similar to: retreiving callid of call from the dial application

Hi, In a normal PBX environment a user usually calls in and IVR's are played according to a predefined dialplan. Iam trying to develop an application where asterisk dials out to a user and initiates an IVR instead (please note that the IVR is not static and may vary according to different condtions). Can someone guide me how this is possible using Asterisk. Do I need to write some sort of

Hi all, I've note that the variable DIALEDPEERNUMBER is broken. Now i want to know if exist another method to retreive the called number on *, and, if it's possible, an example ;) Regards.

Hi, Dovecot currently treats an LDAP user/password database the same way as a text or SQL based database: it just tries to retreive the (hashed) password for a given username. LDAP however has the capability to authenticate the user itself: dovecot could try to bind to LDAP with the given username and password, and if authentication succeeded, the LDAP server returns the other info (uid,

Hi, To combine two sources of CDR's I want Asterisk to save the SIP callid for all calls. I know there's a variable that contains the SIP CallID value, but is this the callid value of the incoming INVITE message or the outgoing message? Are they the same? (I've not yet checked a trace, I'm sorry for that). I've tried to read chan_sip, but couldn't find something in the

Hello everyone Is there a possibility to catch an outbound callleg ID for the follovong scenario: some carrier -----> ------(asterisk1) --->-----asterisk2 ? I can get inbound callid for asterisk1 with a ${SIPCALLID} in extensions.conf or to look it up in cdrs field (are the same). But how about outbound? I have all calls just forwarded through asterisk1, not answered and for every call I

Hello all, I tried the dev list, but got no answer at all. I'm facing some problems with call-id generation in a heavily loaded Asterisk Server. Asterisk is generating same call-id and from tag for different calls (and this is not desirable). Looking at the source code I noticed that rand() is used four times to get a callid. Is that safe enough? Maybe my system lacks of a good random

Hi, I have a custom agi which at times does not exit gracefull and crashes in between. The logging options are set to the maximum but I dont see something conclusive in the asterisk log. I have noticed it crash after issuing the "SAY NUMBER" and "GET DATA" agi commands and the agi is spawned with no apparent reason after that. I tried running the application locally and

Hi, I am facing this issue, where I get a delay of aroud five seconds when switching between contexts (through extension.conf) . This is how my extensions looks like. [salesivr] exten => _X.,1,NoOp(Incoming call from user ${EXTEN} and caller id ${CALLERID}) exten => _X.,2,Playback(emptyy) exten => _X.,3,Background(Main_Sales) exten => _X.,4,WaitExten(2) exten => _X.,5,Goto(_X.,3)

Hello. I'm installing new FreeBSD 6.2-RELEASE, based on intel machine. Firewall type is OPEN. I have Windows Server 2000 with Active Directory on it, working in Native mode. I've installed samba-3.0.23c_2,1 from /usr/ports/net/samba3 prefix=/usr/local without krb-1.5.1 being installed. Added: nmbd_enable="NO" smbd_enable="NO" winbindd_enable="YES" to

I am not getting any caller Id with my standard T-1. Is a standard "T" capable of sending callerid? I don't want to spend time troubleshooting my PBX if Asterisk can't send it down that type of trunk. Jordan -------------- next part -------------- An HTML attachment was scrubbed... URL:

I'm toying with adding a feature request to provide some sort of gain setting for voicemail when accessed from "certain" interfaces. Maybe something like voicemail=6.0 (db) within a specific channel section of zapata.conf corresponding to a pstn line. Situation: 1. Someone calls into asterisk and leaves a voicemail. The sound is recorded at some volume well below 0 db, and is

Hi All, as a followup to my previous posting (re: not getting caller id to display on a powertouch 350), I've found the following... If I hook the PT350 up to the PSTN line, caller id is diplayed properly... If I connect it to our CAS Acess Bank II, the 350 will not display incoming CID... If I hook up a cheap callerid enabled phone to the same port that the PT350 was on (on the AB II), the

Calling linphone from asterisk 13.9.1.: Dial(SIP/<user>@sip.linphone.org) And it works. But on the linphone side the caller is: <extno>@ipaddress or 2502 at 45.123.987.4 Is there any way to make it more descriptive, at least for the sip user name ? I tried setting SIPCALLID, which had no effect. Set(SIPCALLID=Office) Thanks, sean

Ok, I've got the trace translater running above unify at the moment. When I try to access the directory giving me trouble the following gets logged (note, I accessed one which worked first - /home/lozzar, and then mine /home/will): 2008-09-15 20:16:53 C [dict.c:1125:data_to_str] dict: @data=(nil) 2008-09-15 20:16:53 C [dict.c:1125:data_to_str] dict: @data=(nil) 2008-09-15 20:16:53 T

===================================================================== Asterisk-11.5.1 Centos6 app_confbrige.c ===================================================================== APP: MyConfbridgeCount(Confbridgename,variablename) it will return no of user in conference if conference is created or else zero. Task: Using Dailplan user want to retrive no of user in conference '6050'

This is release 1.0-rc1 of Wine, a free implementation of Windows on Unix. This is the first release candidate for Wine 1.0. Please give it a good testing to help us make 1.0 as good as possible. In particular please help us look for apps that used to work, but don't now. See http://wiki.winehq.org/PlatinumRegressionHunt for details. What's new in this release (see below for details):

Hello Anyone have a working copy of Fail2ban asterisk filter asterisk.conf for Asterisk 16 running PJSIP. I have tried 10 different filters but none of them show any matches when testing with fail2ban-regex I see date template hits but no matches.... My log [2019-06-06 15:37:20] NOTICE[18081] res_pjsip/pjsip_distributor.c: Request 'REGISTER' from '"2405" <sip:2405 at

Hello all, Has anyone have magicjack working with their asterisk? I had patched chan_sip.c with some code that allows asterisk to do the md5 hash that mjmd5 proxy does. * shows that it is registered with magicjack, but incoming calls are not even hitting my * box and outgoing calls get congestion. Here is my relevant configs. I did do a ton of google searching, but it all points to it should

Hola, Lo primero muchas gracias a todos por vuesta habitual ayuda. Llevo meses con las funciones lmer, pero tengo un problema no se como plantear unas comparaciones multiples una vez demostrado que el factor tiene diferencias significativas. Tengo multiples opciones, Solución A: una es hacer un bonferroni a saco (pero eso me chirria ya que tego muchos datos) Solución B: Otra es utilizar glht,

The Wine development release 1.2-rc3 is now available. What's new in this release (see below for details): - Many translation updates. - A lot of bug fixes. The source is available from the following locations: http://ibiblio.org/pub/linux/system/emulators/wine/wine-1.2-rc3.tar.bz2 http://prdownloads.sourceforge.net/wine/wine-1.2-rc3.tar.bz2 Binary packages for various