similar to: Anyone played with app_amd?

Hi people! Sorry for bringing up some annoying issue.. yes, it's AMD again... But I was searching the last days for a solution for my problem and didn't really find anything. Now I'm hoping that someone of you has maybe an idea for me. :) My setup: --------- I use the Asterik Manager API to generate outgoing calls (by using "Originate" messages). These outgoing calls

Hi - I have been developing an auto-dialling application similar to Voiceshot, Call-em-all etc. The one thing I am now struggling to get working is the ability to leave a message on an answering machine or cell phone voicemail. I am using app_amd.c and while it works well for some phones it is proving to be very difficult tweaking the settings to get it to work reliable enough to go to

Hi Asterisk users, I'm one of Asterisk ARI users, and trying to designing the new ARI for application execution in Stasis(). This will be made possible for executing the applications in the Stasis() application. But, before going further, I would like to know which application needs to be considered. Because this feature will introduce new Stasis behavior, I would like to test the

Hi All, I am trying to use the AMD (Answering Machine Detect). But it is not sending the AMD_Status as either the Human or Machine, it hangs up in middle. can any one suggest us, what might be the problem and possible solution to it. below is the log -- Executing AMD("SIP/sip-ffe0", "") in new stack -- AMD: SIP/sip-ffe0 14082284927 (null) (Fmt: 4) Apr 23 08:00:26

Hello, To detect an answering machine I have found following two commands, BackgroundDetect (comes with asterisk) MachineDetect (asterisk add-ons) First question, does BackgroundDetect works well with g729? I havn't try MachineDetect yet, what is the benefit of MachineDetect over BackgroundDetect. If anybody used any of this command successfully, please help me. If possible, please let me

Hi All, I would like to ask for some advice how to solve following situation: We have to record and decode isdn PROGRESS message and when particular message is found call should be hang up and dialplan should continue. So far we have come up with two ways we think solve the problem and would very much appreciate to hear your opinion. 1) add special progress extension to dialplan so we can

On Tue, Apr 2, 2019 at 4:18 PM Joshua C. Colp <jcolp at digium.com> wrote: > On Tue, Apr 2, 2019, at 8:15 PM, BJ Weschke wrote: > > I get the desired use case to run app_amd from within a Stasis > > application, but I’m not sure about app_queue. You have everything at > > your disposal within ARI itself to replicate all of the functionality > > of app_queue and

Hey guys, I've got a part of my dialplan that dials multiple people: exten => direct,n,Dial(${QUEUEEXTS},${RINGTIME}) Multiple extensions are in the ${QUEUEEXTS} from an external script - e.g. SIP/100&SIP/101&SIP/105 etc This works great, however I want to see if I can find a way to work out (and run an AGI script) when the call is picked up by someone. Thanks all!

Hello, I have finished my basic polishing of the Asterisk GUI client I have been writing in Perl/TK and have released a first beta version on sourceforge: http://sourceforge.net/projects/astguiclient/ I am still working on a user manual for the application, but the code works and we have been using the same basic client for the last month here at my company and it is working just fine. I'm

I just got a Polycom soundpoint and I set it up using the phone and web based admin. I cant seem to figure out the config files and they are confusing me greatly and I dont have time for it :) Some things are odd, like on every reboot it seems the volume I set is reset? is there any way to fix that. And the ringer seems low. - Even all the way up Anyone willing to point out a good asterisk

I'm trying get this working. I've looked through the list, and can't see how to get AMD to print out more. I have it call and say Hello like I normally would. I've tried to say more and less doesn't seem to matter. After I hangup it does recognize hangup. Here's logging during an attempt where I make outbound call and answer, but then hangup after 1-2 seconds: Jan 24

I just received my first polycom 501, tried my best to follow all of the documentation and configs on the wiki ( though some conflicted in which case i tried both ways ), and at the end of the day can not connect my phone to asterisk. My Questsions: does the wiki information apply to the 501's? The images and config files at freedomphones, are those supposed to work with the 501? Has anyone

I have a couple of Polycom phones, bootrom 2.5.0, SIP 1.3.1.0056. Works great with Asterisk when I power on the phone. However, after some time, say an hour, I cannot receive calls on this phone. On Asterisk, when I do "database show", it does show the phone in there, but it cannot reach the phone. Is there a way to keep alive the connection with Polycom? Any suggestions? Thanks, --

Hi All, We are in search of the latest Polycom firmware SIP 1.6.5 and BootROM 3.1.3as per http://www.polycom.com/resource_center/1,,pw-492,00.html Can someone help? We have legitimately obtained these phones but even our official distributor can't get their hands on updated firmware. The only thing we have found is http://www.freedomphones.net/polycom/files/?M=A which has only old versions.

The Asterisk Development Team has announced the release of Asterisk 11.17.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 11.17.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

We have 4 systems supported by ASUS M2V mobos with AMD2 cpus on board. No we haven't gotten CentOS of any flavor to work yet. Yes, we did load Fedora Core 6 and it worked, with respect to the SATA driver only. There is a RAID, Audio and LAN driver for Linux that comes on the M2V mobo CD. The audio driver installs and works fine on FC6. I've read many of the messages with respect to

I am getting to my wits end with these phones (and so is my boss). I am getting an random echo on these phones and I have an issue opened with Polycom and its been in their research and development department for almost a month with no results. I have noticed that I get a message "RFC3389 support incomplete. Turn off on client if possible" in asterisk. I have researched this and made

I have one asterisk server where most of the calls that go through AMD get stuck in it, even if the analysis time of 3 seconds has already ended. It doesn't move to the next priority (which is checking the AMD_STATUS). Executing 'show channels' shows that the calls are stuck in the AMD app. I have another identical server where it doesn't happen. Both server have version 1.2.24

Hi, Here is the XML file of the my VM: <!-- WARNING: THIS IS AN AUTO-GENERATED FILE. CHANGES TO IT ARE LIKELY TO BE OVERWRITTEN AND LOST. Changes to this xml configuration should be made using: virsh edit SJ-SIMISPC or other application using the libvirt API. --> <domain type='kvm'> <name>SJ-SIMISPC</name>

I'm looking for an easy way to make asterisk perform as a basic (broadcast)autodialer. Basically all I want to do is give it a list of phone #'s and a pre-recorded message and have it call each one and play the message or leave it on the person's answering machine. The people I'll be calling are all our customers, etc. so I don't need to do any do-not-call checking. Just