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...roduct: xorg Version: unspecified Hardware: x86-64 (AMD64) OS: Linux (All) Status: NEW Severity: critical Priority: medium Component: Driver/nouveau Assignee: nouveau at lists.freedesktop.org Reporter: mgresko8 at gmail.com QA Contact: xorg-team at lists.x.org Xorg hangs randomly with nouveau driver. It could be reproduced cometimes when playing video or starting libreoffice, but not limited to. If pressing Ctrl+Alt+Backspace, monitor goes to sleep immediately. Alt+Sysrq combinations are usually...

Hello, I would like to ask about current state of subscribecontext in pjsip. I found out some 6 years old discussion on that without any plans to implement it in the future. I have phones in different contexts. I suspect, when I use its context to subscribe, they will not see phones from the different contexts. Am I right? Marek

On Fri, Jun 5, 2020 at 6:02 AM Marek Greško <mgresko8 at gmail.com> wrote: > Hello, > > I would like to ask about current state of subscribecontext in pjsip. > I found out some 6 years old discussion on that without any plans to > implement it in the future. > > I have phones in different contexts. I suspect, when I use its co...

Hello, I found the problem and also the workaround. Clearly, since it was working with chan_sip it should not be dialplan problem, but sip stack problem. I have line=yes set up. After asterisk restart the old registration is not unregistered and new one is registered with different line value. Then incoming invites and qualify requests are sent to all the registrations and there the problem

Am 22.06.2020 um 17:01 schrieb Telium Technical Support: > I don't know if there was a prior email with more details, but.... > > Latency is as important as speed. Have you checked latency between your device and pop? What about QoS at your location, and does your ITSP support/respect QoS? That's a very good idea... Could you suggest me how can I check it? The Gateway is a

Hello, after migration from chan_sip to res_pjsip I get strange behavior when receiving call from the outside world. When call is received, it is replicated multiple times. Two of that calls get to the phone. So the phone is ringing on both lines. When having only Dial function in dialplan I am able to place call. But when creating some dialplan procedures containing VoiceMail I get phone ringing

Hello, try pinging your sip peer ip address following way: ping -n -M do -s 1300 -i 0.1 -c 100 ${ipaddress} Post several lines and the statistics. Were you also thinking about MTU problems? Not very probable, but one never knows. Marek 2020-06-22 17:18 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 22.06.2020 um 17:01 schrieb Telium Technical Support: >> I

Hello, this is a correct response: >From 62.156.246.57 (62.156.246.57) icmp_seq=1 Frag needed and DF set (mtu = 1492) So PMTU discovery is working. No problem here. You got correct message to lower the packet size from 62.156.246.57. This is probably the last hop before your site. Marek 2020-06-23 9:40 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 23.06.2020 09:28,

Hello, this could be ip address of the different interface on the same box. I think it works like expected. The only exception would be if the sip peer ignores the icmp packet unreachable. But I doubt this is the case. Anyway you get problems also when calling to LTE phone without using sip provider. Let first concentrate on these calls LTE to LAN. Are you sure you do not block incoming icmp

2020-06-23 15:02 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 23.06.2020 14:49, schrieb Marek Greško: > > Hi Marek, > >> this could be ip address of the different interface on the same box. I >> think it works like expected. The only exception would be if the sip >> peer ignores the icmp packet unreachable. But I doubt this is the > > Do you

It seems your problems lie in something other. Most probably it is not mtu problem. All my suspections are contradicted. If it is true you have inter vlan voice quality problems, it is definitely something different. Formerly I assumed you were trying only LTE vs LAN using internet. Marek 2020-06-23 15:50 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 23.06.2020 15:43,

I interchanged LAN and LTE in the sentence. Do you have some kind of NAT in fron of asterisk? Or is your asterisk having public IP? Could you share sip.conf (without passwords)? One LAN client, one LTE and general section. Marek 2020-06-23 16:29 GMT+02:00, Luca Bertoncello <lucabert at lucabert.de>: > Am 23.06.2020 16:22, schrieb Marek Greško: >> It seems your problems lie in

Hello, if you need clampmss then it is highly probable there is a PMTU discovery problem. The clampmss does not work for UDP. I probably counted the size incorrectly. So you are able to ping with size 1464 and not with 1466. How about trying same ping sizes from the internet towards your site? I mean trying to ping from sites with higher MTU than yours without lower MTU links in the path. You

Hello, there is no need to change canreinvite for provider configuration. Do not change MTU. Probably there will be another problem. I expect packet size 1466 would pass and higher will have the same result. It would be interesting to make the same test from the outside towards your asterisk with size 2 bytes larger the highest you are able to ping. Marek 2020-06-22 22:26 GMT+02:00, Luca

Would you mind repeating the test with canreinvite=no set for all you phones and mobile phones? What is your upload bitrate? Is it guaranteed? I would try also to test the PMTU: Try: ping -M do -s 2000 ${ip address of the sip server} You should receive icmp asking for lowering the packet size. The LTE phones could have lower MTU and thus overcome PMTU problem. Marek 2020-06-22 21:48