similar to: How to prevent a call from going to voicemail when one phone is offline?

Most people seem quite positive about Snom phones, I cannot share this opinion. The displays are dying quite often, and firmware is buggy. I have tried every firmware from 4.5 up to 5.x and 6.04 but keep having problems with phones locking up or rebooting during an ongoing conversation. REALLY annoying for a phone that is advertised / targeted as a business class phone

Is it possible in any way to use an Outlook contacts database as the source for the internal Asterisk database that is used for callerid lookups? Thanks!

Hi List! I installed Asterisk 1.0.3 stable on a RHEL rebuild. Due to problems with * modules refusing to build I replaced the RHEL kernel with stock 2.6.10. Asterisk seems to be working but when I dial voicemail I hear nothing. When I hangup I see a message on the console that the calller did not specify a mailbox number so I guess voicemail app is working. The phone(Grandstream BT100) is

I would like to create a rock solid asterisk server that comes up no matter what. Ofcourse I can consider hardware raid1 but.... for the cost of a hardware raid controller I can also buy a 2GB compact flash card that doesn't produce any heat or noise and is friendly to the electricity bill and our environment :) 2GB is more than sufficient for a linux distro and an * installation but

Hi list! I'm sure the topic has been discussed but I could not find what I was looking for. What would be the best / easiest VPN software solution. I would like to install vpn software on the * server for roadwarriors to connect to with laptops running windows. Ideally the vpn solution will not require any additional software on the client side but will use IPSEC. (Ofcourse call quality

Hi list! I have some sip phones and Sipura ATA 2000's. However after dialling a number I need to dial a # to control a device. When I dial # Asterisk kicks in and puts the call on hold. How can I change this? Thx!! Remco

Hi list ! Is it possible to show the used Useragent of a peer that registered with Asterisk? It's being saved obviously because the console says so when a phone is registering but sip show peers doesn't show it? Is there any other way to view it? Thanks!

Hi list! What exactly is the meaning / function of the pridialplan & prilocaldialplan? I've been trying to find out what the different possibilities for these settings are but couldn't find a clear answer. The possible parameters I could find are are : local,unknown,dynamic,national,international and maybe there are more? Thanks!

Hi! I'm trying to setup a Wildcard TE110P with a PRI in The Netherlands. I get a Red Alarm on the line. Is there any way of debugging this? I've tried some configs that should work but without success. Is there any way of telling if the cabling is correct or what else the problem could be? Thanks!! Remco

Hi, Snom phones firmware 5.0 is now out. Try it if you like: http://www.snom.com/wiki/index.php/Main_Page. Regards, --------------------------------------------------------------------- Usman Tahir snom technology AG www.snom.com --------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi list! I'm trying to setup a new * server with a TE210P in a Dell 2850 box. I'm using zaptel version 1.2.5, the linux flavour I'm using is CentOS 4. In a previous thread I read about the results I should expect from zttest. On my home box (using the crappy Asus A7V600) I got really bad results from zttest (just over 97.5) but I know that this motherboard just sucks. To my

I have Asterisk 1.0.5-BRIstuffed-0.2.0-RC5 up and running. Everything seems to be running fine but after some time asterisk just goes crazy (even withouth any incoming or outgoing call activity perviously). If I leave the box up for some time * goes haywire and the console is flooded with this message: Ouch ... error while writing audio data: : Broken pipe At that time I can see that there

I applied the florz patch but my problems remain. Now I get all sorts of weird errors on the console and I cannot make outgoing calls. Incoming calls work as expected. I am using a single HFC-S card with BRI. Any clue what these errors below are? Ri = 44651 TEI msg = 3 TEI = 7f Ri = 3800 TEI msg = 3 TEI = 7f Ri = 42399 TEI msg = 3 TEI = 7f Ri = 42409 TEI msg = 3 TEI = 7f Ri = 22078 TEI msg = 3

Hi, I read a lot about US providers that can terminate a PSTN number for you and offer IAX or SIP connectivity. Does anyone know such a company in The Netherlands ? I read about Unet. Anyone with experience with them ? Any information is welcome. -- Michiel van Baak http://lunteren.vanbaak.info michiel@vanbaak.info GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D

Hi I dont work much with syslinux and linux. But in my last test with pxelinux I woundert why the timout function is ignored by pxelinux. There is no hint in the documentation or in the web. I've figured out that the timout function doesn't work if the prompt setting is 0. timout 50 prompt 0 doesn't work I've tested it in versio 3.07, 3.06 and an "unknown" old

Hi list! I have a strange echo problem. Two days ago I setup * 1.0.6. at a friend of mine. Just an * server and for outbound calls wengo.fr was used to place calls via sip. He had a strange echo on the line I didn't experience on my setup. Today I upgraded my asterisk 1.0.5 to 1.0.6 and suddenly I have an echo too on sip calls thru wengo!! I already verified wengo was not the source of

Hi! I installed shorewall shorewall-2.1.7-1 on a newly installed box. When using 1.4 versions I could enable shorewall in ntsysv, now shorewall doesn''t show up in ntsysv? I am running shorewall on a RedHat Enterprise Linux rebuild (www.taolinux.org). Is this a bug or a feature?

I installed bristuff0.20-RC3 (which includes * 1.0.3 stable) It works fine until I disconnect the phone jack for the ISDN line. Even after plug it back in asterisk still reports that it could not create a zap channel when I try to dial out and the line gives an engaged tone when I try to dial. Re-starting asterisk doesn't solve this, I have to stop asterisk, unload the modules, reload

Hi all, I have 2 CentOS 3.5 boxes & i do backups on the LAN Recently, I always get FTP timout error with large files "700 MB and more" . i tried to log and do manual backup with FTP command and i also got the same result "FTP timeout" i tried to google and i did # modprobe ip_conntrack_ftp but still get the same timeout message Any one can help??? Hameed

Hi list! Is anyone using the Kirk IP600/3 with SIP firmware connected to Asterisk? Any experiences / caveats? If anyone would be willing to share the dump of their IP600 config file, i would really appreciate it. Is there anything special i should put in my asterisk config? Thanks !!! Remco