similar to: agi never leg1 disconnect

Hello, im having trouble with the following: [Asterisk]<------>[ISP]<------>[ADSL Modem]<------>[Linksys Router]<------>[Grandstream ATA]<------>[Analog Phone] On server: - Asterisk 1.6 - A2Billing 1.4 A2Billing have 2 Trunks: - TrExt: Voip Provider - TrInt: Internal Calls This structure works on first day (Asterisk+A2Billing installation/configuration). But on

Here i am doing a dial command from a php agi... EXEC DIAL H323/123456789@xx.xx.xx.xx:1720|40|HL(585000:61000:30000) But asterisk is not disconnecting the connection after 585 secs... the result is ... answered time is 1926n but thing is time out is working some time and some time not.... LOG: 2005-06-28 20:26:13 VERBOSE[19094] logger.c: callcard.php: string(111) "app_callingcard:

Following up on this, I've figured out what's going on. What I'm not sure of is how best to address the situation. I'd like to avoid going down to the MIB level because I just don't understand it. "Split phase" output devices seem to fall into an undefined area. They're not three phase, but require additional MIB values so APC uses their 3-phase MIB

Hi! Maybe someone could help me out? When a call is routed via a2billing AGI and user does a transfer, the call is dropped. If the trunk is called directly everyhing works. Here's a direct scenario (working fine): [pbx000001] exten => 101,1,Set(__TRANSFER_CONTEXT=pbx000001) exten => 101,n,Dial(SIP/pozitel/37129238254,45,t) exten => 102,1,Dial(SIP/12345,60) so, when user calls ext

Perhaps I should have submitted this as a bug so that it would be officially tracked. It's not a big deal, but here it is again (I can't remember which version this patch is against, but I don't think legend() has changed since then ...) Basically, the problem is that if you want to have "opaque" points that overlay lines (rather than using type="b" and having

Greetings. I have next feature in features.conf : send => *9,peer/both,AGI,/etc/asterisk/agi/map_mail.pl What it does is parsing CALLERID and DNID from AGI input, performing some actions in MySQL with these values, and then running application for peer (for example, PlayBack) Sounds simple, and it really is. When my user is receiving a call (we are the B leg) and presses *9, everything

I am working on a call-back solution where the initiating call should never be answered. I was doing this simply through the dial plan, sending a progress tone, and then dumping the channel, and firing off a DeadAGI which created a call file to make the callback. Now I've tried extending this so that an AGI is fired first to check for things - like no inbound ANI - and play a

Hello I'm using an AGI script in Lua to make a callback through Zaptel. For this to work, I must wait until the channel is idle, or I get this kind of error, even after waiting over 10 seconds after the remote end rings once and hangs up: ============== channel.c:2863 __ast_request_and_dial: Unable to request channel Zap/1/123456 pbx_spool.c:341 attempt_thread: Call failed to go through,

Hi, I am having a strange problem with enum and AGI. Here is what happens: I have in my agi something like that: foreach my $resolver ("e164.arpa", "e164.info", "e164.org") { my @enums = get_enums($phone, $resolver); foreach my $enum (@enums) { $dialstring = $enum .

I'm confused about something.... In Asterisk 1.4 you can collect RTP QoS metrics at the end of a call with: ${CHANNEL(rtpqos,audio,all)} Now, when your using the AMI to do a callout, like this... ACTION: Originate Async: yes Timeout: 60000 Exten: callback Channel: SIP/1000 Variable: callid=849120 Variable: destination=SIP/1001 Variable: timeout=70000 Variable: timeout_warning=60000

Hi all, I'm trying to build an IVR using the Perl AGI module at http://search.cpan.org/~jamesgol/asterisk-perl-0.10/lib/Asterisk/AGI.pm But, I'm having trouble getting my program to play a message and wait for a keystroke. I am able to use this code to play the file, so I know that the $msg variable points to a valid sound file: $result = $agi->exec("background $msg");

sorry for top-posting, the two topics started with 2 different reason subject, but then we finished on the same problem. btw,the 2 show channel are reported above: the channel with DTMF working kcenter*CLI> core show channel SIP/pbx2-000004b9 -- General -- Name: SIP/pbx2-000004b9 Type: SIP UniqueID: 1467323106.1275 Caller ID: xxxx Caller ID Name: xxxx

Hi there, I'm converting a dialplan callback type application to fastagi as I'm hitting the buffers with respects to getting useful results from CDRs. It works by a spool call file triggering a Local extension, that extension then does the first dial to a client. I dial to a local context from the spool file as I need proper return codes as in ${DIALSTATUS} which are not available

Hello All, I manage to configure Asterisk RealTime and now it loads the SIP users/peers from MySQL DB. The table I am using is of A2Billing DB "cc_sip_buddies". Now the only problem I am facing is incoming calls are failing... The ATA which is assigned this DID number is behind NAT and according to Olle's explanations he said "*there's no support for NAT keep-alives

Ok, this one is driving me nuts. I am using smbk5pwd to sync samba, ldap, and heimdal kerberos passwords. I have the overlay working if called from the ldappasswd command. However, if I set "ldap passwd sync = only", I get no joy from Samba. It does a series of searches, but doesn't send any kind of password update request to the server. If I use ldap passwd sync = yes, then

Hi, I've got a setup whereby calls come into the asterisk server (1.2.7.1) over a IAX2 trunk and into a dialplan that launches a php AGI script: [live-full] exten => _X.,1,Set(TIMEOUT(absolute)=0) exten => _X.,2,NoOp(${EXTEN}) exten => _X.,3,DEADAGI(live-full.php) exten => _X.,4,Wait,2 exten => _X.,5,Hangup The script is using phpagi-2 from http://phpagi.sourceforge.net/ and

Requirements: 1. http://sourceforge.net/projects/odbcsock 2. The SQL server must be in the same subnet as your * server Howto: 1. Install ODBCSocketServer on your SQL server and verify connection with the included VB COM app from a Windows box. 2. Decide how you want to transform the Caller ID. In my case, I want to do a lookup of the CallerID number in SQL server and prepend the Caller ID

Regards! During the use of areski a2billing software I'm getting same problem all the time. Actually, after 15 minutes of speaking to someone over calling card, connection brakes. Installation was as smooth as it could be so I don't think I made same kind of a mess in that domain. This is the only problem in the aplication. In the logs everything seems to be fine. I'am sending You

Greetings, Rowland Penny! >>> OK, what does running this command on the DC show: >>> ldbsearch -H /var/lib/samba/private/sam.ldb >>> '(objectSID=S-1-5-21-1031481445-3291699540-3997755762-61000)' | grep >>> 'uidNumber' >>> This relies on ldb-tools being installed and sam.ldb being in >>> '/var/lib/samba/private' if yours

On 04/04/15 18:28, Andrey Repin wrote: > Greetings, Rowland Penny! > >>> # cat /etc/resolv.conf >>> nameserver 192.168.17.4 >>> search ads.ccenter.lan >>> >>> # host -t SRV _ldap._tcp.ads.ccenter.lan. >>> _ldap._tcp.ads.ccenter.lan has SRV record 0 100 389 dc1.ads.ccenter.lan. >>> >>> # nslookup dc1 >>> Server: