[Asterisk-Users] Re: Asterisk-Users Digest, Vol 12, Issue 144
On Thursday 21 July 2005 17:42, [EMAIL PROTECTED] wrote: Thanks Adam. This helps some, but I'm still not sure how you mean for me to acheive 1). I would have to perform a Dial-command no matter what, so I guess I would have to make an interruption from the manager API, but I don't manage to find a command that will acomplish that. Either use the dial parameters to play a sound on a regular basis, eg, you could play $UNIQUEID every 30 seconds, then your other software which is watching the manage interface and deducting money every 30 seconds can change the content of that file once you want the user to start to hear something different see the L option to dial. This may actually be a good idea. I have looked at L, and it seems to be the only documented way to get sound into an ongoing call. I'm actually a bit surprised that that feature is missing. A good thing Olle just announced a code freeze on 1.2... Alternatively, you need to be more creative and put the two calls into a meetme conference, then add your third channel (see the local channel driver) which simply plays whatever audio you need (or in fact this could be some AGI/etc)... These are just some comments I felt like making, I've never had to do this, and this is not necesarily how I would do it if I did need to (ie, if I was being paid to do this, I'd think about it more before implementing it, but for now, I can shoot my mouth off without any concern of needing to deliver on what I've said can be done). Thanks for the shooting :) I need to do this (and I'm paid too!), so I'll do some more thinking. Best regards -- Eivind Trondsen People are destined to be party animals, and the technology will follow - Linus Torvalds ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Re: Working with an ongoing call
1) send sound to the caller of an ongoing call 2) retain control so the call can be terminated based on a timer (or whatever) Any tips would be greatly appreciated! Thanks in advance. Use the manager API to terminate the call if their credit reaches zero, connect and process active channels on an regular basis (as needed), use the AGI to reduce the credit by the needed amount at the end of the call (from h extension, or g option to Dial). Thanks Adam. This helps some, but I'm still not sure how you mean for me to acheive 1). I would have to perform a Dial-command no matter what, so I guess I would have to make an interruption from the manager API, but I don't manage to find a command that will acomplish that. Regards -- Eivind Trondsen People are destined to be party animals, and the technology will follow - Linus Torvalds ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Working with an ongoing call
Hi list
I plan to implement a prepaid solution where the system needs to check for
remaining credit periodically during a call. The reason for this is that this
is a system where the credit pool can be used simultaneously by more people,
and not only for calling.
I have a problem figuring how to be able to run logic while a call is in
progress. The L(x:y:z) option to Dial() is good, but not quite what I need.
In both the dialplan an an AGI the Dial command blocks, so what do I do?
I have not yet tried a multi-threaded AGI, but assume the possibility of
success with that scenario to be slim...
Ideally; I want to code things like this:
Fetch cost of requested call;
# The reserve functions also supplies total remaining credit
unless (Reserve credit for N seconds) exit with message;
INITIATE CALL;
If answered {
while(1) {
Wait for N-x seconds;
Reserve credit for N seconds;
if (close to credit limit) PLAY WARNING BEEP;
else if (out of credit) EXIT WITH MESSAGE;
}
}
hangup_trap:
Commit credit based on actual call length;
EOF
I realize that this probably needs to be done as a combination of dialplan
logic and AGIs, but my main concern is the ability to
1) send sound to the caller of an ongoing call
2) retain control so the call can be terminated based on a timer (or whatever)
Any tips would be greatly appreciated! Thanks in advance.
--
Eivind Trondsen
People are destined to be party animals,
and the technology will follow
- Linus Torvalds
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[Asterisk-Users] Distributed organizations - large scale public sector rollout
Hi List I am working with a pilot project for a Norwegian regional government to evaluate Asterisk for a large number of sites and users. The goal of the project is to have a unified VoIP-system to replace the disorganized collection of legacy PBX in use today. By distributed organization I mean an organization that consists of many, dispersed units, each with it's own existing telephony system, and with distinct number series. The goals of a unified system are several: - Lower traffic cost through a common backbone between sites and a common exit-point to the PSTN (either via IP or legacy lines). - Lower admin cost through unified, centralized management. - Added value through rollout of applications (voicemail, conferencing, IVR) that become globally available in the system. My main concern is manageability. From what I have seen of the available management tools there are none that address the needs of a distributed system. They all seems aimed at the SMB market, and don't leverage resources such as LDAP directories. Does anyone have any experience with management tools for Asterisk in a similar scenario? I am also very interrested in getting in touch with people working in similar projects. There is a large political element in rolling out Open Source telephony on such a scale, and having a network of similar projects could be of great value. I hope to be able to post to this list on our progress. Best regards -- Eivind Trondsen Wingnut Information Systems Norway ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] FW: Getting PHP Config to work?
Richard Folwell wrote: Look at WinSCP: snip It is (almost) worth installing Windows just to be able to use it. :-) If anyone knows of anything similar that runs under Linux please enlighten me! Have a look at the fish io-slave for KDE. Type fish://[EMAIL PROTECTED] in your Konqueror URL-bar and see what happens :) -- Eivind Trondsen | IT-infrastruktur LinuxLabs AS| IP-telefoni | Fri programvare ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[Asterisk-Users] Multiple incomming contexts
Hi list I'm trying to implement sourcerouting on a distributed installation, but I can't get contexts to work right. My goal is to do a Dial([EMAIL PROTECTED]) and vary the somecontext based on different criteria. This is going on over trunked IAX2 links. How do I set up my IAX-accounts to manage this? I have tried to play around with 'context' and 'peercontext' on the server being dialed, but no luck. Is it legal to have multiple 'context' lines in one object? Is what I'm trying to do possible? Any help appreciated. Eivind Trondsen LinuxLabs AS -- Eivind TrondsenTlf: +47 23 89 71 85 LinuxLabs AS Mob: +47 928 40 009 --- http://www.linuxlabs.no--- Drift - Overvåkning - Rådgivning ___ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
