similar to: weird problem

Dear Sir, Can someone help me please to find a free ebook talking about AGI scripting through asterisk? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090218/a59fc299/attachment.htm

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 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 have 320 GB SATA HDD. When I checked my phpsysinfo, it shows 95% HDD is filled. [root at vicidialnow ~]# df Filesystem 1K-blocks Used Available Use% Mounted on /dev/sda2 301924504 285002780 1337472 100% / /dev/sda1 101086 11062 84805 12% /boot tmpfs 1553832 0 1553832 0% /dev/shm [root at vicidialnow ~]# du 16896 . You have new mail in /var/spool/mail/root [root at vicidialnow ~]# df -i

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

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 --------------

hi i am thinking to use some xorcom hardware i only used digium hard... what do you think? it work ok? what about echo? it is plug and play only reload or i need to restar asterisk ? i understand the drivers are in the trunk of asterisk, is this rigth? have any one a conf file of a xorcom device? any other important comments are welcome. Thanks David -- (\__/) (='.'=)This is

hi there are a lot of virtualization solution out there and every one "is the best" and has some pro and some cons... wich one do you recomend? the idea to isolate diferents servers asterisk apache ... it is a good idea? sorry for the off topic but here is a place where are a lot of linux gurus Thanks -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your

hi sorry about the urgent but it is urgent i have problems configuring a connection between asterisk and avaya using H323. the module i am usign is ooh323 what do you need to help me? and any tip or hint? thanks!!! David -- (\__/) (='.'=)This is Bunny. Copy and paste bunny into your (")_(")signature to help him gain world domination. -------------- next part -------------- An

hi i have a 32 bits server asterisk 1.6 will work ok in it? is just for a demo server no more than 4 to 10 calls at the same time. and a tdm board. waht do you think? 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:

I have been using the inbound 800 services from vitelity. Slowly the usage has been rising and in the month of Jan the bill was for $650. I am currently on a 1.9 cents a minute plan. Am I paying too much ? Some suggestions my team generated to reduce the toll free incoming call bill were: 1. When people call in on the 800 number take the local number they are calling from and then call them back

What is the syntax to progress with a dial plan after hangup please? Michael