similar to: How to test BRI lines energy saving mode ?

Hi, In Europe many Telcos implement power-save mode (See http://www.isdn-test.de/ihhe12.htm#Heading37 and scroll down to 'Activation / Deactivation' for more information). Would you agree to have this feature added to the ones already discuused for next Asterisk release ? (See https://wiki.asterisk.org/wiki/display/AST/AstriDevCon+2010) Regards -------------- next part -------------- An

Hello, I always thought that Energy Savings mode existed with ISDN Basic Rate Interface in Point -to-multi-Point but it didn't with Point-to-point. Is this correct ? Have you ever met a public PSTN switch configured to "cut" B ISDN channels in Point-to-point.signalling ? Regards. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I need to configure BRI 4span card in dubai in vicidialnow for dialer perpose. in that i have small confusion which is NT an TE mode . that was i am setting perfectly but dubai telco what they are use for this i dont know which parameters are use for that . please help me. -- Best Regards, Mahesh Katta *BUZZ**WORKS* Business Services Private Limited BANGALORE | CHENNAI | HYDERABAD |

I am facing one way audio problem in sip trunking between asterisk and avaya. +-------------+ +----+ | avaya sip |-------| P1 | +-------------+ +----+ | | | +-------------+ | Asterisk | WAN

I'm looking into getting a new phone and wondering what the difference in functionality is between a single line phone with call waiting and a real 2 line phone (either a real SIP phone or an analog 2 line phone and a 2 port ATA) is. Why would I want the real 2 lines vs. just being able to take an incoming call via call-waiting? Cheers, b. -------------- next part -------------- A non-text

Hello, I have a current asterisk 1.8.13.0 asterisk-addons 1.6.24 asterisk-sounds 1.2.1 dahdi-linux-complete 2.6.1+2.6.1 libpri 1.4.12 and asterisk-gui 2.1.0.rc1 (not trying to use the gui, want to do everything by hand) with a TDM410 with 2FXO and 2FXS. I have my POTS (PTNS) line plugged into port 1 (FXO) and a analog phone connected to port 3 (FXS). I compiled asterisk with asterisk

I am not that good at regex and it's use in Asterisk. I am running Asterisk 1.4.13 Currently I have this in my extensions.conf for incoming calls on our house phone line: [housemenu] exten => s,1,GotoIF($["${CALLERID(num)}" = "815xxxxxxx"]?s|12); 815xxxxxxx is our home phone number, when caller id fails or is missing that is what is recorded. I want to expand this

Does anybody know of a way to test whether a mySQL connection invoked from the dialplan is current or not? For example: extensions.conf =============== [context] exten => _X.,1,MYSQL(Connect connid localhost user pass db) exten => _X.,n,MYSQL(Query resultid ${connid} SELECT `something` FROM `table` WHERE `number` = ${EXTEN}) exten => _X.,n,MYSQL(Fetch foundRow ${resultid} something)

Hi All, Sorry if this has been covered already, but I don't tend to follow this list as close as I should these days. Problem is that if a call comes in to a queue without option 'r' specified - moh plays as expected. Now, when that call is answered, all is fine. Trouble comes when that person then puts the caller on-hold. No moh is heard by the caller (in fact, they get silence).

Hi, Is the any kind of 'tag' that I can include at the end of my message to make the list processing software ignore and dispose of my disclaimer? In other words - something like <disclaimer> at the end of my message would inform the list software to remove any lines after it. My massive disclaimer is added by the server you see - and it's now annoying me - let alone the rest

I had a working system using * 1.0 and then 1.2 and now Asterisk 1.4.13 with addons 1.4.4, zaptel 1.4.6, libpri 1.4.2. I have a mix of Grandstream (GXP2000), Uniden uip-200, Linksys Wireless G, and analog phones via Adtran chan bank. When I went to * 1.4.13, the Uniden phones stopped being able to login to voicemail. All phones are on same lan with Asterisk. I get 'Login incorrect'

I am trying to get distinctive ringing going again with these phones, depending on the outside line the call comes in on. I had a working 1.0.x Asterisk setup using: SetVar(ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) Which used the short quick rings. In Asterisk 1.4, I have tried several things, but I think the correct syntax is: Set(_ALERT_INFO=<http://127.0.0.1/Bellcore-dr2>) But

My PSTN line doesn't allways hang up properly after it goes to voicemail. The problem occurs when a caller hangs up during the initial greeting. Even though the hangup accured, voicemail continues to record, usually a fast busy and/or a teleco generated "please hangup now" message. After the voicemail.conf 'maxmessage=180' expires the line simply stays offhook. The hardware

Hey Guys, I have upgrade my asterisk 1.2 to 1.8 and suddenly my allpage.agi stopped working look like asterisk 1.8 did some changes in manager apps i am doing following.. my phone is ringing but not auto answer could you give me some issue what i am doing wrong ? root at ubuntu-test:~# telnet 127.0.0.1 5038 Trying 127.0.0.1... Connected to 127.0.0.1. Escape character is '^]'. Asterisk

Hi I have come across a problem, with my tdp410 and soekris board (basically pc on a chip amd geode cpu). I am using the box as a firewall/asterisk box. The problem occurs when I drop ppp and I get dead loop dectiotn going, I seem to lose interrupts and get lots of messages in syslog from wctdm24xx saying missed interrupt increasing latency its out lined here

Use Set(CALLERID(num)=9999999999) instead of using CALLERID(all). Remember to set this BEFORE you Dial. -->> -----Original Message----- -->> From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- -->> bounces at lists.digium.com] On Behalf Of Vieri -->> Sent: 06 February 2009 12:36 -->> To: asterisk-users at lists.digium.com -->>

Hi All, We are running to a weird problem, we're using asterisk 1.2 as a production server (I'm wiling to move very soon to more recent version) and our problem is when somebody try to join a conference he's told that he's the only one in the conference but in fact there is some 3 or 5 or whatever people in that same conference, after several tries he can/cannot enter the

Problem as follows: [default] exten => 777,1,Gosub(sub,1,1) exten => 777,n,Hangup() exten => h,1,NoOp(hung up in 'default' context) [sub] exten => 1,1,NoOp(in sub) exten => 1,n,Playback(tt-monkeys) exten => 1,n,Return() exten => h,1,NoOp(hung up in 'sub' context) This works fine if the caller listens to all the 'tt-monkeys' and let's the system

Hi, I need to rebuild a system which has 4 BRI ports and is connected to Point-to-multiPoint lines, in a country where telco often "drop lines for energy savings". I'm planning to use latest 11.4.0 asterisk version along with dahdi and libpri (no misdn). Which version are recommended for Dahdi and Libpri ? My main requirements are, beside having calls coming in and out: - keep a

Hi, Here http://www.voip-info.org/wiki/view/Asterisk+func+device_State you can find a link to download a backported for Asterisk 1.4 version of DEVICE_STATE function. (Elsewhere, you can find reference to another backported function DEVSTATE which seems to behave the same as DEVICE_STATE). As I would like to prepare as much as possible, my dialplan to 1.6 and beyond, I would prefer to use