similar to: Call not going through and failing because "never answered"

Sent from my BlackBerry wireless device from MTN -----Original Message----- From: adekunleadekoya at gmail.com Date: Fri, 28 Sep 2012 18:49:33 To: <CentOS at mailing.list> Reply-To: adekunleadekoya at gmail.com Subject: Apache : Listen on ports other than 80 Hi, I just install centos 6.3 on my oracle virtualbox. I tried 2 configure apache 2 listen on port say 1010 but after adding a

First Thursday in August is this Thursday – will the Bay Area social going to take place that day? From: llvmdev-bounces at cs.uiuc.edu [mailto:llvmdev-bounces at cs.uiuc.edu] On Behalf Of Marshall Clow Sent: Thursday, July 12, 2012 5:26 PM To: Nick Lewycky Cc: lldb-dev at cs.uiuc.edu; clang-dev Developers; llvmdev at cs.uiuc.edu Subject: Re: [LLVMdev] [cfe-dev] Bay Area LLVM Social - July On

Hi, I need detailed info on channel drivers. Pls help. Want 2 attempt new driver! Tks, Paul Sent through SMSmedia isms via MTN Uganda by 25677212716. [On: 2003/04/06 At: 19:34:35].

Dear R-help, I am trying to overlay a secondary y-axis over a boxplot. The problem is that the point of the line does not correspond to the centrepoint of the boxplot, which makes presentation and interpretation sloppy. Could someone point out where I am going wrong please? #DATA

Greetings!! We're getting some strange results using call files.. no matter the technology, DAHDI, SIP, etc., we get a "Call failed to go through, reason (3) Remote end Ringing" message when attempting to originate a call from a call file. Numbers changed to protect the innocent.... using call file.... //------------CALL FILE------------// Channel: DAHDI/g1/918005551212

On 31 July 2012 14:18, Kipping, David <dkipping at qualcomm.com> wrote: > First Thursday in August is this Thursday – will the Bay Area social > going to take place that day? > Tomorrow has a couple conflicts with sports events (a Giants game and an A's game). Thursday the 9th there's another Giants game but nothing else. Let's hold it then. Sorry for the

Arch = x86_64 OS = CentOS-6.4 (freepbx) Asterisk = 11.4.0 FreePBX = 2.11.0.2 Snom870 Handsets We are in the process of moving to an Asterisk based PBX. At the moment most things work as we wish. However, I have just notices that when I force a reload using 'amportal a reload' I see this loop start in 'asterisk -rvvvvvvvvvv': > Channel Local/s at tc-maint-000002a4;1

Next bay-area social is this week! Thursday August 9th at St. Stephens Green! Come socialize, try to recruit patch reviewers with beer, or just talk about any subject management hasn't asked you not to. Just like every llvm social! If you wouldn't mind RSVP'ing on http://llvmbayarea.appspot.com , that will help me get us the right sized space at the restaurant. Looking forward to

I am getting this error from call manager (4.0) and asterisk 1.2.4 I have canreinvite=yes on the call manager setup. I can call into the asterisk box from call manager. THat seems to work. When I am calling out of the box using a call file I see this entry from call manager... What might be the problem with my setup? THanks, JErry ---------------- <Date>03/06/2006

Hi, I'm having real difficulty in getting calls to go through with Asterisk. I've managed to check that my SIP connection is made to my provider. Below is an email I received from them: ----------------snip--------------------------------snip--------------------------------snip---------------- I am not certain of the reason for rejection but it has to do with the SDP, it does not

If the end user does not pick up the phone, is there a way to log to the CDR about the call file failing? /var/log/asterisk/messages does show a NOTICE message [2015-12-31 06:58:46] NOTICE[28059] pbx_spool.c: Call failed to go through, reason (3) Remote end Ringing [2015-12-31 06:58:46] NOTICE[28059] pbx_spool.c: Queued call to SIP/102 expired without completion after 0 attempts but I would

For quite some time I?ve been using FreeNAS to provide services as a NAS over ethernet and SAN over Fibre Channel to CentOS 7 servers each using their own export, not sharing the same one. It?s time for me to replace my hardware and I have a new R720XD that I?d like to use in the same capacity but configure CentOS 7 as a Fibre Channel target rather than use FreeNAS any further. I?m doing

Does asterisk support using an H.323 provider for outgoing calls? From everything I have found, it looks like it does. However, I have had no success in getting it to work. I would really appreciate if somebody could give me a hand. I am using the channel that comes with asterisk. I have also tried using the channel from inaccessnetoworks but have not had any more success. My provider

Hi all, I've got a program that creates a callfile and most if it working great. However, when a call fails, I'm trying to capture the reason, which I'm told should be in the ${REASON} channel variable. http://www.voip-info.org/wiki/view/Asterisk+auto-dial+out Here is an excerpt from the callfile: Channel: local/155555555 Callerid:Tests <155555555> MaxRetries: 0 RetryTime:

Hi everybody, I think i need a samba guru to solve this issue, because googling for months did not help and the problem is becoming pressing. I'm facing an annoying problem with samba. In detail, there is something wrong with the password handling. It happens from windows, mac or linux clients. Randomly (probably after $num days), the system asks to the user to change the password. After the

hi, I have to activate the thread again ... we keep having preformance problems on the DC, especially on Monday morning when the PCs are switched on and the users log in. some ldap-searches take a very long time, sometimes even over 15 seconds e.g: ldapsrv_SearchRequest: LDAP Query: Duration was 15.74s, SearchRequest by S-1-5-21-xxxxxxxxxx-xxxxxxxxxxxxxx-8585 from ipv4:192.168.35.117:49240

Hi, I am working on calling 2 registrations of same user on 2 different ip or ports. It works fine and both phones ring simultaneously. the problem is that there is one way audio, calling party can hear me but i can't hear calling party. here is the scenario.. SIP/XYZ at 192.168.0.20:5060 SIP/XYZ at 192.168.0.10:5678 i dial using following dial string Dial(SIP/XYZ at

On Mon, 2022-09-19 at 14:04 +0000, Heinz H?lzl via samba wrote: > hello, > I often have the problem of high load on the LDAP processes. > 1-3 LDAP processes cause 100% cpu load for approx. 10 sec. This > happens > regularly in intervals of 2-3 minutes. > How can I find out which client is causing this load and why? > How can I configure the logging to see who/what is causing

I'm using Slackware 11. I unistalled the package that provides libtiff 3.8..... and installed the most current 3.7.... for lib tiff. I downloaded asterisk 1.4 beta3 and the 1.4 beta2 addons and untared them. created a simlink: ln -s asterisk-1.4.0-beta3 asterisk I've compiled spandsp from as follows cd /usr/src wget

I performed the update yesterday that included installing kernel 2.6.9-42.0.2.EL and replacing ethereal with wireshark. First, the trivial... Wireshark would not run from the KDE menu without editing the menu to change the command from "wireshark" to "kdesu wireshark" because dumpcap is at usr/sbin/dumpcap instead of /usr/bin/dumpcap . Making the change causes wireshark to