similar to: Call duration problem or maybe calls not hanging up problem

Hello Asterisk-users, [2013-08-18 05:56:29] NOTICE[17089][C-000000a8] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=2762c06e [2013-08-18 05:56:34] NOTICE[17089][C-000000a9] chan_sip.c: Failed to authenticate device 390<sip:390 at xx.xx.xxx.xxx>;tag=7b909220 I keep getting messages like this where the IP, xx.xx.xxx.xxx, is my own IP. How do I figure

I was quite surprised to find a message in my in box from SellVOIP a day or two ago. It indicated I was running out of credit which was a surprise as I thought they'd gone under a large number of months back. So I ran upstairs, added their entry back to sip.conf, uncommented a couple of lines in extensions.conf and I'm again using sellvoip to make outgoing calls. The reason I was

I get this when my brother in law tries to call in from his box to mine. WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <s> or after changing the register line: WARNING[4855]: chan_sip.c:12675 check_auth: username mismatch, have <100>, digest has <199> I have done everything I can think of and still failure. Currently the

Can someone tell me what I am doing wrong? Why doesn't CDR(duration) or CDR(billsec) return the correct values? cdr.conf endbeforehexten=yes extensions.conf [macro-Dial] ; ${ARG1} - Dial String exten => s,1,Dial(${ARG1},,M(post-dial)) exten => h,1,NoOp(Call was hung up - ${CDR(duration)} seconds long, billed for ${CDR(billsec)} seconds) The log shows: -- Executing [h

Yesterday I upgraded my Zaptel to 1.2.26 or I think that was it, the latest 1.2 version at downloads.digium.com. I have a Digium 4 card populated with 4 red FXO cards using channels 1,2 and 4. Channel 3 is not used. It's been working fine for a few years. After upgrading to 1.2.26 calls stopped coming in on channel 1, Channel 2 still worked fine and I could get dialtone and make calls

Hello Asterisk, Would someone be kind enough as to add the issue: grp_lock error when compiling against pjproject and solution: delete the rogue install in /usr/local/include To the WIKI page about installing pjsip. I tried to update the WIKI but don't seem to have a way to do it. I know it's not supposed to happen and I know what I did wrong, but it's hard to imagine

Hello Asterisk, I've been running a CENTOS 5 box with Asterisk 14 and am trying to move to Asterisk 17.2 on a new Fedora Server 31 box. I built Asterisk from Source as I've always done and copied all the configuration files and other stuff from the old box. Everything comes up as expected and it all seems to work except I have one way audio. I'm still using SIP, not pjsip. As soon as

Hi I am using Trixbox 2.4 and PRI lines..on the CDR i see many calls that have duration of 0 seconds, but they are still shown as ANSWERED . how come its possible when duration is 0.00 ? Are the callers billed for such calls ? Rgds Sriram -------------- next part -------------- An HTML attachment was scrubbed... URL:

I show the file size as 19M on the FTP site. 95MB is the gunzipped size. I suspect something unzipped it as it downloaded, I've seen browsers do that. I also pulled the file personally, and found the sizes lined up: ira at ira-t430:~/Downloads [/dev/pts/1](64/0)$ ls -la samba-4.1.19.tar.gz -rw-rw-r--. 1 ira ira 19558250 Jun 23 20:16 samba-4.1.19.tar.gz ira at ira-t430:~/Downloads

What are people's thoughts on asterisk 1.2.26? Any show stopping bugs? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20080117/57d1002d/attachment.htm

Hello everybody, I am running 1.2.10-BRIstuffed-0.3.0-PRE-1s with florz-patches on Linux 2.6.16 with 4 HFC-Cards in TE-mode connected to 4 PtP ISDN-"Anlagenanschluesse". There are about 40 SIP-clients connected (mostly Sipura/Linksys PAP2, and some SNOMs and softclients) to this server. Everything works fine, except that my CDR reports some very long _concurrent_ calls from one sip

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

The Asterisk Development Team has announced the release of Asterisk 1.8.0-beta1. This release marks the beginning of the testing process for the eventual release of Asterisk 1.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ All interested users of Asterisk are encouraged to participate in the 1.8 testing process. Please report any

Hello, i have call time irregularites in my asterisk CDR. I a currently using a mysqly backent to save CDR records and use this to generate bills at the end of each month. However, my users are complaining that they gety charged for even uncompleted calls (i.e. calls they make whaich have already be setup but canclled). i have noticed that only 'AWNSERED' and 'Busy' show up in my

On 12/14/2017 10:18 PM, Leslie Zhai wrote: > Hi GCC and LLVM developers, > > I am learning Register Allocation algorithms and I am clear that: > > * Unlimited VirtReg (pseudo) -> limited or fixed or alias[1] PhysReg > (hard) > > * Memory (20 - 100 cycles) is expensive than Register (1 cycle), but > it has to spill code when PhysReg is unavailable > It might be

Hello Asterisk, Back in 2009 I built a small Intel Atom based computer running Centos 5 for my asterisk system. 5 phones, 2 people 1 POTs line and six or so SIP numbers. So basically no load. I'm feeling like it's time to build another machine. It's probably silly, but it's been six years and I can't upgrade the OS which is falling behind. I'd likely just put

I'm interested in crafting firewall rules that throttle connections that have lasted more than a certain amount of time. (Most such connections are P2P traffic, which should be given a lower priority than other connections and may constitute network abuse.) Alas, it doesn't appear that FreeBSD's IPFW can keep tabs on how long a connection has been established. Is there another firewall

[ add Ira ] On Wed, Jun 26, 2019 at 5:27 AM Christoph Hellwig <hch at lst.de> wrote: > > The code hasn't been used since it was added to the tree, and doesn't > appear to actually be usable. > > Signed-off-by: Christoph Hellwig <hch at lst.de> > Reviewed-by: Jason Gunthorpe <jgg at mellanox.com> > Acked-by: Michal Hocko <mhocko at suse.com>

The behaviour of my Asterisk appears to have changed suddenly without any apparent cause. The version is use is 1.4.27.1 When a call is not answered billsec is set to duration, and calls are charged. I can't see any change I could have made to cause this problem. Is it something already known in Asterisk? /voipfc

I am using RStudio Version1.0.143 on a Windows 7 machine. R version 3.4.0 I am trying to connect to a postgreSQL database with the following command, but I receive an error message that says my password is incorrect. Since I saved my password in a file, I think I remember it. I searched for a solution online, but cannot figure out what to do. If I have to change my password, please provide