search for: indreia

Hi All, I am in desperate need of this feature. I want to play background music during a call while the 2 parties are having some lovely conversation (or maybe give them a sort of cursing background if they are cursing each other). I found this post which talks about creating a ghost call with the help of queues and putting that queue in a meetme room where queue will play the song/curse and the

Loan, Thanks for your help in this matter. Having never used astdb before, can you point me to an example on this?? Thanks hugely, Clara >> Hi Clara, You could put some data into astdb and query for the outgoing line and callerid based on internal callerid (extension). something like user/201/outline 89859715 user/201/outcallerid 89859715 and so on... By the way:

please can someone send to me his files like zaptel & zapta if he si using TE110P thank you

Hello, Trying to upgrade (from asterisk18-1.8.6.0-1) to the latest RPM version from Asterisk repo I found that asterisknow-version is needed by package asterisk18-core-1.8.7.0-2 How could this be explained? Best regards, Ioan ######### [root at localhost ~]# yum update asterisk18* -x asterisknow-version Loaded plugins: fastestmirror, kmod Loading mirror speeds from cached hostfile * base:

...e testing machine) Even that the translation service provided free by Google is not the best way to read our article in English (or other languages) I encourage you to read it (the pictures and the results are very easy to understand) and send your feedback or comments here. Best regards, -- Ioan Indreias www.modulo.ro Notes: [1] - http://www.modulo.ro/Modulo/ro/Articole/Determinarea_capacitatii_maxime_a_unei_centrale_Asterisk.html [2] Maximum number of G.711 ulaw - ulaw SIP calls 38 - Norhtec MicroClient Jr DX 130 - VIA EPIA EN12000EG 176 - Asus Pundit R350 320 - Gigabyte 945GCM-S2L

Hello. I have a TE212 configured in E1 mode. This is shown in a cat /proc/zaptel/2 and 3 (where the card is configured): cat /proc/zaptel/2 Span 2: TE2/0/1 "T2XXP (PCI) Card 0 Span 1" HDB3/CCS/CRC4 RED NOTOPEN 25 TE2/0/1/1 Clear 26 TE2/0/1/2 Clear 27 TE2/0/1/3 Clear 28 TE2/0/1/4 Clear 29 TE2/0/1/5 Clear 30 TE2/0/1/6 Clear

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

Dear all, I am currently trying to configure a PBX make use of a multiple of outgoing lines, currently my extensions.conf looks something like below >> ; extensions.conf ; 20th October 2008 [globals] sip1=201 sip2=202 sip3=203 sip4=204 [general] autofallthrough=yes [default] [incoming_calls] exten => _89859715,1,Dial(SIP/201) exten =>

Is anyone using asterisk with fail2ban? I have it working except it takes way more break-in attempts than what is set in "maxretry" in jail.conf For example, I get an email saying: "The IP 199.204.45.19 has just been banned by Fail2Ban after 181 attempts against ASTERISK." when "maxretry = 5" in jail.conf Perhaps someone else is experiencing this or has resolved it,

Hello, We have chosen to upgrade our Trixbox installations (2.6.2.3, asterisk 1.4.20) and everything work smooth. The problem we face now is that asterisk14-addons-mysql looks to have not been compiled with uniqueID feature and we are asking your opinion about what should be the best fix for this problem. Our workarround was to overwrite (from backup) the cdr_addon_mysql.so module, but this is

...ailing List - Non-Commercial Discussion <asterisk-users at lists.digium.com> Message-ID: <BANLkTi=6-8X2bbDTQUU_VfeJ_3+7O7MzWg at mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" Thanks a lot loan. Will try it today. Cheers On Mon, May 9, 2011 at 6:25 PM, Ioan Indreias <indreias at gmail.com> wrote: > Updated dialplan: fix a typo when using MOH variable and now you have > truly dynamic conference rooms. > > Have fun, > Ioan. > > +++++++++++++++++++++++++++++++++++++++++ > exten => _[12]XXX,1,Set(__MM=${EPOCH}) > exten => _...

I want to use a voip speaker phone as a room monitor. Requirements: A phone that I can set to auto answer in speaker mode. A phone with a good speaker phone. Ability to make the audio one way. I want to monitor the room but not have my voice heard in the room. Yes, the mute button can accomplish this also. I have been using the SPA942's around the house (the speaker is just ok but

Hello mailinglist, I want to connect Asterisk with OpenBTS and make a call with a mobile phone. I use: Ubuntu 11.10 + Kernel 3.0.22 GnuRadio 3.3.0 Asterisk 1.8.13 OpenBTS 2.8 Nokia Mobile Phone OpenBTS works and I can send sms from the OpenBTS server to the mobile phone. What I also need is a call between Asterisk and OpenBTS. I have also two soft phones which works with Asterisk. And also

Hey list, I have a problem when I host 2 SIP-accounts on the same Asterisk-server. Asterisk picks out the SIP-account on alphabetic order A --> Z. In my sip.conf : register => user1:passwd1 at server/user1 register => user2:passwd2 at server/user2 [YOCAN-3starsnet] type=peer host=server username=user1 secret=passwd1 fromuser=user1 accountcode=user1_in [ITCENTER-3starsnet] type=peer

Hi people, I have the following dialplan, but it doesn't have the behavior that I think it should have. [default] exten => 2001,1,Answer exten => 2001,n,Dial(local/3005) exten => 2001,n,Hangup exten => 3005,1,Set(__RINGTIMER=10) exten => 3005,n,Macro(exten-vm,novm,3005) exten => 3005,n,Hangup When I execute the Originate (AMI) with the argument Channel=local/2001, It rings

Hello, I'm writing here hoping to have a hint from Asterisk/Digium packager maintainer, Jason Parker (of course that any other's opinion is welcomed). We have installed an asterisk machine using Asterisk and Digium repos. Unfortunately we have found that an Astribank could not be connected due to the lack of the firmware files May 18 14:19:22 tenora1 kernel: usb 1-6: new high speed USB