similar to: IAX pauses

I'm having trouble with a WX100USB adapter and a Siemens Gigaset cordless phone. If I select fxols as a signalling method, calls are being disconnected. Usually after about 4 minutes, and asterisk just says that the phone has hung up. If I choose fxogs, I immediately get a LINE IN USE message on my phone and I can't even get a dialtone. If I choose fxoks, it mostly works, but sometimes

I'm having an issue with a connection between two sip phones, specfically sjphone, The two phones connect just fine, but after about 5 sec the call is dropped. On a side note a call does'nt got dropped between sip/sjphone and the outside line with a wx100p card. The communcation is on a full 100mbit network. I have a text file of the debug output of the call. Kevin,

Hi, I'm experiencing brief pauses during my calls: 0.5-1.0 sec of silence if call continues for more than a few minutes. I'm sure that problem is in the phone (a cheap ATCOM AT-320 with latest SIP firmware) but I'd like to diagnose better. During a little test, it seems that there is no problem with IAX2 firmware (but there are others... I'm not able to transfer and pickup

Hi all My * implementation is working brilliantly with only one small fault left to kill. I'm using IAXTalk from Telappliant for my incoming/outgoing calls to the pstn network; if I set my codec to GSM everything works great - no pauses but quality is a bit poor. If it set the codec to alaw (I think I'm using the correct one - I'm in the UK) I get intermittent pauses on the call.

I know, that jitterbuffer should be set at receiving side and on outgoing call leg, ie. if sipphone calls to asterisk and outgoing to zap chanel, I should set jitterbuffer on zap channel (to dejjitter audio stream from sipphone) but what about pure voip situation (i.e. iax-iax, sip-iax, skinny-iax etc.)? I have following setup (homeworkers using sip phone connected to home asterisk via SIP and

I have looked through the archives, and can only find old references to this problem that appear to be no longer relevant, so I thought I'd ask again. I am having a problem with periodic breaks in audio over an IAX trunk. The interruption only happens in one direction, and (I think) only with clients built on the open source libiax. Codec is irrelevant, and jitterbuffer on/off seems to

Hello list, I've got a problem (maybe only a problem of understanding how * works) with IAX and WaitExten. To simplify the problem I've brought it down to the following scenario: - 3 Asterisk Server A,B and C (central). - A and B both register with C. Now I want to be able to dial an extension at A to become connected to C and there I want to dial an extension to become connected to B.

We have a customer who is connected to our PSTN gateway using IAX and noticing that even when the traffic from their site is modest their outbound audio has short dropouts. Inbound audio is fine. (They have ADSL so it is expected that outbound audio would be the first to experience problems.) We have several questions to pose to the collective wisdom of this list. Q1: Are there any statistics

Hello everybody, I have a FreeBSD 4.8-STABLE server running Samba 2.2.8a and a workstation running Windows 2000 SP4. Whereas FTP transfers between these boxes average 700 KB/s (10 mbps LAN) in both directions, Samba transfers are exhibiting this odd behavior: Windows 2000 --> Samba = 700 KB/s (perfect) Samba --> Windows 2000 = 100 KB/s (terrible, inconsistent, with LONG pauses) Trust me,

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

I've been curious how IAX has been working for users running without a jitterbuffer and no echo cancellation. In my particular installation, IAX2 traffic remains within a pretty high bandwith WAN ( - but busy as it is linking buildings in a school district). It seems that at most times, things could run quite nicely in this mode. The ability to dynamically disable echo cancellation when end

Hello, I have a 3 server asterisk configuration where one asterisk (say A) (v 1.4.25) has a digiuim card connected to E1 from which calls are routed to another asterisk server (B) (1.6.0.9) over IAX trunk from which calls get routed to third server (C) (1.6.0.9) again via IAX trunk. SIP clients are connected to third server. A is the PSTN termination server, B runs the menu and AGI and C is

I'm trying to setup zoiper ( formerly idefisk ) to use my asterisk server at work from home. I've setup zoiper for iax, set the ip address to work's fixed ip address, user: home, password: password but the zoiper log shows: 11:02:35 Rejected registration for 'home@<my-office-ip-address>' with cause 'facility rejected' 11:03:35 Rejected registration for

<!DOCTYPE html PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> </head> <body bgcolor="#ffffff" text="#000000"> <div class="moz-text-flowed" style="font-family: -moz-fixed; font-size: 13px;" lang="x-western">Hi, <br> <br> I try to find a solution for this problem : <br>

I'm using iaxagent on a Droid X to connect by IAX to 1.8.3 at the office. 1.8.3 has sip phones. The audio is fine on the Droid X side. On the office side, they hear an echo of _their_ speech, not mine. The office uses sip-providers generally without any echo problem. Where do I start to figure this out? How do I narrow it down? Can I figure out if it is an iaxagent problem? Could using

>Thanks a lot for your reply. Alright, it was not a >bug in Samba after all. You were right, I was wrong, >excuse me for blaming Samba and being an idiot :) >But please read on, because I'm still lost here. It's all right to feel repentant but don't exaggerate. You are OK, only a little stressed. >As I expected, this was not an issue with my physical >network

Hi My first post get no answer :=<, i post new with new elements. I have two Asterisk server, running on Asterisk 1.6: SRV1 = 192.168.0.5 on Asterisk 1.6.1.4 SRV2 = 192.168.0.20 on Asterisk 1.6.1.8 I want create a link for exchange call. on Srv1: iax.conf: [general] bindport=4569 bindaddr=0.0.0.0 language=fr bandwidth=low jitterbuffer=no forcejitterbuffer=no autokill=yes

I've set up a 1GB tmpfs filesystem on a system with a single IDE disk and 2GB's of memory. I'm storing a large amount of RRD files (~300MB) on the tmpfs filesystem to make their generation a bit speedier... this part works great. However, I want to rsync these files over from time to time to a directory on the local filesystem (same physical server). I'm using rsync 2.6.4pre3

rsync pauses during transfers. That is, while I'm watching the verbose output, it will run through lots of files and then pause at an arbitrary point and do nothing for around 3-15 minutes and then it will pick up again. I'm using both -v and --stats. Why would this be? Is it scanning the directory? Both source and destination have rsync 3.0.8 and FreeBSD 8 -------------- next part

With C:\Users\User>R --version R version 2.14.0 RC (2011-10-23 r57410) Copyright (C) 2011 The R Foundation for Statistical Computing ISBN 3-900051-07-0 Platform: x86_64-pc-mingw32/x64 (64-bit) running download.file('http://www.ebi.ac.uk/microarray-as/ae/files/E-TABM-25/E-TABM-25.raw.1.zip', tempfile()) sometimes gets 52% (according to the dialog box) of the way through, and then