similar to: CDR feature not working properly for "failed call attempt"

Joshua, If I'm understanding this correctly, you're saying that the Playback is failing because it isn't connected to anything on the other end, because the Dial() failed. When the channel is created on the "OutgoingSpoolFailed" extension, what context is it created in, one of the origin legs? Is there a way detect this condition in the target context ([outbound-swift]),

I am using asterisk 1.4.21 with outgoing call files. I am call a line that is busy as you can see below. How can my AGI ask what the status of the last call was so I can tell if there was NO ANSWER or it was BUSY? Thanks, Jerry -- Attempting call on SIP/401 for smvoice_callprogress at smvoice-dialout:1 (Retry 1) -- Got SIP response 486 "Busy" back from 192.168.1.161

Hi, I copied a setup from an older 1.8.5 installation to an 11.15 installation, and I'm having problems getting call files to work. Here is the extension setup I'm using: [outbound-swift] exten => _[a-zA-Z].,1,Answer exten => _[a-zA-Z].,n,Playback(AAA/check_ip_failure) ;exten => _[a-zA-Z].,1,Swift("${EXTEN}") exten => _[a-zA-Z].,n,Goto(1) [mis-phone] exten =>

Hi all, I've got a program that creates a callfile and most if it working great. However, when a call fails, I'm trying to capture the reason, which I'm told should be in the ${REASON} channel variable. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Here is an excerpt from the callfile: Channel: local/155555555 Callerid:Tests <155555555> MaxRetries: 0 RetryTime:

I am using asterisk 1.4.41 and the AMI I am trying to execute a command over AMI, specifically "core show channels concise" "sometimes" I get this back: asterisk_command_show_channels() execute failed. 'Response: Follows[CR ][LF ]Privilege: Command[CR ][LF ]OutgoingSpoolFailed!smvoice-dialout!failed!1!Down!AGI!smvoice|-digium_failed!!!3!0!(None)[LF ]' I'm not

Hello: I have found the following problems with outgoing calls with asterisk, compiled with an updated CVS on 22 Oct. 1.- Problem with retries: Whenever I set the MaxRetries parameter, to something greater than 0 in a call-fille, Asterisk ignores the RetryTime parameter and retries every file in the outgoing folder when a new call-file is copied into that folder. So, if I make a call placing

I am getting this error from call manager (4.0) and asterisk 1.2.4 I have canreinvite=yes on the call manager setup. I can call into the asterisk box from call manager. THat seems to work. When I am calling out of the box using a call file I see this entry from call manager... What might be the problem with my setup? THanks, JErry ---------------- <Date>03/06/2006

I am sending this again since I haven't get it back for twelve hours: When I originate call to IAXComm, more or less one of tree calls fails for no aparent reason. Originating calls to SIP clients works as expected. Anybody has similar problems? Is it asterisk or client problem? Asterisk log: Jul 15 00:00:04 DEBUG[1179663]: manager.c:1018 process_message: Manager received command

Hi I am new to Linux. I am using Redhat 9.0. I have installed the Samba software, configured the smb.conf, added my client Windows XP box into the hosts.allow, hosts.equiv etc and started the smb service. Still I am unable to map the Linux share on to my Windows XP box. I have tried enabling and disabling the XP firewall also. I can see the Linux machine name in the Windows explorer "My

Using the php script below. I am able to enter my number and the number to call, however I get the following error: -- AGI Script cid-spoof.agi completed, returning 0 == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/manager_custom.conf': Found == Manager 'admin' logged on from 127.0.0.1 -- Executing Wait("OutgoingSpoolFailed",

Justin Killen wrote: <snip> > > Whenever I try to copy this callfile into /var/spool/asterisk/outgoing/ > I get these 3 lines repeating over and over (I?m not 100% sure which > entry is first): > > [2015-02-16 16:56:02] WARNING[9737][C-0000f8a7]: channel.c:5353 > set_format: Unable to find a codec translation path from (nothing) to (slin) > > [2015-02-16

Hi all, I'm using Asterisk 1.0.9-BRIstuffed-0.2.0-RC8o on box A with a TE410P (EuroISDN cpe) connected to another similar asterisk box B acting as EuroISDN master. I'm performing some load tests by contiously feeding up to concurrent 30 call files to /var/spool/asterisk/outgoing/ on box A which inititate via a dialplan context/extension a outbound call (redirected via chan_local) to

I AM PARVEEN KUMAR. I HAVE CONFIGURE SAMBA AS A PDC WITH TDBSAM DATABASE. I WANT TO APPLY A PASSWORD AGING FOR MY DOMAIN users. FOR THAT I HAVE USED PDBEDIT COMMAND. root# pdbedit -P "min password length" -C 8 account policy value for min password length was 5 account policy value for min password length is now 8 root# pdbedit -P "password history" -C 4 account policy

I have the following situation: I drop a call-file into the Asterisk spool directory and I get called back. That all works. And I have this script: #!/usr/bin/perl -w use Asterisk::AGI; my $AGI = new Asterisk::AGI; my %input = $AGI->ReadParse(); $AGI->answer(); my $i; $i = $AGI->channel_status(); $AGI->say_digits($i); $i =

I recently went through a system upgrade, which resulted in an upgrade of Samba to 3.0.28a. This is for a home network, so it isn't heavily taxed (only by family members), but I do use the environment as a sandbox of sorts for stuff that I plan to roll in to production in the office. I had some funky issues with authentication to start with after the upgrade, so I went through the