similar to: silence in virtual extension

HI All asterisk user I Have one Asterisk with this scenario: i have two ip Address one Private IP one Public IP, my internals terminals using private IP works very fine but my terminals using public ip don't work audio , make rings but streamer don't work. thks for you attetion best Regards -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello! I was using Speex all the time, and I am now moving to Opus. I had encapsulated the decoder a bit, I had the following cpp file: #include "StdAfx.h" #include "spxcodec.h" #define MAX_FRAME_SIZE 2000 #define MAX_FRAME_BYTES 2000 CSpxCodec::CSpxCodec() : enh(1), rate(8000) { } CSpxCodec::~CSpxCodec() { } void CSpxCodec::Init() { speex_bits_init(&bits);

Hi, I get fascinated when I dial someone and get an IVR play " for accounts department press 1, for sales, press 2 or hold the line for an operator" and then have MOH play before the line is picked up at the desired extesion. Please, permit me as I know this will be one of the dumbest questions to ask in a group like this. I'll apprecaite any specific guide/instruction. Thanks

> Is it possible to tell asterisk not to strip the leading 0 > of *incoming* MSNs? I use asterisk with i4l and whenever > I get a call from an long-distance party, the leading 0, which > should be there according the german numbering, is not. Are you *really* sure that the 0 is transmitted in the CLI, and that it isn't stripped already by the phone company? I think the easiest

hi, all I have configure tdm400p analog fxo card. that's ok. but how to chek that is working properly or not. i chek with ztcfg -vvvv and zttool . that's ok. i want to dial from my fxo port to another extesion. zaptel.conf ------------------ fxsls=1,2,3,4 defaultzone=in loadzone=in zapata.conf ---------------- context=mycontext signalling=fxl_ls group=1 channel=1-4 thanks' in

I have a very simple question . I have 2 internal extension 301 and 300 sip phone . I want to these extesion can call each other, and ext 300 can call outside to pstn, and ext 301 to call internatonal. How can I do that ? [x1] exten => 300,1,Dial(SIP/300) include => pstnlocal [x2] exten => 301,1,Dial(SIP/301) include =>international [pstnlocal] exten =>

Hello all! First of all, this is my first post to the list. I've tried to find my answers in the forums and by Googling , but no luck. My apologies if this question has been answered before. I've set up an Asterisk box with four local SIP users. The Asterisk box uses a SIP provider for placing external calls and receiving incoming calls as well. In other words, there's no PSTN

Hello!! Hehehe Then, as already changed the values and problem had my idei and leave everything as it was, the two idmap config *: range = 5000-16777216 idmap config SERVERAD: range = 5000-33554431 It is running more than one year and occurred only problems that I changed, I know the right and leave the range as you passed, but I can not have the ID change issues again (caused much

When i made internal call into my LAN using x-lite sip phone client I retrive in askterisk CLI : ----------- ERROR ---------- Verbosity is at least 6 -- Remote UNIX connection -- Executing Dial("SIP/201-979d", "SIP/201|60|t") in new stack -- Called 201 May 29 18:09:28 WARNING[6082]: chan_sip.c:694 retrans_pkt: Maximum retries exceeded on call

I searched the 22490 messages I have in my own personal asterisk-users archive and have not found the answer, and it also does not appear on the wiki. I have a SIP phone and a regular phone on a TDM400P FXS interface. Extensions are 100 and 101, respectively. On the SIP phone I can hit #, get the "Transfer" prompt and enter an extension I want to transfer to. No problem. I can do

Hi, asterisk gurus: My purpose is to key in some number after the call is finished. The number keyed in will be stored in the database with the phone number dialed. But whenever a key is pressed in/after h extesion, asterisk exits the call flow. Does anybody has a solution for this? Is DEADAGI is possible solution? Really need help! Thanks!Thanks!Thanks! Manny

I've been using AMP to manage my * test system. I've been trying to activate an extension that I don't want AMP to manage. It would appear that the extesion definitions are placed in the appropriate "custom" files which are then added with an include command to the appropriate master file (sip.conf,extension.conf, etc). So far I've not been able to get it to work. Anyone

Hi i have installed AAH 2.6 and configured some extensions the calls are working fine. but if i dont answer a call then it says " the person at extension " and hangs up . it doesnt spell out the extesion number nor it goes to voice mail box. *************************** Asterisk CLI log **************************** dialparties.agi: Extension 200 is available...skipping checks --

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Hello list, I would like to seek your expert opinion on a setup I am trying as part of my research. I have not been able to successfully make a call so far. In my setup, I use two laptops that are interconnected by means of a stand-alone IS1581 switch. Thus there is no LAN involved. I have assigned static IPs to the two laptops, say 10.0.0.1 and 10.0.0.2. I have installed Asterisk 1.6.2.6 and

Hello, I have problem with interpretation of CDR entries. What happened? ------------- There was: 1. at 09:00:26 we received call from unknown caller (no callerid) to secretary with extension 17 2. secretary answered and after some conversation called to extension 18 to check if she could transfer customer call 3. at 09:02:55 extension 18 answered and call was forwarded 4. call was very long

Hello! Any opinion on that? Thank you Em 13-07-2016 10:52, Carlos A. P. Cunha escreveu: > > Thank you for the explanation. > Yes, it was a mistake to leave my two faxias that way, by the ID > exchange reason the low range will leave as it was to have no problems > idmap config SERVERAD: range = 5000-33554431 > > The range of up'm thinking of changing to something >

I tried to get icecast2 and BLAMO Error: Failed to spawn GNU rlog on '/cvsroot/icecast-1.1//Makefile.in,v' '/cvsroot/icecast-1.1//acconfig.h,v' '/cvsroot/icecast-1.1//commandline.c,v' '/cvsroot/icecast-1.1//commandline.h,v' '/cvsroot/icecast-1.1//commands.c,v' '/cvsroot/icecast-1.1//commands.h,v' '/cvsroot/icecast-1.1//config.h.in,v'

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Thank you for the explanation. Yes, it was a mistake to leave my two faxias that way, by the ID exchange reason the low range will leave as it was to have no problems idmap config SERVERAD: range = 5000-33554431 The range of up'm thinking of changing to something idmap config *: range = 2000-4500 Not to be superimposed. But it will it not cause problem ids trading again? Since it was