similar to: Service Level Compliance

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hello, I''m using HTB to shape my outgoing traffic over a ADSL-link with PPPoE with a nominal bandwidth of 128kbit/s. My goal is to favour small packets like ACKs and interactive services like ssh; in other words, I want to achieve low lantency. If there are some big packets going over the wire, everything works fine. But if there are

Hi all, I've just updated to current CVS, and have 2 polycom IP phones, one is a IP600 and the other is a IP300. The IP600 shows the status of the IP300 and a ZAP line quite nicely, but the IP300 won't show the status of the IP600.... Is there any additional debug apart from "show hints" to see why this might not be working ?? -= Registered Asterisk Dial Plan Hints =-

I am trying to download the latest release of Asterisk from: ftp://ftp.digium.com/pub/asterisk/ The link provided by Digium is incorrect for the Asterisk Tarball as there is no such file at ftp://ftp.digium.com/pub/asterisk/asterisk-1.0.7.tar.gz However the links for the Asterisk-Addons and other Tarballs is OK ftp://ftp.digium.com/pub/asterisk/asterisk/asterisk-addons-1.0.7.tar.gz Does anyone

I''ve been trying to migrate a win nt 4 machine to a xen domu for the past few months with no success. However, on my current attempt, the original hardware no longer boots, so I''m trying to resolve the issues with xen properly, or else take a long holiday... Anyway, the physical machine had a 9G drive (OS drive), a 147 G drive (not in use) and a 300G drive (all SCSI Ultra320 on

Has anyone been able to find a way to disable call-waiting on Polycom phones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050118/b7ea7b5d/attachment.htm

Hi, I am mostly happy with my Polycom IP600 but it apparently needs to check the FTP server every minute. I couldn't find any obvious setting related to that behavior in the configuration files. Any idea how to curb the IP600's spurious network activity? Thanks, -- Lord, protect me from your followers.

Hi, I intend to let several SIP-phones on my asterisk ring cascaded on incoming calls. First only phone 1 should ring, after 5 seconds phone 2 should ring in addition and after additional 5 Seconds phone 3 should also ring. How can I realize that correctly? Currently I do use Dial(SIP/1,5) Dial(SIP/1&SIP/2,5) Dial(SIP&1&SIP/2&SIP/3) But this seems not to work correctly on

Hi to everybody, seems that I cannot load the zaptel modules: ztdummy says the following: [root@Hayabusa misc]# modprobe ztdummy /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_unregister /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_transmit /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_receive /lib/modules/2.4.22/misc/ztdummy.o: unresolved symbol zt_register

I am trying to use the MYSQL command to insert a record into a database and I can't seem to get it to work. I can do an UPDATE with no problem. Here is the line in my dialplan exten => s,12,MYSQL(QUERY resultid ${connid} INSERT INTO `member` ( `id` , `member_num` , `active` )VALUES ('',${number}' , '1')) Does anyone have an example of an INSERT INTO that I could look

Hi All I've bought a TDM400P and need some help with configuration. Can you tell me what to do ? I've tried to install and the message below has appeared: [root@voip asterisk]# modprobe zaptel [root@voip asterisk]# modprobe wcfxo /lib/modules/2.4.22-1.2115.nptl/misc/wcfxo.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid

For those interested, I'm giving a talk about VoIP/enum.164/asterisk tonight in Sydney at the Sydney LUG meeting which is about 7pm in the UTS build #2, 4th floor, room 10. Sorry for the late notice, it didn't occur to me that there might be people on this list interested and able to attend etc... -- Best regards, Duane http://www.cacert.org - Free Security Certificates

I'm all up for reading and looking round for people in the same boat to try and solve the issue together, but there appears to not be large community yet, just the asterisk mail lists. I got Asterisk working with X-Lite great now for internal calls and also calling land line numbers etc. The two problems i'm currently having are: 1. When someone calls in the phones ring 3 times then

Hello. I'm trying to transfer calls from an analog phone (Zap/1, TDM400P card) to some other terminal connected to my Asterisk PBX. If I make a flash hook pressing the phone hangup button quickly it works as expected, I get a new dialtone and the other side is put on hold. But I would like to use my phone's "R" key instead for some different reasons (it's quite easier to use

Hello all, We recently configure an asterisk server to use with an VoIP provider to make calls to a PSTN. We use (voipjet, nufone, diamond....) We feel that we haven't got the quality that we hope. Sometimes our calls gets mute, or we feel communication cuts on our phone calls. We have got an QOS router (Draytek) reserving 1/2 of our wideband to the SIP an IAX2 protocols, and an ADSL line

I've noticed that some callers listen to our main menu and don't press any keys. I have it set up to restart the menu a few times and eventually hang up. I'm wondering if these are wrong numbers (in that case, why don't they hang up) or they really want to speak to someone here but don't understand the menu (what's so hard about "for the operator, press

-----Original Message----- My question concern outgoing calls. How can I configure my extensions.conf to get a PSTN line on my TDM04B card in the following order : first trying on the channel 4 then if 4 is busy then switch to 3 if 3 is busy then switch to 2 and if 2 is busy then say there's no more line available. I don't want to dial on the first channel as it's my main number

I am managing two Xen containers which are exhibiting a spooky behavior. There are intermittent network failures that affect only the VM''s, and not dom0. This exhibits itself as problems with nfs locks, snmp, and even icmp ping. The arp tables look normal on both the dom0 and the guest. However, it is definitely arp related. Causing arp traffic of any sort results the problem curing

i installed cacti following the INSTALL file but if i try to view cacti url it shows this error: Warning: mb_ereg_replace(): mbregex compile err: premature end of regular expression in /var/www/html/cacti/include/config.php on line 43 Warning: mb_ereg_replace(): mbregex compile err: premature end of regular expression in /var/www/html/cacti/include/config.php on line 44 Warning: Cannot modify

Hello all, After reading through the Wiki and archives, I decided to take a stab at installing * on Solaris 9 SPARC. I checked it out via CVS last night as well as about an hour ago, and have run into a compile problem that I can't quite figure out. After running into some unlisted dependencies, such as popt, things are almost compiling. Right now the make bombs when trying to find setenv

Hello Just installed fresh Debian testing box, checked out Asterisk and others from CVS stable (-r 1.0), and now trying to 'make install' in Asterisk. I get this error: if [ -d CVS ] && ! [ -f .version ]; then echo CVS-v1-0-03/29/05-15:19:53 > .version; fi gcc -g -o asterisk -Wl,-E io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o