similar to: IAX TO IAX call between two registered servers

The value of 14800 is correct. I had issues with my TDM400p with 2x FXO's installed and using the Xlite client. I could not get the echo stable at the initial call. Changing to a hard phone made everything work correctly. I still had problems with the off location I called, but mostly worked great. To see the TX gain I created an extension with the Milliwatt command attached to it and called

Hi! I'm here again with echo canceller problem... :-( I think I've done everything to enable echo canceller feature, but it still doesn't work... Can anybody tell me if there is some error or something missing in this configuration please? I'm using Eicon Diva Server 4Bri. http://www.eicon.com/worldwide/products/MediaGateways/disv4bri.htm?techspec=1&regID=4999 Card

I noticed the following: Suppose that we have a server called "SMALLSERVER" working as a PDC for "SMALLDOMAIN**". When I enter "net getlocalsid" I get the following output: SID for domain SMALLSERVER is: S-1-5-21-xxxxxxxxx-xxxxxxxxx-xxxxxxxxx But when I enter "net getdomainsid" I get: SID for local machine SMALLSERVER is:

Being one the many people trying to track down echo 'ghosts' I ran across this page: http://lists.digium.com/pipermail/asterisk-users/2004-November/071301.html suggesting ways to adjust the gain. I have a TDM400P 2x2 config with Kewlstart lines configured. I've located a local telco milliwatt test line, and when I call it, the gain numbers are no where near 14844. Now, this

Hi, Try: ?vec1<-c(1,1,1,1,1,1,1,1,1) if(all(vec1==1)) "xxxxxxxxx" else? "yyyyyyyyyyy" #[1] "xxxxxxxxx" ?vec2<-c(rep(1,4),2) ?if(all(vec2==1)) "xxxxxxxxx" else? "yyyyyyyyyyy" #[1] "yyyyyyyyyyy" #or if(length(unique(vec1))==1) "xxxxxxxxx" else? "yyyyyyyyyyy" #[1] "xxxxxxxxx" ? if(length(unique(vec2))==1)

An attacker is scanning my Asterisk Switch to gain illegitimate access to VoIP call functionality. Using a sip scanning tool, *it* sends REGISTERs with random identities. And when it discovers one identity subscribed in my switch, it tries to authenticate with random passwords using this user name. For the moment, I have replaced this account. And also blocked the IP it has used but each time

Good day list, I am feeling extra stupid this Monday morning and am hoping someone can come to the rescue. I am trying to use the ztmonitor utility on my wildfire 4 FXO card. and have read the following from the wiki. *********Wiki start******** If you set this to yes, use ztmonitor to adjust the rxgain and txgain. Ztmonitor isn't installed by default; but it is included with the Zaptel

I'm seeing this: 1 UID FETCH 31734 (ENVELOPE) * 23 FETCH (UID 31734 ENVELOPE ("Fri, 2 Mar 2012 19:05:24 -0500 (EST)" "XXXXXX" (({22} XXXXX \"X-XX\" XXXXXX NIL "XXXXXXX" "XXXXXXXXX.XXX")) (({22} XXXXX \"X-XX\" XXXXXX NIL "XXXXXXX" "XXXXXXXXX.XXXXXX.XXX")) ((NIL NIL "XXXXXXX"

I tried to connect my * to IAXtel, but i always get this errors. chan_iax2.c:5849 socket_read: Registration of 'mnetwork' rejected: Registration Refused On dial a iax number i get: chan_iax2.c:5526 socket_read: Call rejected by 69.73.19.178: No authority found chan_iax2.c:5528 socket_read: Immediately destroying 3, having received reject chan_iax2.c:2411 iax2_hangup: We're hanging

Hi guys, I've registered some g729 licenses, during register process everything worked fine. astk2*CLI> show g729 0/0 encoders/decoders of 20 licensed channels are currently in use But I'm not able to use this codec. I'm trying to use a linksys PAP2 to talk using g729 but I got this answer from asterisk: Got SIP response 488 "Not Acceptable Here" back from

