Discussion: Asterisk auto-dial out

Hi guys.
Please put the following string in "Minimal Call File Example" section:
"chown asterisk:asterisk hello-world.call"
right before "mv hello-world.call /var/spool/asterisk/outgoing/" command.
I believe if we warn users to make chown before attempting to move call files, it will save their time.
I spent about two hours to debug why asterisk can't dial out (( And I guess I'm not the only one.
I wouldn't read further this article(where solution is described), because I didn't get any results in beginning of the article.
Thanks in advance.

My employer has tasked me with getting call files to work in our Asterisk box. I've managed to get the system to call out, both to a phone number, and an extension, and play one of the pre-recorded files in Asterisk, with little trouble. The problem I am now having is, we want to be able to generate a message in text, plug it into a call file which will be dynamically generated, and have it call.

While I can create this file, and have it call, it never converts the text to speech. I have tried setting the "Application" to Festival, and Swift. In both cases, the system will make the call, then the instant the line is picked up, hang up. I have checked for permissions on /tmp/ (for Festival), made sure a /var/lib/asterisk/sounds/tts/ exists (for Swift) In both cases, it does the same thing, and immediately hangs up on answer.

Any help will be appreciated.
Thank you,
Jason A.

my call files callback set is:
"Channel: SIP/0$to@kani\r\n".
"CallerID: 0702662200\r\n".
"MaxRetries: 3\r\n".
"RetryTime: 60\r\n".
"WaitTime: 15\r\n".
"Context: dialout\r\n".
"Exten: 0$from\r\n".
"Priority: 1\r\n\r\n");

i set WaitTime is 15sec. but , it is not work, ringing over 15sec more, and into the mobile's voice mail. and asterisk record a none data voice message to my mobile voice mail >~<
is any thing wrong for my callback set!? some one can help me, please >~<

Hi,

I have followed the instructions here and tried to create .call files for both internal extension and external calls. The files have the correct permissions and disappear when copied to the outgoing directory but nothing then happens. I have looked through the logs and watched Asterisk CLI in the verbose mode and nothing appears. Is there somewhere else I can look for troubleshooting this?

Thanks very much for any help. I am a newbie at this so please excuse me if this has been answered before.

I have Asterisk 1.4.18 and i would like to make a auto-dial out through Zap channel using callingpres=prohib_not screened.
I tried to set up CallingPres using application SetCallerPres, but it doesn't work because it must be set before making a call.
Is there any way to make it except changing source code of chan_zap.c?

The steps should work the same no matter who your ITSP is, the important thing is that you DONT NEED a DIAL step in your extensions.conf.

But you DO NEED to format your call file as follows (the # to be called needs to be in the channel statement) :

Channel: SIP/3012345678@teliax
CallerID: Spiderman <3015551212>
MaxRetries: 1
RetryTime: 600
WaitTime: 30
Context: outgoing
Extension: spidey
Priority: 1

Then in extensions.conf you need to have an "outgoing" context with extension "spidey" as follows :

[outgoing]
exten => spidey,1,Answer()
exten => spidey,n,Wait(0.5)
exten => spidey,n,Playback(hello-from-spiderman)
exten => spidey,n,Hangup()

so you are calling (301) 234-5678 as Spiderman from (301) 555-1212.

Is it possible to run multiple applications from one call file using Application field, or this can be done only using Extension with multiple commands?
If i put two Application fields in one file, it seems to use the last one only, ie this one calls only playback , and if i swap them - only UserEvent :
<...>
Application: UserEvent
Data: <some data>
Application: Playback
Data: tt-monkeys

I had tried to add sip header by put

fputs($cf,"Set: SIP_HEADER(X-myACCT)=foo@bar.cc "\n");

before placing the .call to outgoing folder. but failed due to the SIP_HEADER is read-only.
Is there any possible solution to accomplish this request ? Thanks in advance :)

Suppose you place a .call file in the outgoing spool directory as you normally would — and set the maximum number of retries to 5. If the outgoing call is never answered (all 6 times, 1 try, and 5 retries), then the failed extension (priority #1) is jumped to after each of the "failed" calls. This is useful for CDR and such.

However, I find myself in a situation where all I really care about is whether the the .call file was removed from the spool directory because a call was successful.. or because it exceeded max retries.

Is there a way to do this? I'd have thought that there would be a special extension for the case of the final attempt of a call.

Hi - This page gives the following examples of channels that can be used in your call file:

Channel: Zap/1/1XXXXXXXXXXXX

Channel: Zap/g2/14109850123

Channel: SIP/bt101

Channel: Local/1000@from-internal

How do I know what channel is the best to use in a given suituation?