow=ulaw I can make calls in my LAN but i can?t ear comunications with another client trought Internet. My Asterisk is in my LAN and i not have a

Dear R&Mac developers(users), at http://www.eco-dip.unimi.it/R you can find the latest (and last) beat version of R for MacOs corresponding to R vers 1.1.1. Release date is Dec 10 2000. It is no longer a pre-alpha version but a beta one :) In what follows you can find what's new in this version but please remember that it is still a beta version intended mainly for R developers

Dear R&Mac developers(users), at http://www.eco-dip.unimi.it/R you can find the latest (and last) beat version of R for MacOs corresponding to R vers 1.1.1. Release date is Dec 10 2000. It is no longer a pre-alpha version but a beta one :) In what follows you can find what's new in this version but please remember that it is still a beta version intended mainly for R developers