similar to: Originate to IAXComm problem once again

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]),

Hi Asterisk Developers/users, I am facing a problem while using the cdr feature of asterisk(version asterisk1.4.24.1). Whenever I make a call using a ?*.call? file and it gets failed , it don't produce the CDR for that channel as it falls into ?OutgoingSpoolFailed? channel As there is no such channel defined for ?OutgoingSpoolfailed?.. I am using this line in extension.conf for capturing the

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

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 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:

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 =>

I have Meetme conferencing compiled for Debian as per http://powerontech.com/freepbx-on-debian.htm . I drop a callfile in the outgoing dir, and it intitiates a call to a local extension as a channel, but the call seems to block on the Meetme() command. That extension completes the outgoing Dial(SIP) command to my phone, announcing that leg is the only member of the conference, and just waits. If I

Is it well known that setting the ActionID when connecting to AMI has absolutely no effect? Is this fixed in Asterisk 1.4? If you add an ActionID to your Originate command for example, it looks like the only events that come back with an ActionID associated are the initial response, OriginateSuccess and OriginateFailure. That's it. No other events have an ActionID associated. This pretty much

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've discovered that when dialing out using API's Originate action, a no answer is considered a failed attempt, while a busy is considered a successful attempt. The problem I'm having is that when I dial an invalid number, say a disconnected number that gives a fast busy, my CDRs are identical to those generated by a no answer attempt. Is there a way to distinguish between a no

hi, afaik, the action-id provided with the OriginateAction should only show up in the OriginateSuccess or OriginateFailure event. Intermediate events that are generated when the channels are create will NOT carry the action-id of the originate. The async flag tells asterisk to process originates in parallel, i.e. if you have two users originating calls and NO async flag set, the second originate

Apparently there are subtle diferences between meaning of MeetmeJoin event depending on channel type. Problem is: after originating a call from channel to MeetMe room i.e.: [meetme] exten => 1,1,Answer exten => 1,2,Meetme(kolejka|dqM) than: Context: meetme Exten: 1 Priority: 1 ActionID: 1077925740-00000004 Timeout: 5000 Action: Originate Async: true Channel: somechannel I get eventually

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

Everythings works great with asterisk exept one feature with redirect : it doesn't redirect when ringing ... BTW are their any plans to extend the manager API ?? Michael Devenijn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20031212/1200012e/attachment.htm

Hi, This morning I had some failed calls. On the console (and in the log) I saw the error "Unable to allocate channel structure". Before I restarted the process, I checked it's memory usage in ps and glanced at my free memory in top. Asterisk was using a normal ammount of memory, about 40M. I don't think this was a system limit. This was running Asterisk v1.0.2. Below is

Hello, I have an Asterisk box with a TE410P connected to a PRI line and agents with X-Lite installed on the same LAN as the Asterisk server. Sometimes, when I make outbound calls it hangs up as soon as other party tries to picks up the call. Does someone ever experienced this situation? On X-Lite, only G711-ulaw is enabled and here is what i put in sip.conf: [4001] type=friend username=4001

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

Having some trouble with segfaults and sound quality all of a sudden (since I recompiled from the latest source) when 2 iaxComm clients connect. First off immediately after the server reports: <> <> -- Attempting native bridge of IAX2/4587@10.9.1.32:4569/1 and IAX2/4589/5 <> <>One or both client may sometimes segfault. Additionally, when they do get properly connected,