similar to: Help bridging 2 outbound IAX2 calls !

Displaying 20 results from an estimated 8000 matches similar to: "Help bridging 2 outbound IAX2 calls !"

2004 Dec 15
1
MAC address question
Dear list, I have a network of 600 users and im using shorewall as the firewalling system for our Linux gateway. We would like to allow all the users to online by entering the MAC addresses of them. Is it possible for shorewall to handle 600 entries of mac address and how about the performance issues? Please advice. __________________________________ Do you Yahoo!? Jazz up your holiday
2004 Dec 29
3
icecast2.2 and aac?
it seems that icecast 2.2 can only stream aac at 128kbps! mp3 and ogg are ok at all bitrates. is it a bug or my config wrong? --------------------------------- Do you Yahoo!? Jazz up your holiday email with celebrity designs. Learn more. -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Dec 10
1
Decoder performance
Hi all, I'm thinking of using Speex for an embedded project. I would only need the decode part. My question is what percentage of the CPU is used on an optimized (assembly will be done) SH4 or ARM7 or ARM9 speex decoder running at 100Mhz. Thanks, Bolt __________________________________ Do you Yahoo!? Jazz up your holiday email with celebrity designs. Learn more.
2004 Dec 29
0
icecast2.2 and aac?
Dennis- Orban Opticodec-PC LE is available now, as of last Tuesday, the release date of Icecast2 v2.2. StreamGuys is filling all orders now. Thanks. -greg. At 12:53 2004-12-29, Dennis Heerema wrote: >Trying to get the LE version for months now, still treamguys can?t deliver.... > >Regards, > >Dennis Heerema > >-----Original Message----- >From: "Greg J.
2005 Jan 21
0
IAX2 trunking, Voicepulse Connect, and Outbound Faxing
I've just stumbled across a rather weird problem and was wondering if someone could shed some light on the situation. In testing faxing through Asterisk using Voicepulse Connect for trunking I am able to receive faxes without a hitch. Quite impressive considering previous experience with certain other VOIP providers. Today I finally got around to testing outbound faxing and found that if
2004 Dec 29
2
icecast2.2 and aac?
Trying to get the LE version for months now, still treamguys can?t deliver.... Regards, Dennis Heerema -----Original Message----- From: "Greg J. Ogonowski" <greg@orban.com> To: qiang Bao <jakobao@yahoo.com>, icecast@xiph.org Date: Wed, 29 Dec 2004 11:08:04 -0800 Subject: Re: [Icecast] icecast2.2 and aac? Icecast 2.2 works fine with AAC/aacPlus at any bitrate. 32kbps
2004 May 24
0
IAX problems using CVS HEAD, but not CVS STABLE
Hi All, Sorry if this has been covered in the past; I've tried searching the archives, but haven't had any luck finding a similar problem. Basically I have problems when using IAX2 (which I now understand is just IAX). I have three IAX connections setup - VoicePulse, IAXtel, and an Asterisk IAX<->PSTN termination provider here in Sydney (ATP) If I try to use the CVS STABLE version
2004 Dec 15
0
Re: [S] using Hmisc and Design library
sorry, I had a typo there, it's datadist(b) for the analysis of data frame "b". --- Robert Balshaw <Robert.Balshaw at syreon.com> wrote: > Not sure if this will help, but did you mean to use > datadist(a) for > the analysis of B? > > Rob > > > -----Original Message----- > > From: r-help-bounces at stat.math.ethz.ch > >
2004 Aug 24
0
Warning when I use iax2 for inbound and outbound calls
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
2004 Apr 08
0
IAX2 Trunk to PSTN (voicepulse) questions...
