similar to: hardware requirements of asterisk

This is ifconfig on openbsd box: fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 I think this output shows that the fxp0 interface is on simplex mode. The voice degradation I referred was by using xlite soft phone. I open 2 line similtaneously and dial to FWD and back to my incoming extension. Xlite is runnning on a w2k box with realtek 100M nic in auto mode. I can

Hi all, I''ve just built Mandrake 10.1 on a Compaq Deskpro that I''ve built as a router/firewall and am redirecting port 80 outbound to force users through the Content Filter. I''ve run this setup on Mandrake 9.0 and 10.0 without any problems but this time the following happens. Squid is accessed through port 3128 and Dansguardian via 8080. If I set my browser on a

In relation to voice degradation when having 2 or more connection to Asterisk. The comment on the network setup is quite possible. I am not too familiar with linux. How do I check whether the asterisk server's nic is running at full-duplex mode. Does Asterisk use the sound card on the box to do voice processing? I am running xlite on 2 pc and making calls through iax, FWD and back to my

On Thu, 2004-01-15 at 19:18, dkwok@iware.com.au wrote: >> I do not have any zaptel hardware on the Asterisk box, I could not have >> meetme functioning. I did modify the Makefile in zaptel directory on >> line 168 by including ztdummy as one of the modules to compile in. try modprobe ztdummy This works. Should I include this in /etc/asterisk/modules.conf so that it will

Chris Albertson wrote: |What firmware version do you have? program version 1.0.4.39 -- David Kwok Iaxtel/FWD # 17001813482 ext 1002 -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/x-pkcs7-signature Size: 1878 bytes Desc: S/MIME Cryptographic Signature Url :

For people who are using GS 101, what do you think the sidetone generated by the phone. I find mind a bit annoying. It has a delay and you notice it as an echo. The volume of the sidetone is also quite hight. I am distracted when both caller and called party talking over each other occasssionally. The volume of the sidetone can be turned down using the volume button but it also control the

I do not have any zaptel hardware on the Asterisk box, I could not have meetme functioning. I did modify the Makefile in zaptel directory on line 168 by including ztdummy as one of the modules to compile in. The error message from the concole: -- Executing MeetMe("SIP/1002-e9ca", "4700") in new stack == Parsing '/etc/asterisk/meetme.conf': Found

I have just checked the Openbsd box on the if interface. fxp0: flags=8843<UP,BROADCAST,RUNNING,SIMPLEX,MULTICAST> mtu 1500 address: 00:02:55:30:54:28 media: Ethernet autoselect (100baseTX full-duplex) status: active inet 192.168.1.1 netmask 0xffffff00 broadcast 192.168.1.255 inet6 fe80::202:55ff:fe30:5428%fxp0 prefixlen 64 scopeid 0x1 xl0:

I configure an old Compaq Deskpro workstation to boot LTSP using PXE and DHCP. I have an error that say TFTP Open Timeout after this workstation got it IP Address from DHCP server. I have done some configuration settings, and this workstation can load the kernel (vmlinuz-2-4-18-ltsp-1), but after loading the kernel, there only flashing characters and colors on the screen. I also have do chmod

I'm on redhat and libxml2 is installed: [root@deskpro davidcam]# rpm -Uvh libxml2-2.4.23-1.i386.rpm warning: libxml2-2.4.23-1.i386.rpm: V3 DSA signature: NOKEY, key ID db42a60e Preparing... ########################################### [100%] package libxml2-2.4.23-1 is already installed I complied ices/icecast fromm src - do I need any addtitional flags in ./configure?

I need a cheap platform for installing a tdm400. Could someone tell me if the cheap cpubuilders computer at sams $179 (cbs110l) is pci 2.2 compliance? I ve got a compaq deskpro en 700 that does not seems to be compliant and I need to change it to start developing. Thanks for the help. Computer Model: CBS110L --------------------------------- Do you Yahoo!? New and Improved Yahoo! Mail

Hello all, I am using a headset and the X-lite softphone (sometimes I use IAXComm, but I'm having difficulties using OSS emulation with it) to connect via uLaw to my internal Asterisk server, which is a Pentium II 400 with 128 megs of RAM. After getting this headset, most or all of the echo people on the other line were complaining about is now gone, according to them. However, every

Referring to my previous post about degradation of voice quality when having more than 2 connection. The actual route is: pc xlite -> local asterisk box -> iaxtel -> local asterisk I have tried out a different situation: pc xlite -> local asterisk box -> iaxtel and the second connection pc xlite -> local asterisk box -> iaxtel -> local asterisk The same degradation

My ADSL connection speed is 512Kb up and 128Kb down. When making calls from Asterisk to IAX and back to the Asterisk, the sound is choppy and 20% of voice messages was lost. What is the production bandwidth requirement per internet call. I understand there is no guarantee of QoS but at least a benchmark to follow. -- David Kwok Iaxtel/FWD # 17001813482 -------------- next part

Just got GS 101 phone and plugged into the network. Got ip setup however, the following problems arise: 1. when dialing an extension, I cannot further send any key tone to Asterisk. 2. there is no sound coming from the other end. I have a sip.conf setup for GS: [General] disallow=all allow=ulaw allow=alaw [gs] canreinvite=no dtmfmode=info In the GS101 setting rtp port = 5004 sip port = 5060

Does any one have any recommendations on an off-the-shelve PC for an Asterisk Server? This is for a proof of concept, so it needs to be inexpensive. I have tried 2 different PC's and had problems with the sound cards. I am thinking of PC's I can buy from local dealers like Best Buy, Office Depot. SO a cheap HP, Compaq or eMachine would work fine for me. --Steve

Any thoughts on the following? I am running asterisk from CVS (downloaded yesterday's version, just to be sure) on a test system with no digium cards in it, so I have installed ztdummy (see logs and screenshots below) as a timing source. When I call the music on hold extension from a Sipura Sip connected analog phone, I hear nothing and start getting Warning[98310]: chan_sip.c:674

I am running v2.0.7 and have set up a network drive for an accounting ledger system. The software is called MYOB and is quite popular in Australia. This is the first time I have to deal with multi user access to 1 data file. My setup is: Global oplocks = yes socket options = TCP_NODELAY socket options = IPTOS_LOWDELAY [MYOB] path=/home/office/MYOB force group = office directory mask = 0770

Any one has documented how-tos for making voicetronix openline 4 to work with Asterisk. I have been contacting Australian Digium resellers and Digium cards are not approved in Australia. So I suppose Australian users are interested into putting Voicetronix in use. Any expereience to share will be most appreciated. David Kwok -------------- next part -------------- A non-text attachment was

Hello, The harddisk of my etch/bristuffed asterisk1.2 box finally died. I moved the cheap (1397:2bd0) HFC-S card to a squeeze host (i686) and built dahdi modules 2.3.0.1 using m-a. After zaptel->dahdi and asterisk 1.2->1.6 config adaptations, everything seems ok, except for the BRI side, unable to bring layers 1/2 up. Asterisk reports: chan_dahdi.c:12393 dahdi_pri_error: 1 Unable to