similar to: WTB: Digium 1 or 4 ports E1 Cards

Hi, A have an asterisk connected to a legacy PBX trought an E1 and to the PSTN trought another E1. When the legacy user dial to the PSTN the call pass trought Asterisk. All works OK, the only problem is the delay on the Asterisk server when it receives the digits from the 1st E1 link. It will only make the call when the digit timeout expires. Is there a way to make something like

Hi, Does OCFS require NAS hardware to run or does normal PC hard disk work? Thanks, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://oss.oracle.com/pipermail/ocfs2-users/attachments/20081219/1bc9110b/attachment.html

Hi, I am havind deep trouble with a bunch of our newly arrived Optiplex 330 as it can't run Centos 5.2 property. The installation works fine, but when it boots up, it can't be connected to the network. I am getting error saying " link is not ready" when doing system-config-network. I check lspci and it can detect the network controller no problem. The light next to the cable

hi i need a link or something about asterisk load balancing i cant find any, i only found a paragraf in an email anything wiil be wolcome thanks! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, I am running a VOIP application on Centos 5.3. It is a 16 core box with 12 G of mem and all what it does is passing packets. What happens is that at around 2K channels running g711 ( 64k) codec, all eth0 is used up and no more traffic can go through. I have checked google and it talked about interrupt scheduler. does anyone know how to configure the kernel to allow it to use all CPSs for

If anyone is looking at upgrading their Digium hardware, or has an excess of any of these cards, please let me know. I am not interested in non-Digium hardware, but thanks for asking :) E100P E400P TE410P I only need one card, in that preference order. Would also prefer if the card is already in Australia. I'm in Sydney/Australia if that matters to you. Regards, Adam -- Adam Goryachev

Attention Management and Sales. Remember the computers you bought for either yourselves or for your customers. Remember all the software that came with them that was useless to you and you didn't know what to do with it. Maybe you bought systems that came with Windows 95 or 98 and you needed Windows NT or some other operating system. Until now that software just collected dust in your store

hi i am using asterisk 1.4.22 ubuntu 8.4 i have two Ethernet one for ssh and other one only for voip calls when i start a call using originate in the manager or the cli in the voip Ethernet i get something like 4Mbits/sec of traffic only 1 G711 call. if i start the call using a soft phone everything is normal. any idea? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny

I've got a Wildcard T100P along with a Zhone Zplex 10 24S/O which has been working fine for me now for a while. These have been pulled out of a working Asterisk installation (as they were no longer required) to use at home only to find that the fan noise is too loud. As such I'm looking to sell off this hardware and replace it with some combination of fanless hardware that will allow me to

I am looking for someone that could share their code for this function: Outgoing call -> macro that checks if line is (not human) or machine, fax, busy, subscriber problem and other fault tones -> if human connect to agent else hangup and write status to cdr. Need help with this! Regards / Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi I have an asterisk 1.6 running, and our provider have an openser on their end. When I get an incoming call I get this on my end [Jan 8 14:51:56] NOTICE[16680]: chan_sip.c:16869 handle_request_invite: Call from '' to extension '0840303395' rejected because extension not found. If I wait approx a minute and try again, the call will go trough. We don't use REGISTER or

hi wich soft phone do you recomend but i need this feature it must ask for user name and password when it start. i know xline and zoipper but they dont have that i can acomplish this whit twinkle but i need it for Windows :-( any ideas? thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next

Hi, I have a typical call center with queues and agents added via AddQueueMember. One of my requirement is to implement a forgot password function. If a caller does not remember the password, he calls up an unauthenticated line and the agent manually authenticates him. Then the caller should have a provision to reset his password. The requirement is that the agent should not know the new password

Yes it is. I'm remote at the moment so I can't send you the code but google for mobile remote receiver and you'll find what you are looking for. Lots of people do it so they don't have calls to cell phones picked up by voicemail. Cheers dean -----Original Message----- From: Pete Kay <petedao at gmail.com> Sent: Monday, March 31, 2008 2:27 PM To: Asterisk Users Mailing

hi i want to ask for your opinion what is better for a call center 100 current calls and other 200 current calls make the server step by step or use a auto install cd like asterisk now, druid elastix ....? and why? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part

hi wich predicitive dialer are you using and wich one do you recomend? a link to the project/product and a link to a how to will be VERY apreciated. Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi this is mi first email and just for say hello. David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20081202/375a91e5/attachment.htm

hi i need an open source callcenter manager system like queuemetrics but opensource any one know any? i prefer to search before start a new one thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi where i should load the module for the trasncoder wctc4XX (lspci shows TC400P) thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi i am reading about new codecs and new stuff to be added to asterisk. (and i say thanks to all the guys who are working to add all the new features). will be R2 added to the main core of asterisk like ISDN? Thanks David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part --------------