similar to: Sending Calls via SIP trunk from two different IP addresses from same Asterisk Machine

getting calls, but I can only send calls from my main machine IP address so I can't control where I am sending calls to. I am hoping to have this developped somehow (a per SIP peer bindaddr and bindport), even if it means some bounty. I can't imagine this being this difficult, so a few of us who need this putting a couple hundred dollar would probably do it. Mike > -----Original

Hi, I'm using gotoiftime on asterisk, but it seems  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, 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

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

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

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

Is it OK to send patch series to llvm-commits for review as a series of emails in the "git format-patch" / Linux kernel [PATCH nn/mm] format? http://linux.yyz.us/patch-format.html Or is some other format preferred? Thanks, Jay.

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

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 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

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.

We're using 2.2.13 with pigeonhole 0.4.3, in a clustered environment (maildir on netapp, dual dovecot instances where each server is both a proxy and a backend). Every now and then (once a month per user, maybe?), users will see a blank email in their inbox. Investigating further, and we will see that the only information recorded in the maildir file for the message is the Return-Path,

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

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

Hi, Whatever the verbosity level (even 0), my Asterisk console is full of "Really destroying SIP dialog" messages. Is there a way to get rid of those ? If not, do you think it deserves to marked as a bug ? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080918/54ad90fe/attachment.htm

In Asterisk 1.4.15 if I try to configure cdr_pgsql.conf , asterisk crash with this message asterisk: symbol lookup error: /usr/lib/asterisk/modules/cdr_pgsql.so: undefined symbol: PQescapeStringConn Is this a knowed error? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20071201/a3aecc1d/attachment.htm

If there are multiple NICs in the box, how do we specify the local IP address to be used for RTP? Anything in rtp.conf ? Thank you. Alex Zarubin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20030609/d151f190/attachment.htm