similar to: --backup-dir or --link-dest?

Hi, At one location we have a user whose Polycom IP430 suffers from erratic reboots. We swapped his phone for a brand new one, but the problem remains. Has anyone seen that?

Hi, How hard is it to integrate asterisk with Microsoft CRM? Thanks for any suggestions, pointers, etc.

Hello, We are using asterisk 0.4.0 on debian sid with Cisco 7960 and ATA186 phones. All sip entries have: callgroup=1 pickupgroup=1 However I am unable to remotely pickup a ringing phone using *8#. I get fast busy tone. Is there some flag to add in extensions.conf ? Thanks in advance,

Hi, I have 2 linux accounts on different machines (same login, same password). Can you please tell me how I use rsync directories between 2 accounts? Thank you.

Hello, On your BRI or PRI's what do you guys get as HANGUPCAUSE when dialing an unalocated number? I always get 3 (no route) which is less than helpful.

Hi, In our backup script we sometimes would like to override the common (i.e: static) excludes filter list. For example we exclude "- *.ext" for all backups but would like to include "+ *.ext" only for 'local' backups. Are such entries supposed to cancel each other? How can one override an earlier exclude in a filter list? Thanks,

Hi, I tried "upgrading" from debian's 1.4.21.2 package to vanilla 1.4.22 but then my TE410P alarms stay RED and no zap channels can be created, even if they are correctly listed by "zap show channels". I tried adding "dahdichanname = no" to asterisk.conf's [options] to no effect. Going back to 1.4.21.2 brings my alarms back to OK. This is with zaptel

Hi, This used to work fine in 1.4: exten => 2131/,1,NoOp(reject3: ${CALLERID(num)}) exten => 2131/,n,Playback(no_unknow_callerid_here) exten => 2131/,n,Hangup And now, after upgrading to 1.6.1.x it matches every callerid. Did something change? Thanks,

Hi, Is there a way to have markdown automatically convert obvious (http, mailto) URL's to links? i.e: http://example.com -> <a href="http://example.com>http://example.com</a> Thanks, -- http://www.critikart.net

Hi, If several phones register to the same sip.conf section what will happen with a "Dial SIP/shared" in asterisk? All phones ringing and the first one to answer gets the call? Undefined behavior? Thanks, -- Jesus is coming! Everyone look busy!

Hello, We are at wit's end on this. One (and only one) of our five asterisk installation is giving us real headaches. Buzzing and/or choppy sound interfere with conversations. I recorded some conversations with monitor() and no problem whatsoever appear in the recording, while the local user was hearing the buzz and half my words. This is a 1.2.16 installation with mISDN but mostly using

If like me you run * on a VPN (or multihomed) gateway and want to serve remote SIP clients, make sure you have bindaddr = 192.168.0.1 ; or whatever is your box's private IP otherwise * might bind to its public IP and send it as return address in the SIP call setup, which will (should) be rejected by your firewall. To * experts: might this setting interfer with NATed SIP clients? -- I

Hi, After upgrading to 1.5.2 I no longer can directly access to my voicemail by pressing the "Message" button, I have to go through the "urgent,new,old" report first. The oneTouchVoicemail parameter is set to 1 but not taken into account apparently. Anyone noticed that problem?

Hello, After reading all the docs and going through the menus, I still can't find the voicemail access button or menu sequence on the ST2030 (http://www.voip-info.org/wiki/view/Thomson+ST2030) Also I can't get phone provisionning through tftp to work. Configuration files are loaded but the phone seems to ignore them. Any idea?

Hi, In the samba debian package there used to be a very convenient utility to scan networks for open samba shares and weak passwords. Its name AFAIR is nat. Where can one find this tool? Is it still maintained? Is there a better alternative? Thanks, -- THERAMENE: Il veut les rappeler, et sa voix les effraie ; Ils courent. Tout son corps n'est bient?t qu'une plaie.

Hi, It seems rsync no longer resumes partial transfers after a SIGINT (CTRL-C). I tried the following: % rsync -avzP ~/video/Gone_In_60_Seconds_-_DivX.avi 192.168.0.3:/backup/DivX building file list ... 1 files to consider Gone_In_60_Seconds_-_DivX.avi 262144 0% 10.34kB/s 19:40:22 [CTRL-C] [testing remote size] % rsync -avzP 192.168.0.3:/backup/DivX/Gone_In_60_Seconds_-_DivX.avi

Hi, To push voice through a long thin wan (dsl) there are two choices: (1) have the cisco's (7912G) talk g729a to each other (reinvite=yes), or (2) have the cisco's talk to their local * in ulaw (reinvite=no), which talk to each other through a more advanced low-bandwidth codec (ilbc or speex) which is best? (2) would have more latency, wouldn't it? Did I miss a third option?

Having purchased a license for 5 g729 channels on Digium's web shop I thought registration and installation would be a snap. NOT. I followed registration instructions to the letter but it failed with that message: ERROR! Your Internet connection is probably behind a proxy and the Registration program can't communicate with our server Which is stupid as my * box is a firewall and sits

I current do some rsync backups with a command like so every day rsync -az -e ssh --stats --delete --exclude "stuff" / user@server:/home/user/ What I want to do is have some incremental backups in there in subdirectories. So, for example, something like this on the remote server /home/user/something.tuesday /home/user/something.friday I thought the --backup --backup-dir Switches were

hi all, What is the best Asterisk-compliant for Dell machine is recommended? I will have roughly 400 users in a production office. thanks!!