similar to: calls stuck in AMD even after analysis time

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

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

I saw a couple of posts about this in the archive, but none seemed specifically to address the problem I am having. If I missed something please let me know. Right now I would classify myself as "novice," and there is probably really nothing so trivial that I couldn't possibly have screwed it up. :-) I'm trying to use the AMD command to detect answering machines, and have

I'm guessing this may be a question for dev list, but wanted to try my luck here first. I'm trying to compile app_amd (Answering Machine Detection) against 1.2.7.1 and am getting some errors. I should point out that I simply snarfed app_amd.c from http://svn.digium.com/view/asterisk/trunk/apps/app_amd.c?rev=14714 ...so if there are other includes and such that are required, that would

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

Trying to install wanpipe 3.5.18. No errors during compile. But when I reach the point where wanpipe and dahdi_cfg is started, I encountered an error. Starting WAN Router... Loading WAN drivers: wanpipe done. Starting up device: wanpipe1 wanconfig: WAN device wanpipe1 driver load failed !! : ioctl(wanpipe1,ROUTER_SETUP) failed: : 22 - Invalid

Hi.. We are facing a problem that is making the channel to be stuck. we are using asterisk 1.4.22.1 version, and we have a direct sip trunk...We have 2 queues and one has 2 agents and the other 5 agents, from last week the second queue's channel is getting stuck, it happened 3 times till now and the problem is calls come into the queue and just the calls will be in the queue and will not ring

Anyone here who has configured zaptel/dahdi for a singtel E1 line? What are the settings for coding, framing, line type and switchtype? -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110210/4288ed84/attachment.htm>

I am currently testing a 57i unit. No problems configuring the phone's config via phone/web UI. We are trying to avoid using the web UI, the reason is it will take a long time typing the softkey xml applications URIs on each phone, so we chose TFTP. Tried configuring the phone via a TFTP config server, but no changes took effect. I wonder why it doesn't work with TFTP even if I was able

Hi, I have some general questions about statistical analysis for a research dataset and a request for advice on using R and associated packages for a valid analysis of this data. I can only pose the problem as how to run multiple ANOVA tests on time series data, with reasonable controls of the family-wise error rate. If we run analysis at many small sections of a long time-series, the Type-I

This is the first time I have used a sip device on a queue. Iogged in under the extension and now I can't logout. No kidding. I have restarted asterisk with persistant agents=off and also done a show agents. It shows no agents logged on and I am still receiving calls. To complicate things I am using Flash operator panel to see logged in agents and it has, at best, been sporadic. I have had no

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

Also posting this question to people working on manager interface and dialers. I have a simple auto dialing script (using Originate) that forwards all incoming calls to a queue full of waiting agents instead of a meetme conference room. I use queues rather than meetme so I can leave the automatic call distribution to the queue itself. The problem is when the calls reach the agents, some of the

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

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

The Asterisk Development Team would like to announce the release of Certified Asterisk 16.8-cert1. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 16.8-cert1 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following

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

The Asterisk Development Team would like to announce the release of Asterisk 13.28.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.28.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

The Asterisk Development Team would like to announce the release of Asterisk 16.5.0. This release is available for immediate download at https://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 16.5.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following issues are resolved in this release:

Ronald, I've been experimenting with something similar. You might want to check this out: http://www.voip-info.org/wiki-Asterisk+auto-dial+out+deliver+message What kind of trunks do you have for your outbound calls? (BRI/PRI/analog POTS/SIP/IAX etc.) I'm using PRI and it works very well - the dialplan doesn't execute the message playback until after the call has been answered. I