similar to: Solved: [SetAccount in extensions.conf]

I'm using Asterisk 1.4 and I'm wanting to set an account code for incoming calls. In the extensions.conf file I have the following: exten => s,1,SetAccount(1234) exten => s,n,Dial(SIP/1234) Then when I dial the extension the following error message pops up in the CLI: [Jun 6 19:12:40] WARNING[28167]: pbx.c:1783 pbx_extension_helper: No application 'SetAccount' for

Hi List; How to upgrade the Asterisk, Zaptel and LibPri and keep the configuration the same? I do not need to remove current asterisk, zaptel and libpri and download new one and write new configuration. Regards, -------------- ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 00965 9849460

My asterisk should follow 302 redirect which it receives from other sip server(10.10.10.10). By running network sniffer I see, that asterisk receives 302 answer, but doesn't follow it. My config is: sip.conf: ....... [out4] type=peer host=10.10.10.10 canreinvite=no promiscredir=yes insecure=very disallow=all allow=g729 allow=g723 ....... extensions.conf: [to-sip] exten => _0011X., 1,

Hi List; Any one can advise me to a good link to see and buy Polycom IP Phones? Also, if I need support (in case the Phone was damaged and need to replace, so the warantee), so which web can provide that? I do not need to buy from one and he is not responsible for support. Regards Bilal ____________________________________________________________________________________ Be a better

Hi List; I need to use an prepaid billing system with Asterisk, and I do not know which one is more stable and integrated with Asterisk? A2Billing or AstBill or ASTCC? Also, from where I can download it and ready about its configuration? Regards ITS IP Telephony and Contact Center Engineer Eng. Bilal Ghayad Mobile: 009659849460

Hi List; Where I determine the codec to be used for the SIP Trunk (between Asterik and another SIP softswitch)? Regards Bilal ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi! I have a situation where we want to set up two different rails applications on the same server and they both have ferret related functionality that needs to be implemented. How does one setup the ferret_server.yml file and then start each ferrt server to reflect that these two applications should access specific ferret servers that are running? I''ve tried changing the ports in

Hi all Can centos have way to limit users to use resouce in the computer? thank you ____________________________________________________________________________________ Be a better Heartthrob. Get better relationship answers from someone who knows. Yahoo! Answers - Check it out. http://answers.yahoo.com/dir/?link=list&sid=396545433

Hi, As I recall, 3.0.25a creates it's OWN krb5.conf file based on info it gets back from the DC to try to handle site stuff (so it uses the 'nearest' kdc, etc). I forget exactly how this mech. works, but if the kdc returned with the site info (which subsequently gets built into samba's personal 'krb5.conf' file) is down, or replication is off, your kinit would work,

I am trying to retrieve the "dialed peer number" but it seems that ${DIALEDPEERNUMBER} is "broken". Also, I know that I could extract the dialed number from the ${CHANNEL} variable but this only works for SIP and maybe IAX (untested). However, it doesn't work for ZAP. All I get when using ZAP is something like "Zap/1-1" (for SIP I would get

Dear List; How can I place a call via Zap/g1 (group) but need to determine the line (FXO port) that will go via it? Also, how it will be possible to assign an dedicated line (connected to FXO) to an button on the Polycom IP Phone or Broadtel IP Phone, so if user select that button then he will be sure that his outside call will be via that specific line. Regards Bilal

Hi List; If I need to do a trunk between Asterisk and another SIP softswitch (so Asterisk will send a SIP calls to that softswitch), then I have to configure this on the sip.conf file or where exactly? And is it the same when I configure iax trunk? Should I determine the context in this case for this SIP trunk? Regards Bilal

Hi, I setup a queue (number 4050) with one static agent (extension 4054). What I would like is that when someone calls the 4050 queue and there are neither "dynamic" agents logged in nor is the static agent 4054 "on-line" then the caller gets out of the queue and falls into another context (eg. voicemail or anything). Not "on-line" means that either the SIP

My cdrs are missing accountcodes for incoming calls from other asterisk servers.. I've seen a few people mentioning this on the list and the solution seems to be setting up a dialplan for incoming calls from a particular sip peer.. in my opinion this does not scale well at all and I am looking for a solution to correct this problem. example sip peer: [asterisk_gw] type=friend

Hi, I want to assign different accountcodes (for billing) according to the IP address and or the H.323 name (chan_oh323). I tried in extensions.conf something like setVar(ACCOUNTCODE=userid) but in cdr I find the accountcode set in oh323.conf. Howto change it in extensions.conf? Roger.

i am set var Set(CDR(accountcode)=forkcdr-test) into agiphp probe $agi->exec('Set(CDR(accountcode)=55555)'); $agi->exec('SetAccount','123123123'); and no work ... how to solutions. thanks people!!!!! -------------- next part -------------- An HTML attachment was scrubbed... URL:

Both of our production asterisk servers are dumping core when making writes to our cdr tables. Here is a backtrace of the problems we are having: #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at util.c:347 347 if (tds_ctx && tds_ctx->err_handler) { (gdb) bt #0 0x00447b1f in tdserror (tds_ctx=0x1, tds=0xb7938c90, msgno=20004, errnum=9) at

hello perl experts i am working with "ast-rad-acc.pl" from http://www.voip-info.org/tiki-index.php?page=PortaOne+Radius+auth i dont know why $cdr{'DNID'} and $cdr{'CALLERID'} under 'sub send_acc {' are empty. i m successfully connected with asterisk manager and when call i hangup my perl application is getting that all other thing are ok but i dont know why only

Hi, I have problem with setting up a conferences. When I dial the reserved conference number from xlite the line gets hunged up and on a console there is a following message: WARNING[5924]: pbx.c:1783 pbx_extension_helper: No application 'MeetMe' for extension (internal, 1234, 3) exten => 1234,1,Answer() exten => 1234,4,MeetMe(1234|Md) exten => 1234,101,HangUp()

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 i fixed wrong capi.conf (there was [controller1] after [interfaces]) now capi.conf is: ; ; CAPI config ; [general] nationalprefix=0 internationalprefix=00 rxgain=0.8 txgain=0.8 [interfaces] msn=855285,859609 incomingmsn=* controller=1,2,3,4 softdtmf=0 accountcode= context=local ;echosquelch=1 ;echocancel=yes ;echotail=64 ;callgroup=1