search for: ksyblast

...sk VM. Kseniya, could you try to reserve 100% of virtual CPUs for Asterisk VM (just to test this configuration)? I am sorry I don't have free hardware and time to test your Ubuntu 16, asterisk 13.1, kernel 4.4.0-104 configuration( On 18 Dec 2017 12:09 p.m., "Kseniya Blashchuk" <ksyblast at gmail.com <mailto:ksyblast at gmail.com>> wrote: Dmitry, are you using CentOS? What kernel version are you using? I will try with the same to see if it can be also a kernel-related issue. ??, 18 ???. 2017 ?. ? 11:35, Kseniya Blashchuk <ksyblast at gmail.com <mail...

Hi all! Does anybody know if it's possible to completely disable all asterisk call features (even the default ones like xfer)? Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20180111/8964a156/attachment.html>

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

...h pjsip, but more > chances of support on the issues tracker of Asterisk to have help. > > Regards. > > > -- > Ludovic Gasc (GMLudo) > Lead Developer Architect at ALLOcloud > https://be.linkedin.com/in/ludovicgasc > > 2017-03-13 14:41 GMT+01:00 Kseniya Blashchuk <ksyblast at gmail.com>: > >> Ok, thank you for the assistance! >> >> ??, 13 ???. 2017 ?. ? 16:38, Joshua Colp <jcolp at digium.com>: >> >>> On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote: >>> > Tested with latest Asterisk 14.3.0 on Ubuntu...

...all file is created), I see NativeFormats: (slin) WriteFormat: slin ReadFormat: slin WriteTranscode: No ReadTranscode: No Please note - I do not use any manager API Can anybody explain how the native format is chosen in these cases? On Wed, Nov 22, 2017, 9:33 PM Kseniya Blashchuk <ksyblast at gmail.com> wrote: > This thread is not the point how to create call files and not about > manager commands. > I am asking why naive codec is different for channel Local with cmd > Originate when comparing to the same action in a call file. Call files work > for me in this way....

Ok, thank you for the assistance! ??, 13 ???. 2017 ?. ? 16:38, Joshua Colp <jcolp at digium.com>: > On Mon, Mar 13, 2017, at 10:32 AM, Kseniya Blashchuk wrote: > > Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic > > and > > Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same behavior. > > Joshua, maybe you can advice what can

Hi all! I am running asterisk 13.1.0 on Ubuntu server 16.04. There are two IP addresses from the same subnet set on one interface, and bindaddr is set to the second on them in sip.conf and in iax.conf. Incoming connections work as expected. However, for outgoing connections it seems that asterisk tells the kernel to use the specific "bind" address only in case of UDP usage (both SIP and

Tested with latest Asterisk 14.3.0 on Ubuntu 16 kernel 4.4.0-66-generic and Centos 7 kernel 3.10.0-514.10.2.el7.x86_64. Absolutely the same behavior. Joshua, maybe you can advice what can be done further? ??, 13 ???. 2017 ?. ? 14:52, Kseniya Blashchuk <ksyblast at gmail.com>: > Ah ok, thank you for checking. > I'll maybe also try with the latest asterisk and/or other distro and see > if this behavior is reproduced. > > ??, 13 ???. 2017 ?. ? 14:46, Joshua Colp <jcolp at digium.com>: > > On Mon, Mar 13, 2017, at 08:43 AM,...

On Mon, Mar 13, 2017, at 08:43 AM, Kseniya Blashchuk wrote: > Mmh sorry I'm afraid I did not understand your last message. Yes the code > does that but only with UDP, for TCP the source address is 192.168.0.172 > though it's bound to 192.168.0.177: > IP 192.168.0.172.47596 > <mydestip>.5061 > If it was a system/kernel issue, then why is the behavior different for

On Wed, 22 Nov 2017, Dmitriy Serov wrote: > ?same => n,System(printf "Action: Originate\nActionID: 1\nChannel: Local/${number}@mycontext\nApplication: Confbridge\nData: ${confnum}\n" > > /var/spool/asterisk/outgoing/${number}-${confnum}) I get: Unknown keyword 'Action' at line 1 of /var/spool/asterisk/outgoing/... Unknown keyword 'ActionID' at line 2 of

Hello! I have a Polycom phone and sometimes I need to transfer calls without picking them up to local extensions. Polycom has a transfer button which sends SIP 302 packet to asterisk. Another local extension, receiving the call, sees not the original number but the local number that was transferring the call. I would like that the original number is shown. I am stuck at this point. I see messages

Same about me - need to re-enable membership all the time. Annoying (( ??, 12 ???. 2017 ?. ? 15:59, John Novack <jnovack at comcast.net>: > Not just gmail > Happening as well with Comcast.net > > My Comcast address is set to forward to another domain, as Comcast seems > to now block sending mail with a non Comcast "from" address. they turned > that on a couple

Well, correct me if I'm wrong, but I would say this conversation you have posted is a bit outdated, now fail2ban can be used with asterisk security log https://wiki.asterisk.org/wiki/display/AST/Asterisk+Security+Event+Logger. On Thu, Aug 17, 2017, 4:53 AM Telium Technical Support <support at telium.ca> wrote: > Keep in mind that the attacks you are seeing in the log are ONLY the

Thanks for trying, what asterisk version do you use? вт, 25 июн. 2019 г. в 17:50, Doug Lytle <support at drdos.info>: > We have Polycom phones (I'm using a VVX601, the destination is a VVX301). > We're also on Asterisk 13. > > I forwarded my call to the VVX301 and then dialed my phones DID. The > forwarded call showed my cell phone number, so I cannot reproduce.

Hi all! Asterisk 13.1.0 Ubuntu 16.04, all latest. Can anybody explain this to me - I run Originate command from dialplan: same => n,Originate(Local/${number}@mycontext,app,ConfBridge,${confnum}) and I get crazy sound distortion in the conference, and I see that transcoding takes place here: NativeFormats: (slin192) WriteFormat: slin ReadFormat: slin192 WriteTranscode: Yes (slin

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

Can it happen that the routes lead the traffic through another interface? Did you try a packet capture with tcpdump? Do the packets really leave the usb adapter? Can asymmetric routing be in effect? Maybe there were some static routes that disappeared when the adapter was unplugged... On Thu, Apr 20, 2017, 12:41 AM Antony Stone < Antony.Stone at asterisk.open.source.it> wrote: > On

If SIP goes to the same provider then yes. Still I would check a packet capture for better understanding. BTW, did you try iax debug? ??, 20 ???. 2017 ?. ? 19:46, Carlos Chavez <cursor at telecomab.mx>: > On 4/20/17 12:45 AM, Kseniya Blashchuk wrote: > > Can it happen that the routes lead the traffic through another interface? > Did you try a packet capture with tcpdump? Do the