similar to: spam filtering with amavis

Displaying 20 results from an estimated 2000 matches similar to: "spam filtering with amavis"

2006 Mar 06
2
getting lock error
I'm getting this error for al my users does anybody know wat the problem is and how i can fix it pop3(bas at giskit.nl): Mar 06 09:10:37 Error: file_wait_lock() failed with index file /mailstore/vmail/giskit.nl/bas/.INBOX/.imap.index: No locks available Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax:
2006 Mar 02
2
ipv6 not working
I added an listner to dovecot bye adding the following lines to config but no result no connectivity with ipv6: imap_listen = [::] pop3_listen = [::] imap_listen = * pop3_listen = * Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel: +3130-6340430 Fax: +3130-6342433 Prive Tel: +3130-6372769 Mob:
2005 Aug 22
1
Hangup Faster
Hello - My single line extension users (connected via channel banks) need to be able to hang up faster. If they just flash the hook it doesn't disconnect right away. Any ideas on how to resolve this? Thanks, Dave -------------- next part -------------- An HTML attachment was scrubbed... URL:
2004 Sep 21
2
SIP termination in Brazil
Is there an up and running provider of SIP termination in Brazil? I know that there are some people building on a SIP termination solution. But who as it up and running ? Best regards, Han -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040921/f1043e19/attachment.htm
2007 Nov 08
3
Asterisk as a SIP to XMPP Jingle voice gateway
Hello, I'm looking for a SIP to XMPP Jingle voice gateway. I see that Asterisk has Jabber and Jingle support, but it looks like Asterisk acts as a Jabber client. Are there any Jabber server solutions, where Jabber users can call SIP users by using the SIP URI and vice versa? -- Eric Chamberlain, CISSP Chief Technical Officer Voxilla - http://voxilla.com/
2008 Jan 12
2
Asterisk RFC2833 to SIP INFO DTMF conversion erros.
Hi, I am using asterisk 1.4.17 which is connected to a SIP trunk supporting rfc2833 dtmf events. Asterisk stays in the media path. In sip.conf I have set dtmfmode=rfc2833 for the outbound sip proxy (SIP Trunk account) and for SIP clients I have set dtmfmode=info. So when I make a call to a cell number using the sip trunk and then press digits I can see the 2833 dtmf events coming to asterisk
2006 May 18
2
VoiceMail Groups
Has anyone seen good scripts or documentation on Voicemail groups? We are looking to have a system where you can send a voicemail to multiple mailboxes. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20060518/777a7b83/attachment.htm
2007 Dec 10
2
asterisk linkedin group
asterisk linkedin group I have created an asterisk linkedin group for anyone interested. http://www.linkedin.com/e/gis/45252/66270A773F53 Thank You, Steven BerkHolz - MCSA - MCSE - Manager of Information Systems HIROTEC AMERICA Board member of Connectech Greater Detroit www.connectech.org ________________________________ Please visit us on the web at www.hirotecamerica.com HIROTEC AMERICA Ph.
2006 Jan 09
8
Pri Gateway Hardware
Does anyone have any experience using a PRI gateway, I am looking for a way to have multiple asterisk boxes use one PRI, and send that over the network. I herd there are copper gateway devices (like a X100P card, only it registers with asterisk using sip, and it doesn't have to be physically connected to the box) Does anyone have any experience with a PRI gateway? And could tell me the cost
2007 Jun 26
2
number of samples in input_frame
Hi all Sorry if this is a dumb question: does the input_frame passed to speex_encode_int *have* to be frame_size samples long? e.g., If I only have 100 samples left to encode (which is less than the frame_size of 160 samples), can I just use an array that contains 100 samples, or do I need to create an array containing the 100 "real" samples plus 60 null samples at the end?
2007 Aug 28
2
Black screen when maximazing
After opening a few windows, some start to open completely black. So I have a frame, and a black rectangle. If I resize the window some display ok, some don't. if I maximize them again, they go black What is causing this? Does anyone know how to fix this? I am running compiz + Ubuntu 7.10 gutsy Xeon 2.4ghz + nvidia quadro nvs 280 Thanks IMPORTANT NOTICE: This message may
2006 Mar 31
1
Re: BRI cards, HFC, and bristuff - a generalquestionto clear up my understanding.
Thanks. I think our problem ca be similar. Have you tried to call from analog phone #1 to another analog phone #2? It works. But when you try to call vice versa from #2 to #1 it does not work. When you restart asterisk it works again - but only one direction. -David ________________________________ From: asterisk-users-bounces@lists.digium.com
2011 Jan 15
14
Top Posting
Bruce et al. I'm posting a new thread with the "Top Posting" subject so I won't draw complaints about "hijacking" the 4-port thread. Top Posting refers to the practice of sending a message with a reply at the top and including the entire thread below the reply. I prefer this. If I'm actively following a thread, the most-recent information appears at the top
2007 Apr 23
1
Make an iso image or a kickstart
Dears Can anyone guide me ?? I want to put my asterisk system on an iso image like trixbox ,or how to make a. how can I do that ,I am using centos 4.4 final Regards _____ ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written
2007 Nov 28
1
Digium TE120P versus Sangoma A101D-X
Hello List, We purchased a TE120P card from Digium and it works great. The only problem is that we are still experiencing echo on some calls. I've tried various echo cancellers (right now we are using OSLEC) and still no luck. My question has anyone gone from the TE120P to a Sangoma A101D-X Single Port T1/E1/J1 w/ echo cancellation? Have you noticed a difference? Also I called
2005 Mar 29
3
help w/ basics
Hello, I am new to Asterisk and new to this list. I got Asterisk setup and running using Asterisk@home, and purchased a PolyCom SoundPoint IP500 phone to test out. I cannot get the phone to talk to the Asterisk box. On bootup of the phone, it tells me that it cannot contact boot server. Why is that? It gets an IP fine, and I have also tried manually setting the IP of the phone and the Asterisk
2007 Apr 04
2
Remastering asterisk
Anyone have an idea to re master centos,in other worlds I have an asterisk on centos with all libraries and modules,how can I make it as an iso image ? Regards ********************************************* No employee or agent is authorized to conclude any binding agreement on behalf of Xplorium with another party by e-mail without express written confirmation by an officer of
2009 Oct 29
5
Dynamic DNS trunk
I have a trunk, and its host=dynamic dns. The problem is, when the IP changes the Sip show peers Still show the old IP of the DNS, I have to reload and save the configuration again so that asterisk recognize the new IP of the DNS. Any idea how to automate such a thing? Or how can I keep asterisk to deal with NAMES as NAMES, and IPs as IPs. Let me know. Thanks. -------------- next
2006 Mar 01
0
(no subject)
I have a problem Trying to connect with pop3 to my server with ipv4 it works great. But when I use ipv6 it doesn't want to work. I added the following lines: imap_listen = [2001:7b8:210::4] pop3_listen = [2001:7b8:210::4] but this doesn't work also Wat am doing wrong Met vriendelijk groet, Bas van Dikkenberg GISkit bv BFVD1-RIPE Tel:
2007 Jan 11
1
Installation on CYGWIN Failed (PR#9442)
Hi, I tried to install R-2.4.1 on cygwin system. "./configure" succeeded, but make failed. Below, I provide the output from the process: error message, and info from configure output, in that order. I appreciate that someone can guide me (technically in-sophisticated) through this process. Again, thanks for your help. Michael Niu (1). Output from make make[3]: