similar to: I see Asterisk 1.2.2 into the ftp or was a vision?

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

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 =>

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

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

What are the alternatives to FWD with IAX2 registration capability. FWD is great, but their IAX2 is not the priority and if it goes down it takes days to restore it. I want to use IAX2 protocol but the end point (Sipura unit) need to be able to register over SIP behind firewall. Line1 is registered with FWD PSTN need to be registered with somebody else. What are my alternatives? -- #Joseph

Hi Fran, you could do it using Adobe/Macromedia Flash Media Server 2, but I guess that's not the answer you are looking for. If you manage to do this and release it under GPL I'll kick in $50 for a bounty. Regards, Dean Collins dean@collins.net.pr +1-212-203-4357 +61-2-9016-5642 (Sydney in-dial). -----Original Message----- From: asterisk-users-bounces@lists.digium.com

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

should I use mpg123 with asterisk 1.2.7 or should i use the native player asterisk has? the target machine will receive heavy load. also, has anyone succedded in compiling mpg123 in a dual core pentium with centos 4.3 ? -- ------------------------------------------- Erick Perez Linux User 376588 http://counter.li.org/ (Get counted!!!) Panama, Republic of Panama

I'm getting "Got SIP response 406 "Not Acceptable" back from 10.0.1.2" occasionally when try to dial to SPA942 , anyone has any idea on this before i consider Firmware upgrade? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080326/2fd2c557/attachment.htm

Hi list, I have for a year I have an account to call with broadvoice from about 3 days beginning a not registered problem of, asterisk shows to a message of error with the DNS, and my dns this working fine WARNING[5770]: chan_sip.c:7595 transmit_register: Probably a DNS error for registration to 908XXXXXXX at sip.broadvoice.com@sip.broadvoice.com, trying REGISTER again (after 20 seconds) [Dec 16

My provider migrated from an old EOL softswitch to Trixbox. I have a number (8159093011) on a different server on a different network. It appears as though the incoming calls are trying to authenticate against that number, which isn't present on the box. Could someone help me decode this debugging output? I was calling 8159911010. My server is 208.100.1.33. Theirs is 208.1.87.235. I

Hi friends, I am using Real Time Asterisk Architecture where I have put the Sip users/peers and extensions defining the dialplan in tables in a mysql database. Currently, asterisk points to my single database server as configured: ------------------------------------------ /etc/asterisk/res_mysql.conf ------------------------------------------ [general] dbhost = xxx dbname =

I have asterisk 1.0.5 installed via apt on a debian system. It's a custom distrobution called Voyage Linux that runs from a flash card and I have a hard drive installed with mysql installed on it as well as apache. I have been though the AMP install guide (asterisk management portal) and in the interface it has a place for me to record new IVR menus. I have to dial *77 to begin recording

Hello all, I need to setup a new asterisk system with the following requirements: 1. Will be moving from chan_sccp to sip (7960's), but I want to support the sccp phones until everyone has been migrated. 2. Need to maintain current portability of the 7960's. (ie a user can unplug his phone from the internal LAN, take it home or wherever, and plugin and have the phone register and

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

I've just started using AMP and found that I have a problem with escaped characters in config files. In particular, I have a custom config item that needs a semicolon in... SetVar(_ALERT_INFO=info=auto-answer;delay=1) To get the part of the line after the ; to be accepted by Asterisk as a non-comment it needs to be escaped with a backslash, but I have found that I need to put two

I am having trouble understanding how to use this. I want to transfer certain incoming calls from an IAX ITSP based on caller ID. From what I can make of the docs, I thought I need to do something like this... exten => _NXXNXXXXXX,n(nocid),transfer(1000) exten => _NXXNXXXXXX,n,noop(boo,${TRANSFERSTATUS}) exten => _NXXNXXXXXX,n,hangup exten =>

I've an asterisk 1.2 connecting to a quescom gateway via SIP, the caller (asterisk) can hear the called, but the called hears nothing. Since both machines are on public ip, what other problem can it be ? There's one configuration working : lynksys pap -sip-> asterisk server -sip-> quescom this way both sides can hear voice but with : lynksys pap connected to a switch -sip->

Is there any way to have more specific disposition codes in the CDR? Currently there are only 3 values: ANSWER, NO ANSWER, BUSY. In this way, when i call a cell phone that is switched off i get "NO ANSWER", while i would like to be able to log that the call is not answered because "The customer you have dialed is unavailable at the moment". The same for "non