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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

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

In what appears to be the most current documentation of FFA I'm directed to http://www.digium.com/en/docs/FAX/faa-download.php However the download selector utility found there "ain't working". Specifically, none of the drop lists appear to be populated (in any browser, on any platform), thereby preventing any actual 'selection'. Has anyone else noticed this? Is there

I understand that standalone macros have been deprecated in 1.6 for gosub routines. I've been working on converting them all but was wondering about dial macros - it doesn't look like there's a replacement yet to call a gosub routine from the dial command. Or am I looking at this wrong? hose

I've been using 1.6.2 for a few weeks and I've managed to get almost everything working perfectly. I can't get the MWI indicators on my Aastra phones to work properly, the did in all the versions of 1.2 I used up to the most recent one, but now they work correctly right after the phone is re-started and rarely thereafter. it's as if something changed in the way the MWI is

Hello, Upgraded from 1.6.1.0 to 1.6.1.1 and my voicemail setup does not work anymore. I use ODBC storage for voicemail. Comes out that the "voicemessages" table schema should have changed, because the log says Asterisk needed to store data to an additional field called "flag". Any new message cannot be saved. The thing is that I'd like to know where I can find an updated

Hello ! Are there any plans at Digium to include also german voice prompts ? Thanks regards Hans

Dear All, i need help on Shared channel variable can any body have example of SHARED function which implemented in 1.6 version i can not find example regars Dhaval -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090723/56ed36c3/attachment.htm

Dear All Can you please let us know how to configure Asterisk to recognize extensions starting with the hash key ? Regards H.Motamedi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090801/4be38941/attachment.htm

I'm trying to figure out which 1.6 releases have the fix I'm looking for by reading the ChangeLog for each release, whether it's in the 1.6.0, 1.6.1 branches, or an -rc release. If I look at the latest -rc releases of 1.6.0 and 1.6.1 (which are 1.6.0.11-rc2 and 1.6.1.3-rc1 respectively), will that be an exhaustive list of changes or not? The reason being I'm still waiting on the

Hi all I am just confused as to why Asterisk appends a strange number after the Channel and Extension number. This moght be obvious to some people but I have no idea why?? Eg: IAX2/100-9123 Where does the -9123 come from?? and can I ditch it? I have made a work around to substring it out using SQL for when doing logs, but I would like to know what it is and if I can remove it. (I am using

Hello, So much keeps changing with the dialplan and Realtime lookups. Just downloaded the latest stable 1.6.1.2. The app_realtime, which was perfectly brilliant and did exactly what I needed, is gone; replaced with func_realtime. The REALTIME function is unacceptable: ; Get the conference number from the user exten => s,n(readconfno),Read(USER_CONFNO,conf-getconfno,0,3,20) ; See if

Hello, I need CAP_FOWNER=ep for the asterisk process, i set it with setcap on the file /usr/sbin/asterisk, it's there when i look on it with getcap, but after starting and loocking with getpcaps there's only cap_net_admin+ep set. So how exactly do I set CAP_FOWNER? Do I have to patch and recompile or is there another solution I did not see yet? thanks, best -- Raimund Sacherer

Hellow, i need the following requirements with asterisk : 1) Can ACD (Automatic Call Distribution) service work with asterisk, and how to set up ACD in asterisk ? 2) How call barging can set up in asterisk ? 3) How call recording can set up in asterisk? Thanks mahboob System Engr SSL -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello all, I'm trying to activate (on Asterisk 1.6.0.13) the cdr_mysql addon, but without success. Is there a proper online manual that describes all the steps to follow and debugging/monitoring information? When I type in the CLI "module show", cdr_addon_mysql.so is not listed, although in modules.conf, I added the line "load => cdr_addon_mysql.so". I also tried

Greetings- I'm wondering if the Digium PRI cards can be used for data (Cisco HDLC, PPP, etc) or if they're for voice circuits only. I haven't been able to find any information on this. All documentation direct from Digium seems to indicate their hardware is for voice applications only. Sangoma's cards work in either voice or data mode but of course this is configured in their

Hello, I try to move our asterisk installation (3 Asterisk servers in different offices connected using IAX and a lot of SIP phones, as well as ISDN connections using CAPI) to use G.722 instead of G.711. Asterisk 1.4.25.1 is used with the G.722 patch (the fixed one, which solves the gain problem). So SIP-to-SIP and to ISDN there is no problem. G.722 itself works and transconding to G.711 for

Hi, I've just had to enable the requirecalltoken=no option in iax.conf for one of my IAX2 trunks, and I don't think it works properly in the realtime version. I've created the requirecalltoken field in my (Postgres via ODBC) database, type text, and have variously tried it with 'yes', 'no' and 'auto' in the field. But the setting never seems to be used and

Hello, I bought paid for and used to use ChanSkype for skype connectivity. Now I tried to download the software again after the my old server got retired and someone else is squatting on their website. Can someone give me the latest ChanSkype binaries? Why do I have to rebuy the functionality I've already paid for too? Is digium going to feel for all the people who have to re-buy what they