I came into this with my eyes wide open. I have read ABSOLUTELY EVERYTHING there is to be found on the net about avoiding echo problems BEFORE I even attempted to create a production system. Since lots of people are apparently using this in production environments now I just assumed that echo IS avoidable. As others have recommended, I created a test system with the proposed production

asterisk-users-bounces@lists.digium.com wrote: > In the following setup: > call coming from a pstn line -> into FXO card -> asterisk -> SIP > phone > > i get an incredible loud echo in the SIP phone (about 0,5-1s) > (everything i speak into SIP phone microphone i hear in its > speaker). The person calling from PSTN is not getting any echo. Make sure you're not

I have looked all over the site(s) for help. But heres the problem. Im missing something. In coming works fine from FreeWorld via IAX. But when Dialing out i get: May 13 13:42:01 WARNING[1150495040]: chan_iax2.c:5256 socket_read: I don't know how to authenticate iaxtel to 65.39.205.121 my IAX.conf if as follows [general] port=5036 register => ######:xxxxxxxxxxxxx@iax2.fwdnet.net

I *think* I may have encountered a bug, or a feature, in the idmap/winbind area. I have recently added rfc2307 attributes to my AD, and am in the process of switching over. This means that I still have (unintentionally) some files/directories/etc. around with old UIDs e.g. 3000007, rather than my rfc2307 specified UIDs. What I am seeing is that the SID2XID mapping is initially correct for a

Hello, I am trying to connect an asterisk 1.6 to a trunking plate forme. With asterisk 1.4.x I added to sip.conf a line asking for registration in the form of: register = XXXXXXXXX at domain.com:Password:XXXXXXXXX at domain.com<assword%3AXXXXXXXXX at domain.com> @domain.com Unfortunately, as you can see, my usernames have to be of the form XXXXXX at domain.com which means that I had to

Hello, After solving the other "low hanging fruit" audio issues in our Asterisk PBX, we are left with occasional cases of severe echo which we have not found a solution for yet. Our system: - Asterisk 1.2.0-beta1 - TE110P on a PRI - TDM04 and TDM40, but these are unrelated to current echo issues - Fedora core 3 - Echo canceller KB1 Most calls have minimal, acceptable echo levels. But

I am currently running Asterisk 1.2.8, on an AMD Sempron 3100+ (ASUS K8N Nforce based board). I am using a Digium wctdm24xxp to terminate 8 (2x quad) FXO lines. Using ztmonitor (From zapata) I cannot see that there is any registerable incoming volume from these lines. I've been running them at rxgain = 25 (zapata.conf) to make the audio audible, however this creates poor call quality issues

I'm replacing a SoundPoint IP 600 with a SoundPoint IP 650. I attempted to simply reuse the existing config files for the old phone on the new phone, but the new phone would lock up on the 4th digit when attempted to dial out certain numbers. So, I downloaded the newest firmware and config templates from Polycom, and attempted to migrate the settings. Seems I'm missing something from

Has anyone seen the error that I'm getting below? After you ssh to the box and enter the password the connection just closes. SSH Version { root at xxxxxxxx} # ssh -V OpenSSH_3.7p1, SSH protocols 1.5/2.0, OpenSSL 0.9.7b 10 Apr 2003 OS Version { root at xxxxxxxx } # oslevel -r AIX 4330-09 SSH Connect Failure { root at xxxxxxxxx } # ssh -l xxxxxxxxx -v localhost OpenSSH_3.7p1, SSH protocols

Hello, I have set up winbind to authenticate linux pc's to a windows 2003 AD. The authentication works, but the performance is not good (takes over 5 minutes) PRELIMINARY ----------- OS: ubuntu 7.04 Samba: 3.0.24 AD: windows 2003 ANALYSIS --------- After analyzing the log.winbindd file in log level 10, I can see three major parts 1) lookup and authenticate the user -> performance OK