similar to: error to load chan_iax.so

Hello all, I'am trying to use 2 gnophones on my LAN. But I can't get any sound. Here is my configuration: 1. on PC1, I have: - Asterisk compiled from CVS (CVS-05/15/03) and it runs. - Gnophone binary version from Debian: 0.2.4+cvs.20020624-3 - a sound card working ( module es1371 for /dev/dsp) - I registered it as "alice" in extensions.conf 2. on PC2, I

Hi all, I try to ask helps again about chan_h323 extensions. I define this in h323.conf: [general] port = 1720 bindaddr = 0.0.0.0 allow=gsm allow=ulaw gatekeeper = DISABLE context=default [gm1] type=friend host=192.168.1.20 context=default [gm2] type=friend host=192.168.1.25 context=default and I have in extensions.conf : [demo]

Hello all, Encouraged by the successful "demo", I'am getting on with Asterisk CVS. I added 2 H.323 extensions in extensions.conf [default] include => demo exten => 701,1,Dial(H323/gm2@192.168.1.20/s) exten => 702,1,Dial(H323/gm2@192.168.1.25/s) With: - [demo] is defined by default in sample.extensions.conf - Asterisk server is running on host 192.168.1.20, on the

Hello all and guest@misery.digium.com , I was surprised to play the "demo" extension from my Asterisk CVS. It was around "Fri May 23 22:18:37 CEST 2003". While trying to make a test with chan_h323, I got a "wrong" number and fell down on someone at this address IAX2/guest@misery.digium.com/s@default. The voice was clear but unfortunately I don't understand

Hello all, It's me again. I would like play with calls between a H.323 client and a SIP client through * inside my LAN. For that, on host 192.168.1.20, I use GnomeMeeting (GM20) and Asterisk; on host 192.168.1.25, I use SJphone (SJ25) as SIP client on Windows and I register into * with a username, no password. The 3 files oh323.conf, sip.conf, extensions.conf are in attachment. In the same

Hi, I have read the wiki entries on IAX(2), but I'm afraid, it still have some questions: I have a working connection between two Asterisk-Servers (Asterisk 1.0.7-BRIstuffed-0.2.0-RC7k on Debian 3.1) via IAX. Does this connection work with IAX or IAX2? When I try to load chan_iax2.so, I get the error message chan_iax2.so]Oct 11 10:09:52 WARNING[2288]: loader.c:258 ast_load_resource:

I'm trying to somewhat reduce the security risk of Asterisk, by loading less modules. In my installation I use SIP and IAX2 for incoming calls, and that's it. No voicemail, no call parking, it just plays back voice clips. I can remove /etc/asterisk/modules.conf modules one by one: ------------------------------------ [modules] autoload=yes noload => pbx_gtkconsole.so noload =>

Hello, I am new to Asterisk, so RTFM answers welcome too (just include the FM's link :). I'd like to build a simple dialout server based on Asterisk. I installed 0.4.0 from package (a Debian SID machine, "server"). The client is gnophone (a Debian SID machine too, "client"). My modem is a GVC 56k voice modem connected to the server's serial port. I modified

Hi I?m a new user and I do test with my hardware . I have a x100p and telephone vozip. And when I run this command asterisk ?vvvvc for to test it . My computer show it ?warning? [chan_iax.so] => (Inter Asterisk eXchange) == Manager registered action IAX1peers == Parsing '/etc/asterisk/iax1.conf': Not found (No such file or directory) Apr 2 07:45:12 ERROR[16384]:

We have three remote call center Asterisk servers communicating with two central Asterisk boxes over a private IP-VPN with QoS. All systems were running Asterisk 1.0.7 communicating via IAX2 with little or no quality issues at all. Once we upgraded to Asterisk 1.2.4 call quality with IAX2 was horrific. We tried with/without jitterbuffer. We messed with every jitterbuffer parameter. We tried

We upgraded all five servers to 1.2.4. We tried trunking/notrunking. End users use an IAX2 softphone on their desktop PCs. Agents are VLANed and all IAX2 traffic is QoS'd on all LAN and WAN legs. Calls flow from the agents to the local Asterisk server as IAX2/ulaw. Then they went over the IP-VPN as IAX2/g729 (we tried ilbc and straight ulaw as well). Calls get to the PSTN from the

-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of Andrew Kohlsmith Sent: Tuesday, March 21, 2006 11:36 AM To: asterisk-users@lists.digium.com Subject: Re: [Asterisk-Users] Problem with chan_iax.cimplimentationcausesbadaudio? On Tuesday 21 March 2006 11:19, Adam Robins wrote: > All switches and routers give

All switches and routers give highest priority to traffic on IAX2 port 4569. We use DSCB values over the IP-VPN to prioritize it as well. This did not change with the upgrade, as we can still see proper packet coding. The softphone is provided by our vendor Aheeva. It is the same IAX2 softphone they use in their own call centers. Funny thing is that they say that moving to Asterisk 1.2.4

I received an e-mail from a vendor who says: "We have recently become aware of an issue in the chan_iax2 implementation of IAX2. This issue leads to degraded audio quality. Due to this we are urging everyone to move to SIP." I don't want to discount what this person is talling me, but I'm curious to know why I would only be having issues connecting to his servers, and also what

>> -----Original Message----- >> From: CentOS [mailto:centos-bounces at centos.org] On Behalf Of Rich Greenwood >> Sent: Monday, 18 May, 2020 08:34 >> To: centos at centos.org >> Subject: Re: [CentOS] ether-wake >> >> Some switch hardware can generate the packets directly, negating the need >> for a box on every VLAN. Meraki hardware can do it, but

Hi, I try to use oh323 package from inaccessnetworks for asterisk, but after make and make install that package, I have this WARNING message hwen a try to launch asterisk from shell command line...asterisk -vvvc... [liboh323wrap.so]WARNING[1024]: File loader.c, Line 235 (ast_load_resource): No load_module in module /usr/lib/asterisk/modules/liboh323wrap.so

I was just noticing that my asterisk box is trying to register with my providers on both IAX1 and IAX2. They do register on IAX2, but it seems like it keeps trying to register with IAX1 also. Is there a way to disable this behavior, or am I just misunderstanding it. I just want to make sure I'm not wasting processor time and bandwidth trying to register IAX1 when it is already

Hi all, I installed all needed RPMs by GnoPhone to be installed without problems but when attempting to install GnoPhone itself I get this message: # rpm -Uvh gnophone-0.2.4-1.i386.rpm error: Failed dependencies: mozilla >= 0.9.2 is needed by gnophone-0.2.4-1 libgtkembedmoz.so is needed by gnophone-0.2.4-1 libgtksuperwin.so is needed by gnophone-0.2.4-1 I'm using

I am having a really hard time getting gnophone working with asterisk. Gnophone tries to register with my server but there is no response. I can direct incoming calls to gnophone but if gnophone answers them, asterisk does not recognize it. Here is my configuration: iax.conf [jambo] type=user host=dynamic defaultip=136.159.99.100 permit=136.159.99.100 username=jambo secret=fubar

I have just finished installing all the pieces of Redhat Linux 9 (2.4.20-8), Asterisk-0.9.1, Gnophone-0.2.4 on my IBM T30. Audio card is SoundMAX Integrated Digital Audio. Not sure what the chips are. But everything works on both W/XP and RH9. (Machine is obviously dual boot) Everything starts, Asterisk is up, sound/audio is great for CD player, Volume controls, Voice recorder, RealPlayer, etc.