similar to: Explain Cause of Error: manager.c: Accept returned -1: Too many open files

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 !

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

[root at abc asterisk]# lsof -u 50771 | wc -l 0 BTW, I'm using CentOS 6.5 > > Date: Thu, 13 Oct 2016 10:20:19 -0400 >> From: Dovid Bender <dovid at telecurve.com> >> To: Asterisk Users Mailing List - Non-Commercial Discussion >> <asterisk-users at lists.digium.com> >> Subject: Re: [asterisk-users] Openfile Issue >> Message-ID:

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

Hi list, Can anyone who has successfully ran asterisk on a home router please give me the modell number as well as how they did it ? Thanks. Dovid -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061207/92c7f946/attachment.htm

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

Hi, Since I upgraded from Asterisk 1.2.18 to 1.4.17 I've been experiencing high CPU utilization from the chan_sip module. I've notice the more sip peers I have loaded, the higher the CPU load goes when there are no active calls. I am currently using a Pentium 4 3.0Ghz with CentOS 4 Kernel 2.6.9-42.0.2.EL. I currently have 1558 sip peers loaded in Asterisk and the current CPU load is

Good morning, I have a Polycom 601 with two side cars. I created a list of contacts in XML and it shows up on the side cars exaclty how I set it up in the XXXXXXXXXXXX-directory.xml file (in the order that I wanted it etc.). However when I hit the directories button and then contact directory I see the list in alphabetical order based on the last name. I want it to show up in this list as well in

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

I am having major problems with the first run of Snom 360s that rolled out last month. I am working with the US vendor and they in turn are working with Snom but I wanted to see of anyone else was using these or having issues with them. Issues: Speakerphone/Hands Free volume spikes up and down during a call. You have to manually set the volume during every call. This makes it totally unusable.

Does anyone know if there is a way to get a DB or any other means to see if I can see if a call is coming from a cell phone or not. If I am able to see if it is cellular or not is there any way to see aprox. what area the phone is in (I know this wont be simple but would it work if I have an agreeement with the cell phone companies) ? This is for the US. Thanks. Dovid -------------- next part

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

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 List, I was thinking the other day that even with T.38 there are still some issues with faxing. I was thinking of a protocol that instead of just sending down the fax tones an ATA or "VOIP fax machine" would get the entire fax convert it into some sort of image and pass it down the line to the receiving end. I got the idea from RFC2833. Yes I know that fax machines send bit by bit and

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

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