similar to: Extenxions Optimization

Hi a new small question ;=) We have two Asterisk, connected in IAX2. On the first, in dialplan, we have: exten => _XX.,1,Set(IAXVAR(ACCOUNTID)=${CDR(accountcode)}) we sent into the IAXVAR "ACCOUNTID" the accountcode. On the second, in dialplan, we have: exten => 18,2,AGI(Caller-ID.agi,${IAXVAR(ACCOUNTID)}) That's work, the second server get the variable. I

Hi all, I am using Digium PCI board to receive PSTN call through regular phone line. It is no problem for me to receive calls, but I am not able to obtain the Caller ID if the calls are from the phone line. exten => s,1,Answer() exten => s, n, Verbose(1|incoming number is ${CHANNEL} calling to ${EXTEN} routing to ${phonenum} ) exten => s,n, Verbose(1|callid is ${CALLID(num)}) exten

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi list, is there a way to use dovecot as proxy for m$ exchange mailboxes? Currently, our clients not only can login to their mailboxes with their e-mail address but also with their accountid. Both the clients e-mail address and accountid are stored in ldap. Now I need a way to let dovecot rewriting the client's accountid to the respective

Hi all, maybe someone on the dev list can jump in here too I have run into the wellknown problem with IAX about passing variables between servers. In my current 1.0.x network with IAX peers and Queues -- passing some extra variable information between Queue member channels is obviously not going to work. Googling shows some old discussions of an IAXVAR patch ......but I can't seem to find

Hi list , someone on the list has seen this type of connection attempts in asterisk, fail2ban does not stop 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="1420750787-386840",Severity="Informational",Service="SIP",EventVersion="1",AccountID="sip:100 at

Hello; Did you remember to uncomment the dateformat in /etc/asterisk/logger.conf? That's necessary for fail2ban to work. Logger.conf [general] dateformat=%F %T Regards; John -----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of ricky gutierrez Sent: Thursday, January 08, 2015 4:38 PM To: Asterisk

I've noticed that the Asterisk (v11) security log captures attempts do dial without first authenticating, and places the number dialed into the "accountid" field. I'm trying to distinguish between failed attempts to register and attempts to dial without registering, but the security log treats them identically (using the accountid field for either the username or number

Hi all, I''ve got a rails and database question. Accounts have jobs, and job numbers should be assigned per account. Account 89 should have jobs 1-whatever, which are not the same as account 67, which has jobs 1-whatever. In the past, I have accomplished this quite easily with: create table job ( accountid int(10) unsigned not null default 0, jobid int(10) unsigned not null

My Google-Fu skills have failed me, I have not been able to find a solution to the problem I am facing. asterisk + from + asterisk + options + qualify != what I am looking for -- When qualify is enabled on a trunk, the From line shows "asterisk". See the SIP message below. I would like to keep qualify enabled without sending the other end any reference to "asterisk". Can

On 01/08/2015 11:37 PM, ricky gutierrez wrote: > Hi list , someone on the list has seen this type of connection > attempts in asterisk, fail2ban does not stop > > 2015-01-08 14:59:47] SECURITY[21515] res_security_log.c: >

I recently upgraded from Asterisk 13.19 to 16.6.1. Everything is working fine with a few minor tweaks except outgoinf fax. Incoming works fine. I do outgoing faxing through an AMI call. Here is the output from the security log: [Nov 27 06:16:05] SECURITY[101222] res_security_log.c:

Hi list, I would like to now what is the sense of such type of entry in security.log [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: SecurityEvent="ChallengeSent",EventTV="2019-09-27T15:12:24.181+0200",Severity="Informational",Servic e="PJSIP",EventVersion="1",AccountID="<unknown>",

I have been racking my brain attempting to get the remote callerid information for calls made to extensions on another Asterisk system connected via IAX2 but nothing has worked. To clarify, I would like to display the number AND name on the calling phone when calling extensions on another Asterisk system. I seem to be able to 'send' all the information I want to the system I am calling but

Hello: I have a series of newspaper articles from a Canadian newspaper database (Canadian Newsstand) that look just like below. I've read through this vignette (http://cran.r-project.org/web/packages/tm/vignettes/extensions.pdf) about creating a custom reader to extract meta-data, but I can't understand how to apply this in the context of a text document, rather than in the tabular format

Le 30/09/2019 à 11:45, Joshua C. Colp a écrit : > On Fri, Sep 27, 2019, at 11:31 AM, Administrator TOOTAI wrote: >> Hi list, >> >> I would like to now what is the sense of such type of entry in security.log >> >> [2019-09-27 15:12:24] SECURITY[26964] res_security_log.c: >>

Hello, I am trying to produce some graphics to visualize my data. I think I want histograms. I have a grouping variable that has 48 different groups. I would like to produce a graphic that contains three of of these groups in the same device (they are time steps). I would like create graphics for about 6 different variables (6 variables X 16 groupings of there = 96 graphics). I am unsure of how

I have an existing openldap schema which is handling mail, web and ftp services right now. I am trying to get a windows machine talking to the same filesystem as apache on linux via samba and read/write using the correct uid/gid. I was trying to shy away from using pam_ldap as there is no need to tie the user in ldap directly to the filesystem. The problem is it looks like the samba ldap module

Hi All I have a problem with mixmonitor in 13.4.0 doing the following: 1. Caller phones in 2. Reception picks up 3. Talks to caller 4. Does attended transfer, talks to manager to screen the caller wanting to speak to him 5. Complete the transfer by putting down her handset so the caller can speak to the manager 6. Caller talks to the manager The problem is that mixmonitor only records

Ok, Please, check your manager.conf and logger.conf for any clue about debugging options, into the Asterisk configuration directory. El 26 mar. 2017 14:52, "Telium Technical Support" <support at telium.ca> escribi?: > I tried that but it had no effect. Still see things like: > > > > [2017-03-26 13:49:39] DEBUG[2088]: manager.c:5693 match_filter: Examining >

Hello, I have a problem with a cisco GW, if i only set g711 ulaw or alow as codec in my ata the the GW return a media not acceptable error. but If i add the g729 codec the all is ok. I see in the config of the cisco where to define codec for imcoming call but not for outgoing *Jul 17 15:57:02.604: Received: INVITE sip:0041787518551 at 192.168.0.110 SIP/2.0 Via: SIP/2.0/UDP