similar to: Scalable IVR with asterisk

Hello all: Thank you for taking the time to read this post. Background: I am a new user to IVR systems and asterisk. I have been tasked with helping to set up a system that will only handle IVR (eg no PBX functions) incomming calls for 45 or so people that will call in 3 or 4 time each day during (approx) normal business hours. We have started to look at the Ivrs perl module from

Hi, I am an asterisk newbie and looking around for information . I wish someone could take their valuable time off to answer my query in detail. I wish to set up an IVR system that can allow user authentication and therefter accept 2-3 inputs from users ..generate a key and transmit the same in voice back to the user . The system will intially have small load but if the whole package in future

If you are looking for real high density VOIP termination I would look at > something like a Lucent APX 8000, configure correctly it can pass 2500+ > g.729 calls to the PSTN course we paid lots of $ for ours. > > Chris > Chris, My experience has been that the APX and TNT products require a single SIP proxy, how are you load balancing 2500 calls? If all of the traffic is

Hi i am looking for a good ivr system for my company. these are my question are there any good ivr's that can be easily integrated with asterisk ? and are there any large scale deployment of asterisk to date ? thanks Giridhar Bandi -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I'm currently testing Asterisk over a T1 cross connect to a MaxTNT chassis that we have. It is working fine switching the calls through, but there is about a 10 second delay from the time Asterisk initiates the call until the TNT accepts it. It appears to be a ANI issue, I've changed several settings and formatting options on the T1 between the two, as well as turning on/off the

Hello folks, I've been trying to get T.38 over SIP working with calls terminated by a MAX/Lucent/Ascent TNT. As far as I can tell, SIP and T.38 are actually working perfectly; however, I can't get the TNT to properly terminate a FAX call. Does anyone have a working configuration for SIP and T.38 for calls from a TNT or APX? Here's a brief description/diagram of my test setup:

Hi All, I have a lab setup with two asterisk servers and a MAX TNT in the middle like this: asterisk sip >< sip TNT pri >< pri asterisk The TNT is running 11.0.6 and the asterisk servers are running 1.2.9.1. I can get calls to pass from asterisk sip to tnt to pri to asterisk but not the other way. The call from asterisk to pri to tnt is good, the TNT is passing SIP invite to the

I'm trying to get inbound calls from a TNT working but get 407 errors from the TNT. This is what I have in sip.conf: [maxtnt] type=friend host=x.x.x.x dtmfmode=rfc2833 callerid="MaxTNT" <maxtnt> context=demo qualify=yes disallow=all allow=g729 allow=ulaw insecure=very This is what the TNT is spitting out: Jul 24 14:55:12 tnt1 1/17: Releasing

I'm looking for someone familiar with setting up some of the more advanced features of the Lucent TNT, preferably someone with knowledge of Trunk Groups and choosing outgoing PRI channels based on call type and perhaps NPA-NXX We currently have 8 PRI's. 7 of them are for our dialup pool, the 8th is for our voip. We currently run the dialup PRI's to a seperate TNT We want to

Hello, I have a Lucent MAX TNT, (DS-3, 672 modem ports, 28 PRIs). I'd like to be able to direct an inbound fax call into my TNT, have it answer the fax and send the image file over to Asterisk, or some other system to deliver to an e-mail address(s). I'm not sure if I need Asterisk to any of the call control or not. I'd also like to setup a print queue and have outbound

Is anyone using a media gateway with a DS3 for TDM/PSTN access? I am trying to determine the best way to scale Asterisk beyond 4 PRI's. What about the TNT Max? Can it be configured with a DS3 PSTN interface, the appropriate software and hardware, and Ethernet ports to support SIP-PSTN/DS3 media gateway services for multiple * boxes? What about TNT - SER - * ? Is there anyone here with real

I'm thinking about using asterisk as an IVR system with an existing avaya index system. I've got 2x PRI 30 lines coming in to the Index, and I have 4 spare PRI cards in the Index. I was thinking about using a QUAD PRI card from Digium and having the calls come into the Index then transfer to Asterisk for IVR then back to the Index. That way if we get 60 inbound calls we'd in

Hi Jonathan, On Sun, 23 Aug 2020, Jonathan Billings wrote: > On Aug 23, 2020, at 12:07, me at tdiehl.org wrote: >> >> ?Hi, >> >> I am trying to create a mock configuration file that points to my local dnf repo. >> I have the following in the mock config template: >> >> [local-repo] >> name=my-Local >>

nouveau supports the Tegra K1 and higher after the SoC-based GPUs converged with the main GeForce GPU families. Signed-off-by: Rhys Kidd <rhyskidd at gmail.com> --- drivers/gpu/drm/nouveau/nouveau_drv.h | 2 +- 1 file changed, 1 insertion(+), 1 deletion(-) diff --git a/drivers/gpu/drm/nouveau/nouveau_drv.h b/drivers/gpu/drm/nouveau/nouveau_drv.h index 822fe1d4d35e..fbe42ec0a5f1 100644 ---

I am working on directed network data of the following form: Origin farm Destination farm Year Farm 1 -> Farm 2 1999 Farm 1 -> Farm 3 1999 Farm 3 -> Farm 2 2000 Farm 1 -> Farm 2 2000 I am using the package 'igraph' and I would like to generate and plot (as

Found this. http://www.tnt.uni-hannover.de/~baumgart/peri_ear_v0.1.tgz The peripheral ear model is based on the structure of Eberhard Zwicker's "Analogmodell". That model consists of analog electrical elements. ftp://ftp.tnt.uni-hannover.de/pub/papers/2000/Diss-FB.pdf Input: Raw 100KHz monoural 16bit audio Output: hair cell excitation signal at 251 equidistant locations in

Im having some problems with connecting a TNT to asterisk. The problem is when the call is sent to asterisk and signaling is done the RTP syncs however no audio is produced. Can someone give me some idea of how to accomplish this? I am using the standard configs and g711 and 729 do the same. No audio. Public IPs on both ends. No nat. Any ideas would be appreciated.

Hi, Thinking about an IVR application and trying to get a handle on the best way to structure it so that the maximum number of concurrent calls can be achieved.. If the voice prompts were stored in a GSM format and were being played out through an IAX trunk that uses GSM compression would asterisk do a decompress/compress on the audio or would it simply pass through the GSM encoding?

Hi All, We are using MAX TNT to for some T1 PRI interconnects. I'm seeing the voice channels connect at 56K. Does anyone have the DS0 channels connecting at 64K for voice, if so what is the parameter to select 56k or 64k channels? I'm not having any issues that I know of, just wanted to bounce this off the group for a sanity check. Thanks. JR -- JR Richardson Engineering for the

I'm running SIP between my Lucent TNT acting as a gateway, and an asterisk server. We have a PRI coming into the Lucent. Basically the problem I'm having is mostly on inbound calls but some outbound calls as well. I hear echo and sometimes some weird artifacting on calls coming in from the lucent. Everything routed over IAX to VoIP Jet or Nufone sounds fine. It seems like every 3