similar to: High CPU load after upgrading to 1.4

I have a instance of Asterisk 1.2.14 that is being run from safe_asterisk. Asterisk is seg faulting and NOT generating a core dump. Why would that be? How can I make it dump core? Is there a setting in the safe_asterisk script that I am missing? Thanks, Doug. -------------- next part -------------- An HTML attachment was scrubbed... URL:

I have setup Asterisk on 2 Fedora Core 8 machines, and have made it to record all incoming calls. One of the box that have Asterisk 1.4.18 is properly merging calls and the other box that has Asterisk 1.4.15 is recording the calls but not merging them, I have made sure that SOX is installed on the box. Here is the Dialplan of both the machines : exten => 1234,1,Answer() exten =>

Hi *, does someone has a directory that works with the Cisco 7960 and astdb or mysql/ldap? Regards, Andreas _________________________________________________________________ Gaming galore at http://xtramsn.co.nz/gaming !

Hello, I have a few entries for sip peers in sip.conf with different name and username, like [TestSIPUser] type=peer host=dynamic username=testuser secret=1234 context=test_context [TestNewUser] type=peer host=dynamic username=newsipuser secret=3456 context=test_context When a call is made from any of these peers I want to get the username of the peer. for eg:- If a call is being made from

Hello list, When you have an asterisk box connected between the VoIP phones and an PSTN gateway what is the role of asterisk. Proxy server: stateful or stateless? From what i read in the: "Understanding the SIP, second edition" from Alan B. Johnston i think that asterisk is a stateful proxy server as well as registration server. Am I right? Can asterisk be configured to work as

I have a carrier who tells me he will be sending me traffic from a wide range of IP addresses. so I set up a realtime peer as follows: [peer] defaultip=xxx.xxx.xxx.xxx host=xxx.xxx.xxx.xxx deny=0.0.0.0/0.0.0.0 allow=xxx.xxx.xxx.0/255.255.255.0 insecure=port,invite Yes, he's really claiming to originate from any of the IP in the block When I leave the host blank, we reject calls with a

Hi List, While I know that "upping" ulimit will fix the issue I am trying to understand what will cause it. I have a few set ups that are almost exactly the same yet some machines used to give this error often and others don't. I also noticed the error a lot more on my boxes running 1.4.X. TIA. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hi, Is there any Royalty one needs to pay when using the inbuilt exisimg asterisk on hold music or when using any other mp3 from a music album. I think we need to pay for the later, but I am not sure if we need to pay for the inbuilt asterisk(freepbx) on hold music. -- Deepak --------------------------------- Yahoo! Answers - Get better answers from someone who

I have in my sip.conf the following [jon.moore] type=friend mailbox=8100,8150 In voicemail.conf, both mailboxes are defined. On my Aastra 480i phone, I only see the first mailbox listed. I've verified this, by changing mailbox= to reverse the order, and I then see 8150 when I go to Services > Voicemail on the phone. I also only get MWI events for whichever mailbox is listed

Hello 99% of all my users are calling from GSM phones, and my system basically just plays some sound files back. The PBX is connected to an ISDN-30 connection. Are there any modules for playing MP3 files, so I can use them with commands like Play() and Background()? And will it have any effect on the quality? Load issues should be a problem, the number of concurrent calls are pretty low.

Hi, What i want to do - is to give ability for answered call to hear regular dial tone and be able to enter phone number - that i would dial later. I tried to look at WaitExten and PlayTones, but they seem to not work together - WaitExten doesn't interrupt going on PlayTones. Is there any way how i could do that - so that it looks really natural? It would be silly to create long-long-long

Hi All; How can I test manager.conf? Can I telnet to the asterisk machine at the port 5038 and send and receive commands to test if the manager is working fine? How? Regards Bilal

Hello, I am trying to Bridge 2 E1 interfaces over a long distance link exactly the same way Redfone does. How can asterisk be configured to do that? Best regards Arinze Izukanne ___________________________________________________________ Want ideas for reducing your carbon footprint? Visit Yahoo! For Good http://uk.promotions.yahoo.com/forgood/environment.html -------------- next

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I have an installation which is currenly using 14 DID Trunk Lines. I need to be able to use Caller ID information and currently it is not available on these lines. Is there another way to access this information? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20040105/62559e22/attachment.htm

I've got Asterisk up and running nicely using a couple of different softphones. Audio quality is suffering a bit due to the hardware that I am working with. So I tried to use a Polycom hardphone but the politics is enough to give you a headache. Polycom seems to support SIP only if you buy it thought their vendors. So I'm looking at a Cisco phone. Has anyone successfully implemented

Hi, Has anyone ever had any problem with the TE220B card with it showing up as four ports instead of two. I RMA'd the first one with the retailer (Digium tech advice), but I just got another brand new card and it is coming up as four ports again. The card identifier is showing 0420 when I do lspci. Has this happened to anyone and if so is there a fix? Remi

https://fedoraproject.org/wiki/Ovirt_Node_Spin I noticed that virt-manager-tui is in rawhide finally, but the package name implies that it's not going to be in f16? virt-manager-tui.noarch 0.9.0-4.fc17 rawhide Cole, is this intentional or just smth that we need to follow up on? Also, I've noticed that ovirt-node needs refreshing... iirc apevec did tag/release of 2.0.1 from

> -----Urspr?ngliche Nachricht----- > Von: Mojo with Horan & Company, LLC [mailto:mojo at horanappraisals.com] > Gesendet: Dienstag, 25. M?rz 2008 23:23 > An: Asterisk Users Mailing List - Non-Commercial Discussion > Betreff: Re: [asterisk-users] Asterisk parking hold and > transferdigittimeout > > It seems that the dialplan comes into play. If your parking >

Hi, We've been running tinc for a while now but, have started hitting a bottleneck where the number of packets/sec able to be processed by our Tinc nodes is maxing out around 4,000 packets/sec. Right now, we are using the default cipher and digest settings (so, blowfish and sha1). I've been testing using aes-256-cbc for the cipher and seeing ~5% increases across the board. Each Tinc node