similar to: One way choppy sound

Hi Guys I am new working with lumenvox products, and unfortunately I had not been able to install it properly, I follow the steps in lumenvox site and it looks like it works I mean: ========================================================= [root at pbx-millenium examples]# ./example 127.0.0.1 Connecting to 127.0.0.1 Interpretation 1: 8587070707 count=0, decode returns 1 Interpretation 1:

Hi I have the last firmware for zplex, if you like i send it to you, about the second question 24s means 24 extensions so you can configurate as you wish as fxo or fxs. Att Yelson Vivas

Hi everybody i have a problem trying to connect an incomming phone call from pstn to my (soft phone) iaxcomm, the phone rings but when i try to answer the call, asterisk sends a message like this. Jun 1 19:33:17 NOTICE[5013528]: channel.c:1223 ast_read: Dropping incompatible voice frame on IAX2[192.168.222.99:4569]/16 of format GSM since our native format has changed to ALAW i'm working

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HI Asterisk Gurus I'm connecting my * server to a zplex 10B, I'm using a slot on my TE410P card as T1 to connect my server to the channel bank (i use a cross cable), the zplex doesn't show any alarm neither the server. I'm trying to make calls from (ext 1) from the channel bank, to other (ext 2) in the same channel bank, but when I dial the ext number, the dialtone doesn't

I'm trying to transition from AMI to ARI. Running into a small hiccup when I try to create (originate a call) with the caller id name and number I can pass the Name and Number if the name has no spaces in it and it shows up in my PhonerLite application. curl -v -u asterisk:asterisk -X POST http://asterisk:astersk at localhost:8088/ari/channels/mycallerid.1?endpoint=PJSIP/1003 at

Hi Dan, as far as PPI and PAI Header, we use the channel Vars in order to do that. In Latest Asterisk you can set Channel vars within the create command in the Body. Just set the PJSIP(add,PAI) as you would do it in Dialplan. https://blogs.asterisk.org/2020/07/22/ari-create-channel-with-variables/ BR Jöran On Fri, Aug 7, 2020 at 7:06 PM Dan Cropp <dan at amtelco.com> wrote: > An

Dear R Developers, There is an interesting issue related to "reticulate" R package which discusses how to convert Python's non-32 bit integers to R, which has had quite an exhaustive discussion: https://github.com/rstudio/reticulate/issues/323 Python seems to handle integers differently from R, and is dependant on the system arquitecture: On 32 bit systems uses 32-bit integers,

Hello team; While am aware and active user of astersk monitor function for recording, i would like to know if i can use asterisk as a pure recording server(like nice or witness) for some other PABX's extensions (both inbound, outbound and internal). Setup PSTN---Legacy PABX(with analogy n digital extensions)--- asterisk(record Legacy PABX extensions.) Sam -------------- next part

Hello everybody! I have some problems with my Astersk. I have an analogical OpenVox card and A Billion ISDN card (with mISDN). I load the modules with modprobe zaptel and modprobe wctdm. When I run ztcfg -vv I have this: Zaptel Configuration ====================== Channel map: Channel 01: FXS Kewlstart (Default) (Slaves: 01) 1 channels configured. ZT_CHANCONFIG failed on channel 1: No

Hi Dan, i would do something like this (it is not a copy of what we are doing but an example of how i would do it) Important here is the "--data" and "-H" Option as well as the "variables" section within the Body. I added the callerid function here as well as it is the sample in the asterisk wiki. curl -v -H "Content-Type: application/json" -u

Hi, I have an arquitecture, of squid-Active Directory with NTLM authentication, I have users registered in Active Directory with character '?' in this username, with ldap authentication work well, but with ntlm the user is not found, and can't be authenticated. Is a problem of charset?, my linux use utf-8. Maybe change it to latin-1 could resolve? with the variables unix charset,

Hi all: I have been using ggplot2 graphics for quite some time now and I really like it. However, I haven't used barplots enough to understand the arquitecture behind it. Can someone show me how to make this simple barplot graph with ggplot2? I want "PondName" along the x axis and "avgWt" along the Y axis which represents the avgWt by each Pond. PondName avgWt Pond01 21.5

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, all. Normally, i use 'domain.ltd/username/Maildir' as users' maildir path, if i change them to hash style, e.g. 'A0/B0/domain.ltd/C0/D0/username/Maildir', will it speed up the index operation for MDA? If we have 10000 users, which maildir path style will improve performance? Thanks very much. :) -- Best regards. Zhang Huangbin - Open Source Mail Server Solution for

Thank you Gabriel for valuable insights on the 64-bit integers topic. In addition, my statement was wrong, as Python3 seems to have unlimited (and variable) size integers. Here is related CPython Code: https://github.com/python/cpython/blob/master/Objects/longobject.c Division between Int-32 and Int-64 seems to only happen in Python2. Best, Juan El mi?rcoles, 29 de mayo de 2019, Gabriel

Hi My asterisk output is: chan_sip.so => (Session Initiation Protocol (SIP)) Asterisk Ready. -- Registered SIP '201' at 192.168.0.55 port 33906 -- Saved useragent "X-Lite release 1011s stamp 41150" for peer 201 -- Executing [907768385144 at default:1] Dial("SIP/201-083e75c0", "ZAP/g1/907768385144|60") in new stack [Oct 4 11:54:27]

Hi all, I use astersk 11.7.0 on Ubuntu 12.04.01 TLS (i386). I use cdr_adaptive_odbc to write CDR to my MySQL's cdr table. After my testing, this scenario is working well. After a long idle time, I didn't make any call to the asterisk server. When I try to make a call again after 8 hours, I found that the cdr lost. It cannot be inserted to cdr table. Also, I could not find the insert CDR

Hi, sorry if this is a newbie question, but in fact I am sort of a newbie. Is there a way of connecting two answered and active voice channels together in an AGI script for some time, having the two parties talk to each other, at the same time have asterisk or the AGI script listen for DTMF tones on both channels and react to certain tones, i.e. disconnecting the two channels on reception of

I got no resonses to this and some funny bounces so I'm trying again. First of all Merry Christmas. Second, my first problem with my provider not staying registered with our server was my fault. We moved our server room and I restarted the test system and the production system causing them to ping-pong back and forth registering with our provider causing random problems, they are both