similar to: Combine agent inbound and outbound

I am trying to record a call into a stereo mp3 in Asterisk 1.4, but I can't seem to get it to work correct. Could someone point me to what I need to do? I have attached what I believe are the relevant parts. [globals] ; script to be executed when monitoring has been finished MONITOR_EXEC=/usr/local/bin/2wav2mp3 ; uncomment this line if you are using Ogg Vorbis

Hi. I'm having a terrible time with call quality coming into my * box. I'm using VoicePulse over a 1.5/1.5 mbit line. Outbound calls are crystal clear on both the RX/TX sides of the conversation. Inbound calls, though, are HORRIBLY garbled on the RX side. I can barely hear the caller, but they report my quality is fine. Getting loads of garbled sounds and weird echoes. (Could just be

This is what I'm doing it gets you both sides of the phone call...small size...and playable on windows through a share. My notes: On redhat 9 I have to run the following command for asterisk to start LD_ASSUME_KERNEL=2.4.1 asterisk -vvvvgc [macro-record-on] exten => s,1,SetVar(CALLFILENAME=${TIMESTAMP}-${ARG2}-${ARG1}) exten => s,2,Monitor(wav,${CALLFILENAME}) ;exten =>

Hi, Can someone please comment about the below issue [root at host0040 kaushal]# file obd-demo.mp3 obd-demo.mp3: MPEG ADTS, layer III, v1, 256 kBits, 44.1 kHz, Monaural [root at host0040 kaushal]# sox obd-demo.mp3 -e stat sox: Failed reading obd-demo.mp3: Do not understand format type: mp3 [root at host0040 kaushal]# sox -V obd-demo.mp3 -r 8000 -c 1 -t ul -w vm-intro.ulaw sox: Failed reading

On Tue, 9 Feb 2016, Larry Turnbull wrote: > One of the streams is an old time radio stream and I use ezstream to run the > prerecorded shows. > > Is there a package or some sort of way I can apply dynamic compression to > the stream? Ezstream really isn't designed for this. It's primarily meant for streaming files as they are, which is a very light-weight operation, with

Thanks Geoff, I will give it a try. I am reencoding all files to a 128K mp3, 44100 sample rate stereo stream for consistency. So yes I have the decode and encode lines in the xml file as you outlined. I will check out sox and see how it does. I am also exploring mp3gain to run across the entire library and see if getting all files to the same level will help. Larry -----Original

Last night Mark and I made some changes to the IAXTEL tollfree outbound, and inbound access. The inbound access number has changed to: 248-724-0700. (This number is in Pontiac, MI Ratecenter, and is supplied by Telesthetic LLC, a next gen phone compnay) This number will say "Please dial your number now" at that point you can dial your 1-700-XXX-XXXX IAXTEL number assigned. In the

silly me, I thought the inbound would be the hard part. how little I knew... can someone please give me any insight into why outbound is not working, in fact why trying to enable outbound fouls up everything? I'm using asterisk, most recent from cvs, I'm behind a nat, and I'm trying to use iconnecthere.com for outbound and inbound. Inbound is working fine, no problems. But for

Hello All, I would like to record inbound and outbound calls to and from one number. I tried to add lines to my extensions.conf: DAY=`date "+%m-%d-%y_%H:%m"` ;outbound exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(${TRUNKL3}/${EXTEN}) ;Inbound [line2] exten => 5555551212,1,Record(${DAY}:gsm) exten => 5555551212,2,Dial(SIP/101,20) exten =>

Hi to all, I need some help, I have an Asterisk Server in a small call center, for inbound calls I setup a Queue in queues.conf and their respective Agents in agents.conf, but when an Agent is calling out and a call is coming from PSTN the call is send to that agents which have a call in progress. How I can fix this in order to have only one call at a time. I think in limitonpeer and call-limit

Hi all, I have a budget to work with and was wondering if there are any folks providing SIP/IAX2 trunking for unlimited inbound/outbound for a flat rate? We're in the budget range of roughly $5,000 a month and we need multiple channels per DID. I'm not sure if something like this is feasible in the world of VoIP -- and I only need to be able to make domestic/USA calls. Thanks for any

I'm trying to set up a 2nd DC.  The primary is a Server 2003 (ads1).  But DNS is being done on a Linux proxy server.  It appears that I have joined correctly (wbinfo -t: checking the trust secret for domain SAMDOM via RPC calls succeeded).  But when I check inbound and outbound (samba-tool drs showrepl) I get the following: > Default-First-Site-Name\ATHENA > DSA Options: 0x00000001

Greetings list- I have a box with a single FXO card in it. I'm able to dial out ZAP/1 with no problems and as expected. However, I would like inbound calls on that POTS line to go unanswered by Asterisk since I have other equipment on the line. I've setup zapata.conf for the channel without a context but the line is still answered. I've also setup a blank context with the same result.

Hello! I've run into a problem where a user is making an outbound call at the same time that an inbound call is being made on the same analog line. It appears that as the zap channel is opened for the outbound call, it is simply answering the inbound call. Obviously, both parties involved in the calling get a bit confused. Previously, it happened only on an occasional basis. However, as this

I need to monitor the states of my T1/PRI Zap channels. Specifically, I need to be able to programmatically determine whether a channel is unused, carrying an inbound call, or carrying an outbound call. Using the manager interface, I can easily tell whether a Zap channel is used or not by looking at the results of: Action: Command Command: zap show channel <x> Or: Action:

Hi, I am using the following script to limit my outbound traffic. This scipt runs on a box behind my firewall. It limits my outbound passive ftp traffic to 39K perfectly....just like i want. However, i just noticed that it is also limiting uploads coming to my server. Is there something I can change to make it not limit uploads to my server? #!/bin/bash #shaping passive ftp traffic #

Dear all, I think this is common, or at least how it is supposed to be, but whening dialing over a ZAP channel, it's taking around 5~ seconds to ring on the over end, likewise inbound. This is just with a normal Dial command. Are there any ways to tweak this? Thanks, Gavin.

I have a strange problem in that when I receive a call through the X100P which is forwarded to my budgetone 100 then the voice quality is perfect both directions. However, if I make a call out from the budgetone to the same caller via the X100P the sound level is a lot lower and the quality a lot poorer. I've had to set the rx tx gain to 1.5 or I can hardly hear at all. Any ideas what is

Hello I get this warning all the time when I am using iax2 for inbound calls or outbound. Aug 24 13:48:41 WARNING[-1105474640]: chan_iax2.c:4873 socket_read: Error: Resource temporarily unavailable I get the calls and the sound is fine. But the screen on the cli is full of these warnings and Error: What can I do to fix this. I get it when using calls to iaxtel, FWD, VoicePulse, Nufone and

I can't find this in the wiki/email history..but I'm sure it's based asked before. The port range define in rtp.conf - is that for connections initiated by asterisk? Or the port range asterisk listens on? Or both? Thanks! MD -------------- next part -------------- An HTML attachment was scrubbed... URL: