similar to: Get rid of "Really destroying SIP dialog"

Hi, Why do CDR second field, src field have a dest???? The real src field is 9500. Is a bug?? Example; "Q-aereos","1147938811","9500","outbound","1147938811","DAHDI/31-1","SIP/9500-0de0ea60","Dial","SIP/9500|60|t","2009-08-11 18:12:41","2009-08-11 18:12:45","2009-08-11

Hi, I'm getting the following at asterisk startup. OCBC was working with 1.4 no problem, but now under 1.6 (I've tried 1.6.1, 1.6.2-beta3/beta4) I can't seem to get rid of this .... anyone? WARNING[32664]: loader.c:385 load_dynamic_module: Error loading module 'res_config_odbc.so': /usr/lib/asterisk/modules/res_config_odbc.so: undefined symbol: ast_odbc_clear_cache

Hi everybody, I am having DTMF detection problem on DISA with my callback system. For many users, it keeps playing the dialtone even after they have input their number. I have trunk setup to both g729 and ulaw. What could be the reason for this problem. Some users have to dial a few times before the system can recognize their dialed number. -- Zeeshan A Zakaria -------------- next part

In several places online, and in the Asterisk F.O.T. book, there is a warning against using '_.' saying: "[it] should probably never be used". However, the need often arises act on numeric extensions that begin with *'s and #'s, and '+', and of course _X. does not match I have tried exten => _[0-9*#+]. but that seems to be the functional equivalent to _X.

Hi listers, We want to implement one call center with asterisk. The idea is it should be scalable, with openser as an dispatcher and bunch of asterisk servers to do ACD, Queues, Agents things... Easy to say :( Look closely to the current asterisk, we do see some problem: - SIP registrations was stored in astdb. - And queue members also was stored in astdb. - ... asterisk was built as

On Wed, Oct 1, 2008 at 6:34 AM, Andrew Joakimsen <joakimsen at gmail.com> wrote: > On Sun, Mar 23, 2008 at 11:34 PM, Tilghman Lesher > <tilghman at mail.jeffandtilghman.com> wrote: >> It is completely illegal in any country that recognizes patents. > > You mean countries that recognize software patents, right? As resident of country where the file is hosted - yes we

Hi folks: I know it's come up a few times before, but I need some more detail. I'm looking for a SIP DECT (cordless) phone for North American installations. I've heard only of the Siemens Gigaset S450/C450 phones. Apparently these aren't sold for use in NAm, even though they're supposed to be legal (in the United States, anyway). On top of that, I understand they have some

Hi, I'm using gotoiftime on asterisk, but it seems&nbsp; there is a difference between the asterisk time and the system time. could it be because i adjusted the system timezone on my linux? do asterisk not detect the change of timezone on the system? How can I fix this prob? Regards, nhadie -------------- next part -------------- An HTML attachment was scrubbed... URL:

Howdy, How do i perform a lookup from a remote odbc database in the asterisk dialplan? I can do it with mysql but not sure of commands for odbc connection. Cheers!!!

Hi. Is it possible to override the standard DB function in Asterisk? My dialplan contains a lot of calls to "Set(DB(...))" and "${DB(...)} which of course use astdb to store/read data. I would like to stop using astdb and switch to Clustered MySQL (I don't suppose "clustered astdb" exists?). So instead of rewriting extensions.conf and replacing the DB calls with

Is there any way to return multiple values from functions defined in func_odbc.conf? It appears that you can only return one value. True? Hope not.... Doug. ____________________________________________________________________________________ Be a better pen pal. Text or chat with friends inside Yahoo! Mail. See how. http://overview.mail.yahoo.com/ -------------- next part

Hi, I have a website where customers enter their phone numbers to be called. I'd like them to have to put in information and 'schedule' a call. 1) Call Immediately 2) Call in the next _ minutes 3) Call me tomorrow, same time. So, Asterisk will pull two variables from this php websites, $phonenumber and $timetocall. $timetocall will need to be calculated as

I'm trying to use func_odbc to write to a MS SQL db. Here's my func_odbc conf: [OPTIN] dsn=MSSQL-Optin write=INSERT into OptIn (orgID) values (${VAL1}) Dial Plan exten => +18665551212,n,Set(ODBC_OPTIN()=dave) When I do an odbc show, it shows that I am connected to the db. If I use isql, I can write to the db, however, when I use func_odbc, a record will not write. I'm using

Any know what Digium hasn't released the DS3 card? It was supposed to be out a while ago. -Matt

Is there a way to record your own voice messages ("Welcome to my PBX, Press 1 for ...") using asterisk and an analog phone, or do they need to be recorded using traditional voice recording software? Also, I am confused as to why my replies to the message board are never indented. Call this the ultimate newbie question, but how should the reply be worded so that I don't screw up

Hi All... I have a maddening problem... I have Asterisk configured to pick up a line after 4 rings. I do this to allow my fax machine to pick up a particular distinctive ring pattern, so I don't have to pay for a dedicated fax line. If someone calls the line, lets it ring 3 times and then hangs up, Asterisk answers the line, and holds it off hook forever, constantly playing the

Hi All; I can assign for my Asterisk Machine a two IP addresses (xxx.xxx.xxx.yyy and xxx.xxx.xxx.yyz), how can I use these two IP's so I can let one call sent with a source IP address xxx.xxx.xxx.yyy and another call to be sent with another source IP address xxx.xxx.xxx.yyz, I need this because I need the side to authorize my calls by the IP address, and some calls to be authorized with the

Hi Folks, Are all the astricon presentations up? I'm especially after the one that tilghman did. I caught the tail end of the prez when I decided to skip the session I was attending and go for that one. :)

Hi all, I am planning to implement LCR routing on my already running asterisk server. Uptill now i have found out that asterisk has no support for lcr, i have to do something about it myself, for example using the AGI. Im looking for ideas here. Whats the best way to start implementing lcr in asterisk. Should i use agi and start implementing my own lcr script or is there any plugin available which

Hello,I was using the ResetCDR() right after my IVR answered the calls to set them unanswered until or if somebody pick up the callI just upgraded to version 1.4.19.1 and now ... when the ResetCDR() function is called it seems to just erase the recordIs there another to set the status of the call unanswered ? is it a bug ?Thanks, -------------- next part -------------- An HTML attachment was