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I''m having troubles with my outbound VOIP connection. I''m convinced that I don''t have QOS/traffic shaping configured properly in my shorewall linux firewall, which serves as my Asterisk VOIP server and Internet router/gateway. I don''t have a separate router box. I''ve been using VOIP for about a year now, but just recently realized that I need to

> he graphics card is emulated in Qemu, the graphics operations are > converted to VNC commands there, and you see on the VNC display what > would appear on the Windows VGA display. OK, Thanks. 2 other questions: 1. how do I connect tzo the VNC server for my domU OS? 2. Where is the QEmu code? In xend or in kernel? -jec Jean-Eric Cuendet Senior developer / Technical support Riskpro Technologies SA Av. Louis-Ruchonnet 2 Web : http://www.jesc.ch -------------------------------------------------------- ____________________...

I'm using 1 BudgeTone 100 IP Phone and a Sipura 2000 for all my old analog phones. The analog phones with the Sipura seem to work great. Voice quality is fine on both ends on the Sipura. I'm using the Teliax service and I use the Ulaw codec for all phones. However, I'm struggling with the BudgeTone 100. On my end, I find there is lot's of voice cut outs. I'm told my

Many times the "speed" of an inbound voice call changes. It's similiar to playing a 33 LP at 45 speed. Sometimes the voice becomes uneligible. A "speed" change is the best way to describe it, seems like the voice packets are being played out too fast. Can anyone explain what might cause this? It doesn't always happen, and seem unpredictable. Thanks, Jim

...;: /share/usr/src/openssh-1.2pre12/login.c:92: undefined reference to `login' make: *** [sshd] Error 1 I hope this is useful to someone. Unfortunately I'm NOT going to any development work on this. Bug reporting is on the edge of where I'm willing to go. Greg Weeks -- http://durendal.tzo.com/greg/

...er than that of the budgetone. I find myself wanting to use the analog phones instead of the IP phone, which drives me nuts :-) So, I'm looking for advice on purchasing my next IP phone. Good voice quality with the ablity to display CallerID on the LDC panel. Thanks, Jim <http://duda.tzo.com:1024/webmail/src/read_body.php?passed_id=1305&ent_id=1& mailbox=INBOX.Sent&sort=6&startMessage=1&show_more=0&passed_ent_id=0> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/...

I've upgraded to asterisk-1.6.0-beta2. I'm trying to get the new Telco MWI detection function working. It doesn't appear to be working. I have this in zapata.conf ; PSTN connected here ;immediate=no ;busydetect=yes ;busycount=8 ;musiconhold=default mwimonitor=yes ;mwilevel=512 mwimonitornotify=/usr/local/sbin/zapnotify.sh faxdetect=incoming signalling=fxs_ks context=incoming

How do I get a context to continue to execute commands after the caller hangs up after a Dial command? I'm using the "e" option to the Dial application. I though the "e" option would allow the context to continue. This doesn't want to work for me. I'm using asterisk-1.6.beta5 I never get to "3" below. I get a message saying the "2" ended

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I use Asterisk in my house. Each phone is a different extension. I really like the ability to have multiple simultaneous calls in the house. However, I do miss being able to be able to pick up a phone in a different room. Currently, I have to either transfer the call or transfer the call to a "conference" extension to move around the house. While a connection in progress on one

Hi, I know this request is a a little off-topic but may be interesting for dovecot users (at least i hope!) Does anybody has already had an experience with a calendar sharing solution including a webcalendar and an outlook connector, "compatible" with dovecot ? Today, there are many projects aiming to be a "M$ Exchange Killer" (such as Zimbra, OBM, open-exchange, etc) but

I run CentOS 4.1 on a remote server and did a minimal installation. My problem is that I can not login using the HOSTNAME, must use the IP. 1. edited the - /etc/sysconfig/network 2. changed - HOSTNAME=a266 3. restarted the network Looks ok, hostname appears when logged on via IP, but can't use the HOSTNAME to log on from client. --- [kai at localhost ~]$ ssh a266 ssh: a266: Name or

Hi, I want to run a windows machine on my CENTOS 4.3 laptop 1.8 GHz 768 Mbytes RAM. first what should i choose VMPlayer or VMWare server? Second where can I find some guidelines to have this done? thanks, Guillermo Garron

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I'm struggling to get my VOIP installation to be acceptable. I'm looking for advice on what else I can look for. My system: o Teliax VOIP service, voip-ny1 proxy o RCN Cable Internet Service (3Mbps download, 500kbps upload, 6ms average jitter) o 3.2 GHZ P4 Server (runs asterisk, firewall, other stuff) o server lightly loaded o Linux kernel 2.6.19.2 o Shorewall Firewall software with

Hi - I've been using Hastymail with my Dovecot server for awhile now. However, I have noticed that at some point Hastymail stopped showing the number of unseen messages in its folder drop down list, it just shows 0 for the folders that I have it checking for unseen messages. I'm not sure when it broke, but I think it's when I went to one of the testing builds several months

I use the Teliax service with the IAX2 protocol. I noticed 2 days ago that I was not registered with the Teliax server. I used the "iax2 show registry" command and found I was not registered with Teliax. I issued a reload command in asterisk in order to connect again. I went to the Teliax website and noticed a message which stated clients may need to reboot due to changes made with

I'm using the teliax service and asterisk for my home phone system. I'm quite pleased. So pleased, I dumped my PSTN service and ported my telephone number. In doing so, I of course lost CallerID from the PSTN. My wife really liked having the CallerID box. In its place, I'm using the AGI interface for CallerID directly from Asterisk, announcing the calls verbally using festival