All, I've almost got my Asterisk PBX setup, but I've having some problems with the VoicePulse IAX trunk. On outbound calls, when dialing a PSTN number through the IAX2 trunk, music on hold (moh, using the m option in the dial command) does not work. The console states that "stop sound" on IAX2 channel. Ring works, but only without the r option. MOH works when trying to dial a
2004 Aug 21
1
IAX2 DTMF not recognized - Bug report - Help sought
I have working SIP numbers with broadvoice, and just added a DID from http://connect.voicepulse.com/ . The calls answer, but DTMF is not recognized. With "iax2 debug" active pressing DTMF does nothing. Zilch. Zero. A friend tried a different IAX2 connection, and got the same results. I see the following in the archives: On Fri, 2004-04-09 at 10:12, Robert Jackson wrote: > Hey
2004 Aug 30
2
VoicePulse Connect DTMF with IAX2
Is there anyone out there who has VoicePulse Connect working with DTMF? I've been unable to get it to work from the start, and the recent VoicePulse updates did not help. A caller to my DID's hears Asterisk, but pressing DTMF does nothing: On call setup "iax2 debug" shows: ----------------- Tx-Frame Retry[-01] -- OSeqno: 001 ISeqno: 002 Type: IAX Subclass: ACK
2004 Dec 09
11
Asterisk@Home
I have started to receive a lot of positive response for the Asterisk@Home project. For those of you unfamiliar with this project the goal of Asterisk@Home is to make a full featured version of Asterisk very easy to install. We have created a 1 step .iso that installs RHEL (RedHat Enterprise Linux) and Asterisk. It includes a web GUI that allows easy editing of the Asterisk Config files.
2004 Jan 29
1
re: help with voicepulse connect IAX2
hello, after playing with an asterisk configuration for voip for a few weeks i'm trying to get outbound dialing with voicepulse going - i've cut down the asterisk to a very minimal install (1 SIP client) to try to localize the problem. The SIP client works fine (SIP and * on the same NAT) and could access the demo from samples before i removed it, and can call itself - so i am
2003 Oct 16
1
Weird IAX2 problem
I have an inbound and outbound account with Voicepulse (I am very happy with the service, btw). But I have a weird IAX2 problem. When I get a inbound call on my Voicepulse DID, the call hits my asterisk server correctly with the correct callerid (the DID phone number 617902xxxx). when the call gets passed on to a softphone (X-lite), the caller id that shows up on the X-lite softphone as Lee ,
2004 Feb 02
0
VoicePulse IAX2 lag
Yes, and they are aware of the problem. -----Original Message----- From: asterisk-users-admin@lists.digium.com [mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Jonathan Tew Sent: Monday, February 02, 2004 1:59 PM To: asterisk-users@lists.digium.com Subject: [Asterisk-Users] VoicePulse IAX2 lag Is anyone else noticing high lag on their voicepulse IAX2 connections? We're seeing
2003 Sep 18
2
VoicePulse offering IAX2 services
I don't know if this has been mentioned yet: Voicepulse is now offering wholesale pricing and IAX2 connectivity for Asterisk users. No fees, pay as you go. They also offer incoming calls for $7.99 per month. See wholesale.voicepulse.com.
2003 Dec 09
1
Outbound iax dialing to one #
What I am trying to do is in the 3rd option dial my cell# thru voicepulse I just can't figure how to construct the line [inevans] exten => s,1,setcallerid(${CALLERID}) exten => s,2,Dial(MGCP/aaln/1@Egraph-1,10,tr) exten => s,3,Dial(iax2/passwod@voicepulse.com/ Where do I put the # to dial 18708573287 thanks James Schenck Egraph Design Inc. Arkansas Online Internet Services (870)
2004 Jul 11
1
Stopping reinvite with IAX2?
Hi All, I'm using DISA on my * server to avoid overseas toll charges when making calls to Western Europe from my cell phone. I have DISA working with a DID from a VoicePulse Connect account. The outgoing call to Europe is also made via Voicepulse Connect. I see that the IAX media path is bridging the inbound call to the outbound call so that the media stream entirely bypasses my server once
2004 Apr 09
2
IAX2 DTMF Problem
Hey all, I am dialing a DID through VoicePulse Connect. The number is answered by a main menu type of IVR. The configuration is as specified in both the wiki and VoicePulses documentation. The call comes through without a problem, but when the caller enter any keys they are either not recieved by * or they are ignored. With SIP I would typically put a dtmfmode= line under the peer and