similar to: Connecting Multiple Asterisk Servers!

Hello *'s, I have a question regarding asterisk in asterisk is video confrencing is possible like meetme i am out of touch quite a long time so don't bother with my question if video confrencing is possible what kind of hardware required i already working on softphones setup with asterisk including video support . Thanks In Advance. Adnan Ahmed.

Hi All, I'm having a problem with cross subnet browsing and name resolution across an openvpn tunnel. i've found quite a few people who've had the same on mail lists but none of their fixes have worked. The spec of the setups at both ends of the tunnel are as follows: OS - CentOS 5.5 Samba Version 3.5.4 OpenVPN Version 2.0.9-1 Each server is configured in gateway mode with two NICS,

What about your openvpn config? The tun net must have an entry in your hosts allow. If you work with briding the remote network has to be the same subnet as the local!? Bridging is the best way to have a remote net integrated. I have one logging in form Berlin on my Samba-Domain. Daniel ----------------------------------------------- EDV Daniel M?ller Leitung EDV Tropenklinik

Hi All, I'm having a problem with cross subnet browsing and name resolution across an openvpn tunnel. i've found quite a few people who've had the same on mail lists but none of their fixes have worked. The spec of the setups at both ends of the tunnel are as follows: OS - CentOS 5.5 Samba Version 3.5.4 OpenVPN Version 2.0.9-1 Each server is configured in gateway mode with two NICS,

Hi to all, I have been using Wildcard TDM400P with four fxs modules and it was working fine now I have added another Wildcard TDM400P with four fxs modules . So there are total 8 ports for 8 hard phones. I have modified following configurations In /etc/zaptel.conf loadzone=us defaultzone=us fxoks=1-8 in /etc/asterisk/zapata.conf context=headoffice signalling = fxo_ks

hello all, friends i am new in asterisk. i had just finished the installation requirment of asterisk. i am using Centos 5.3 in which ill be installing asterisk now guys plz guide me my requirment for deploying asterisk is, i am having a client, (HR Consultancy) where 40 executives work and on each 40 desk, phone is there. i want confrencing facility,hold facility,extention nos,music. when ever

Commit-ID: dd94c677c8f33bfbcce12b3c597bf558d60a7338 Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=dd94c677c8f33bfbcce12b3c597bf558d60a7338 Author: Harald van Dijk <harald at gigawatt.nl> AuthorDate: Fri, 23 Aug 2013 20:30:28 +1000 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Fri, 25 Jan 2019 02:57:21 +0000 [klibc] [BUILTIN] Use PRIdMAX

Commit-ID: a436ea0721c8d7ff6d0d9e4d46d8ae4655e8a73a Gitweb: http://git.kernel.org/?p=libs/klibc/klibc.git;a=commit;h=a436ea0721c8d7ff6d0d9e4d46d8ae4655e8a73a Author: Harald van Dijk <harald at gigawatt.nl> AuthorDate: Fri, 23 Aug 2013 20:30:28 +1000 Committer: Ben Hutchings <ben at decadent.org.uk> CommitDate: Sat, 28 Mar 2020 21:42:54 +0000 [klibc] dash: [BUILTIN] Use

Hi all, My client softphone supports IM feature. Does any warmheated expert know if Asterisk can support IM also at server side? If so, is there any related documents or weblinks? -- Thanks & Best Regards! Steven Li -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060409/f131bc17/attachment.htm

Hi All, I am not so expert on rsync. I am using windows XP and Vista. I am able to take the backup using rsync.(It does not give the incremental) I need to take incremental also. After some googling (referring the link- http://www.jeffawaddell.com/2008_04_01_archive.html) I found that rsync + ZFS snapshot does the same job. It use rsync to make delta backups to the file server, and automatic ZFS

I've been working on patches to improve subregister liveness tracking on llvm and I wanted to inform the llvm community about the overal design/motivation for them. I will send the patches to llvm-commits later today. Greetings Matthias Braun Subregisters in llvm ==================== Some targets can access registers in different ways resulting in wider or narrower accesses. For

Hi, I'm having some problem with realtime: let's say I have a dialplan like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer A call from ip phone 300 comes in, and it's been answered. Then I "switch" the sip context to realtime, putting the exten in the db and using the variable name for this as in the file version. [globals] IPPHONES=_3XX [sip]

Hi all, Sorry to repost about this same issue once again, but I'm really at a complete loss as to how to solve the problem with ownership being changed after I've set it. I've set ownership as follows... ( set this by chmod -R 0770 HR, as well as chgroup -R HR-Group, I then signed into a Windows PC that was part of the domain, went to computer management as per the WIKI and chose

Hi guys, I'm battling to understand how the Samba4 user password expiry seems to tie in together and was hoping this could be clarified by someone for me please? Currently I have the following Samba4 domain policies in place... [root at headoffice ~]# samba-tool domain passwordsettings show Password informations for domain 'DC=abc-ho,DC=local' Password complexity: on Store plaintext

Aloha, Has anyone written an XML application for a Ciso 7960 phone running SIP? I can't find any examples anywhere! Anyone know of any resources for this? I have read it can render XML & can get input from the keypad & softkeys. Aloha, Matt

Hi, I have upgraded my PBX to Asterisk 1.2.5 , previously I was using Asterisk 1.0.9, and Every thing was working fine ,But now voice mail is not working. The error I am receiving in log files is like following, WARNING[2413] app_voicemail.c: No entry in voicemail config file for '12' I have searched for solution a lot can Any one of you let me know how can I solve this issue

FYI -------- Original Message -------- Subject: RE: AOL Mail Date: Wed, 25 Jun 2008 13:47:06 -0500 From: Dustin Davis <dustin@txls.com> To: "'Dale Schroeder'" <dale@BriannasSaladDressing.com> References: <419cf800ca3991a24f4b9829f3806aba@briannassaladdressing.com> <145401c8d604$ccfb1780$66f14680$@com>

What I didn't mention in r192119 is that mthi/lo clobbers the other sub-register only if the contents of hi and lo are produced by mult or other arithmetic instructions (div, madd, etc.) It doesn't have this side-effect if it is produced by another mthi/lo. So I don't think making mthi/lo clobber the other half would work. For example, this is an illegal sequence of instructions,

Currently it will always spill / restore the whole vreg but only spilling the parts that are actually live would be a nice addition in the future. Looking at r192119': if "mtlo" writes to $LO and sets $HI to an unpredictable value, then it should just have an additional (dead) def operand for $hi, shouldn't it? Greetings Matthias Am 10/8/13, 11:03 AM, schrieb Akira

Hi, I tried set up a global var for an extension, like this [globals] IPPHONES=_3XX [sip] exten=>${IPPHONES},1,Answer What I would like to do is to make a dialplan without fixed extension numbers to change the entire dialplan according to the customer requests: what exten number do you want for your IP Phones ? let me change a variable and we are set! It seems that this is not supported,