similar to: PADLOCK asterisk 1.6

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

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 , 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 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, What is the easier way to make call using a password? I have A2billing but its authentication is too big, I would like four digits long. Something like that: In any extensons, the user dial the password and make call. Thanks in advanced! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part -------------- An HTML

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, 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

Hello everybody, Anybody here knows about Astmail ? I have set up in a server but something is going wrong! i can open its web interface but when I put the extension number and its password I receive:Invalid mailbox or password My asterisk is 1.6 and my S.O is debian lenny. I know this is not asterisk but as the project has not a forum or mail list, I am trying help here! Thanks in advanced!

Hi there! Anybody knows why I am receiving this output from CLI: No such command 'dialplan reload' (type 'core show help dialplan reload' for other possible commands) Look like asterisk dont see dialplan? Is it possible to restart it ? Thansk Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda -------------- next part

Hi there! I am trying to configure Web-meetme on Asterisk 1.6. I have followed the README and everything looks ok,therefore, when I try to open the webpage appear the folowing messages:DB Error: connect failed Testing with a php script, the message Connected successfully is shown. Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

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 there! I am having some difficult in receiving calls from my E1 link using mfcr2. I can make calls normally , but when I receive an incoming calls, the phone ring I answer it ,so, I listen busy tone and then the phone ring again and again. look the log: -- Executing [4801 at from-pstn-TE1:1] NoOp("DAHDI/2-1", ""1233220567" <1233220567>") in new stack

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 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, 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

Hi all, I can?t pickup calls on my asterisk. When I try to load app_pickupchan.so I receive following message: Module 'app_pickupchan.so' was not compiled with the same compile-time options as this version of Asterisk It was working fine until few time ago. What is going on? Thanks! Att, Flavio Roberto Miranda MSN:flaviormiranda at hotmail.com Skype: flaviormiranda

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, 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

I was wondering if someone with more familiarity with MC on X86 could consider looking into adding support for the hyphenated versions of the VIA Padlock instructions? If anyone is up for it there are details within these two bug reports.. http://www.llvm.org/bugs/show_bug.cgi?id=8556 http://www.llvm.org/bugs/show_bug.cgi?id=10266 -- This message has been scanned for viruses and dangerous