similar to: Asterisk chan_sip registration attempts

I am using CentOS 6, kernel 3.10 from elrepo.org kernels (3.10.102-1.el6.elrepo.x86_64). Asterisk version is 11.21.2 and Asterisk 13.X.X (I can't get it's version now). Is it possible that your network switches' interfaces which are connected to Hyper-V Server are 100% busy? It is possible that my installation works well because my Hyper-V server is not high-load server so it has

Thank you for a quick answer, Dmitry! We have tried the settings you suggested but nothing helped. The machine is running 4.4.0-104 kernel, 4 cores, Intel(R) Xeon(R) CPU E5-2620 v3 @ 2.40GHz, clocksource is hyperv_clocksource_tsc_page, timing module is res_timing_timerfd.so. We have also tried to set 50% Reserve - no luck :(. ??, 18 ???. 2017 ?. ? 10:49, Dmitriy Ermakov <demonihin at

Hi all! Does anybody have experience with asterisk on Hyper-V? My test setup with Ubuntu 16 and asterisk 13.1 (ubuntu repo) shows sound distortion. I have analyzed the RTP flow with wireshark and I see high skew and delta values when the traffic leaves the hypervisor, however everything is okay when a capture is taken from a VM itself. I have read that there can be timing problems with Hyper-V. I

Hello, For the moment I have the following settings in my sip.conf. I want to optimize them to archive the following things: - for the moment all my users will re-register too often. I want that only lagged users to re-register quickly. - check from time to time all users but no too often to see if is logged and can be called. Overall i want only lagged users to reregister and users with good

Hello again! I just got a SIP account and it seems - from a config on the net -, that I've configured it correctly. But I get no call to the outside. Registration was OK. I tried: channel originate sip/1/echo at iptel.org Application ... I see the channel active for a while, but no call gets established. In my config I have defined the section [iptel] for the outgoing call and I

Hello, I want to optimize my registrations and calls of peers to my asterisk with the following options in sip.conf: ---///--- qualify = yes qualify = 500 qualifyfreq=5 registerattempts = 0 registertimeout = 10 maxexpiry = 60 minexpiry = 20 defaultexpiry = 600 ---///--- Can someone more experienced with these settings to help me to optimize connections from peers with mobile phone that using

Hello, I have a local Asterisk server that keep loosing its registration to main Asterisk server. The local asterisk server is on the local subnet, it acts as a client with extension 808. Local server Sip.conf register => 808:password at as2.xxxxx.com registertimeout=20 registerattempts=10 Main Asterisk Server sip.conf [808] type=friend context=sip-phones call-limit=99

Hi all, I am using an asterisk as a client to connect to another asterisk server by registering with the register string. Registration is done without any hassel, but after sometime my asterisk loses the registration with the server and the server starts displaying the following msgs repeatedly: [Aug 9 06:37:59] NOTICE[8380]: chan_sip.c:8151 check_auth: Correct auth, but based on stale nonce

Hello. I have been beating my head over this problem for about 6 hours now. I have a SIP peer, who I register to (successfully), who should be directing all incoming calls at my [default] stanza in my extensions.conf: [ Context 'default' created by 'pbx_config' ] 's' => 1. Wait(1) [pbx_config] 2.

Hi Dmitriy and others and thanks for your help so far. The option "match_auth_username=yes" seems to have had no effect. From my reading, this option will try to match the username of the incoming SIP account to a section heading. If that is how it must work then i can see a big problem. I'm trying to present the receptionist with a nice display of which line the call came in on.

Can anyone tell me why my inbound calls keep getting rejected with 401? Here's the debug information: <--- SIP read from UDP:147.135.32.221:5060 ---> INVITE sip:6087294351 at 216.26.109.22:5060 SIP/2.0 Call-ID: 31007e-31 at 147.135.32.221 CSeq: 1 INVITE From: "Wi M"<sip:4144038968 at 147.135.32.221;user=phone>;tag=9bbc To: "Gregory Malsack"<sip:s at

I lost my internet connection today for a short time. During that time 1.2.12.1 stopped talking to my phones. Asterisk was still working as I got 2 voicemails. I have TDM analog cards for incoming calls. Anyway my cisco phones had X's (lost registration) and my uniden phones said "Registration error". Why would phones loose registration to asterisk when the internet connection

I am having trouble getting Google Chrome to accept a WebRTC call coming from Asterisk, even though Firefox can (now) accept the same call without issue. My setup is as follows: Server: CentOS 7 x86_64 (Elastix 4 RC) with IP: 10.1.0.4 192.168.5.146 asterisk-11.21.0 patched to work around https://issues.asterisk.org/jira/browse/ASTERISK-25659 openssl-1.0.1e-51.el7_2.2.x86_64 [root at elx4 ~]#

Hi mailing list and R-core. Could someone from R-core please help me to create account in bugzilla? I would like to submit issue related to gc() to wishlist. Related context is here - https://stat.ethz.ch/pipermail/r-devel/2017-July/074715.html -- Regards Dmitriy Selivanov [[alternative HTML version deleted]]

Hi Rui. Thanks for answer, I'm aware of drop = FALSE option. Unfortunately it doesn't resolve the issue - I'm expecting to get a vector, not a matrix . ??, 21 ????. 2018 ?. ? 20:54, Rui Barradas <ruipbarradas at sapo.pt>: > Hello, > > Use drop = FALSE. > > a[1, , drop = FALSE] > # col1 > #row1 1 > > > Hope this helps, > > Rui Barradas

Hi All. I'm writing storeRegToStackSlot and loadFromStackSlot function for my Target. This Target can store/load one byte (not all word) from FrameIndex. If I need to store 16 bit register I will must to split it to two instruction like this: BuildMI(MBB, MI, dl, get(Z80::LD8xmr)) .addFrameIndex(FrameIndex).addImm(0) .addReg(SrcReg, 0, Z80::subreg_lo); BuildMI(MBB, MI, dl,

05.10.2015 23:24, Joshua Colp ?????: > On 15-10-05 05:22 PM, Dmitriy Serov wrote: >> Hello. Do I understand correctly that the current implementation >> res_pjsip does not support ZRTP? >> http://lists.digium.com/pipermail/asterisk-dev/2013-December/064401.html > > ZRTP is not supported in Asterisk itself. > >> Nothing has changed since 2013? P.S. I greatly

Good day. Asterisk 13.7.2, res_pjsip. There is a problem of loss of registration of several devices. This happens not on all devices, but problem devices a lot. Below is the log of registration of a contact of one device. Is suspect two things: 1. delete a contact after the contact is added. But, like, it's a feature of code that may already be fixed. 2. deleting a contact much earlier

Stand contains two DC with Windows 2003 Std SP2 En: - root domain company.local - child domain (1) gonzo.company.local -- site 1: Windows 2003 (FSMO) -- site 2: Samba AD DC - child domain (2) depth.company.local I have CentOS 7 and Samba 4.3 in child domain (1) in dedicated active directory site (site 2 ). When I try logon from client (windows 7 workstation) I have error "Insufficient

I really forgot about .local but I don't think that the problem is related with this. -----Original Message----- From: samba [mailto:samba-bounces at lists.samba.org] On Behalf Of Rowland Penny Sent: Friday, October 02, 2015 3:25 PM To: samba at lists.samba.org Subject: Re: [Samba] Samba 4.3 Logon Error (Insufficient system resources exist to complete the requested service) On 02/10/15