similar to: Call using password

Hey there, I am trying to setup a2billing on asterisk 1.6 , but ,when I try to access its web page I see the a2billing directories:Index of /a2billingNameLast modifiedSizeDescriptionParent Directory -admin/15-Sep-2010 19:19-agent/15-Sep-2010 19:21-common/15-Sep-2010 19:18-customer/15-Sep-2010 19:20-Apache/2.2.9 (Debian) PHP/5.2.6-1+lenny8 with Suhosin-Patch Server at Att, Flavio Roberto

Hi all, I am trying to set up DND in my asterisk, I am using the following context: [app-naoperturbe] exten => *11,1,Set(DND=${DB(ddisturbe/${CALLERIDNUM})})exten => *11,2,GotoIf($["${DND}" = "YES"]?*11,3:*11,101)exten => *11,3,Set(DB(ddisturbe/${CALLERIDNUM})=NO)exten => *11,4,Playback(beep)exten => *11,5,Hangup()exten =>

Hi all, I realize that the application Receivefax can't handle with more than one fax at the same time. In a environment with a lot of fax, some caller get the signal but the operation can't be completed. Is there a way to send busy tone to the second caller? Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part

Hi , Is it possible to have two meetme room in asterisk 1.6 which each one have a different language? I mean, one room the annoucement is in Portuguese an another in english? Today I can change over the sip.conf and it is valid for all room. regards! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An

Hi list, I have search for a clear explanation about this mensage " WARNING T.30 ECM carrier not found", but until now I dont succed on it.Anybody know how can I handle with this problem? I have Asterisk 1.6.2.13 with TDM 410p and the Fax is connected to Dlink FXO dvg 2032s. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

Hi all, Please, anybody that have some knowllege about E1 configuration could give some guidance about it? I trying to set an Asterisk with E1 CAS signalling and everything looks good, but when I try to go out with calls I receive the follow message: == Using SIP RTP CoS mark 5 -- Executing [21341400 at local:1] Dial("SIP/4804-00000000", "DAHDI/g11/21341400,,t") in

Hi all, After a lot of trouble with a TE110p working with mfcr2 , brazil variant, everything looks great,but I can not go out of my calls. When I try I receive the following log: == Using SIP RTP CoS mark 5 -- Executing [33220567 at local:1] Dial("SIP/4804-0000001a", "DAHDI/g11/33220567,,T") in new stack == Everyone is busy/congested at this time (1:0/1/0) -- Auto

Hi all, What hell hapen here? asterisk:/etc/asterisk# /etc/init.d/dahdi startLoading DAHDI hardware modules:FATAL: Error inserting dahdi (/lib/modules/2.6.26-2-686/dahdi/dahdi.ko): Device or resource busy wct4xxp: done wcte12xp: done wct1xxp: done wcte11xp: done wctdm24xxp: done wcfxo: done wctdm: done wcb4xxp: error wctc4xxp: done xpp_usb: doneError: missing /dev/dahdi! When

Hi list, Anyone has some guidance in how can I project a SMS gateway with Asterisk. I mean, some good web link,pdf or something like that? Thanks in advanced!!Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormirandaru -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Please, could somebody point me out what is going wrong in this line below? exten => _00XX.,1,Dial(DAHDI/G0/021${EXTEN:4},45,rT) As I know, such line must convert any number dialed to 021, therefore, as we can see, it's kept the number dialed! -- Executing [00151236445600 at a2billing:1] Dial("SIP/2000-00000000", "DAHDI/G0/0151236445600,45,rT}") in new

Hi all, I am trying to set up Web-meetme in Asterisk 1.6. After some attemps, I am receiving the message:DB Error: connect failed What could be ? Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi all, Every time I reload my asterisk it fall down and the following message appear on log: parse error: No category context for line 7 of /etc/asterisk/chan_dahdi.conf If I comment that line, it change to other line. There are some thing wrong with my dahdi? Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda --------------

Hi all, Recently I have instaled one Digium TDM410 on my Asterisk. After instaled , I can do outgoing calls but I cant receive calls. I receive the following messages: chan_dahdi.c: Got event 2 (Ring/Answered)...[Sep 14 11:24:44] NOTICE[2654] chan_dahdi.c: Got event 18 (Ring Begin)...[Sep 14 11:24:44] WARNING[2654] pbx.c: Channel 'DAHDI/4-1' sent into invalid extension 's' in

Hi all, I know Hylafax is an application and not Asterisk but I'd like to post a problem found in configuring such application and Asterisk. I am able to reveive fax,but , I can't receive it in e-mail. Although I put my e-mail in /etc/hylifax/Dispatch I can't receive. Anybody know where I must to add something else in order to make it works! Thanks in advanced!! Att,

Hi all, Is it Openr2 supported by asterisk 1.6.2 without pach instalation ? I am a little bit confuse about that. My asterisk 1.6.2 show me the following warning: Unknown signalling method 'mfcr2' at line 29. I had downloaded and instaled openr2-1.3.0 but the messages is still shown. Which files I must to change in order to have everything working properly. Best regards! Att,

Hi, Trying to configure my tdm410p card, my dahdi in asterisk cli was missing. ! ael agent agi cdr channel cli config console core database devstate dialplan dnsmgr dundi features file group hangup help http iax2 indication keys

I am having a one way audio issue with xlite clients behind NAT. They can connect to the server and make calls but no audio is heard on the other end. my sip conf [general] context=default bindport=5060 bindaddr=0.0.0.0 srvlookup=yes canreinvite=no[tomfmason] type=friend secret=secret callerid="Thomas Johnson" <XXXX> host=dynamic nat=yes canreinvite=no disallow=all allow=gsm

please send me the ways to change asterisk port from 5060 to 5061 i need to configure it because we are already using 5060 port in router then we cant use it again we have to configure other sip server so please suggest me a way.......................... On 4/10/11, asterisk-users-request at lists.digium.com <asterisk-users-request at lists.digium.com> wrote: > Send asterisk-users mailing

Not sure if a2billing can be shared here, but ill give a shot If the credit < min_credit the IVR play: sorry you have 0 credit and hangup, I want it to FW me to the IVR to add voucher, please let me know: here is log: [18/10/2010 07:01:12]:[file:a2billing.php - line:75]:[CallerID:]:[CN:]:[IDCONFIG : 1] [18/10/2010 07:01:12]:[file:a2billing.php - line:76]:[CallerID:]:[CN:]:[MODE : standard]

Hello everyone. I'm using Asterisk 1.4.31 and A2Billing 1.7.0 to manage a small wholesale operation, so I configured A2Billing for not to answer the call nor play any greetings or balance notifications to the caller. I'm authenticating each customer by it's IP address, and each customer has it's own context, in which I set the following: ;=====in extensions.conf======