similar to: can a sip.conf stanza be shared by several phones?

Hi can i set up a group of SIP users and forward a call to it? I am looking for a group, not for a queue. I won't listen any musinc on hold, and i won't that someone has to pay if nobody of the user's in the group accept the call. Can i do that? Thanks to all

Hello. I'm a new Asterisk user, but I'm impressed with the flexibility and versatility of Asterisk, and am moving quickly to adopt it's main-line use in our company. Hopefully, you'll be hearing more from me as the project moves forward. Right now, though, I have a question about SIP peer registration. Right now, for our SIP-based phone,s, we're using the Sip Express Router

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?

That could be it. What I want to do is set a group of callers and have the event cause the phone to ring them in order. I will tie it to my IVR portion and thus I can make sure peole in sales get calls based on our hierarchy in the office. So if I am reading your example right the syntax is.... Exten => 501,1,Dial(SIP/PHONE1&SIP/PHONE2&SIP/PHONE3), rtf) Is that a valid way to cause

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

I want the callers don't join in a queue when the agents are busy. I suposse it is easy but i can't get the solution for this. Can you suggest me something? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081229/3545ab8c/attachment.htm

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.

Actually doing both sounds good to me. Can you explain further about ringing them all at once? Here is how I tried to make mine work and failed... {global} PHONES0=SIP/2000 PHONES1=SIP/2001 [local] exten => 6001,1,Dial(${PHONES0&PHONES1),20,trf) When I dial 6001 I see my debugger tell me that I am using the wrong syntax. Do you know the correct syntax for ringing them all at once? I

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

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,

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