similar to: 1.6.0.9 sip.c: "Serious Network Trouble" ??

I'm trying to upgrade from 1.4.24.1 to 1.6.0.9 with a TE120P card. I can't make any connection over the T1. From CLI: ERROR[26017]: chan_dahdi.c:14300 process_dahdi: Unknown signalling method 'pri_cpe' at line 37. cat chan_dahdi.conf cat chan_dahdi.conf [trunkgroups] [channels] language=en ;internationalprefix = 00 ;nationalprefix = 0 context=from-pstn switchtype=national

OK, Thanks. I have a couple of questions -- the line numbers do not match exactly, so can you tell me a couple of lines before and after the line in question? Also, when will this be logged, if its only during sip debug, I need to change it to log when I can see it more readily. Thanks. On Wed, 29 Aug 2018 20:31:15 -0400, sean darcy wrote: > > On 08/29/2018 08:07 PM, John Covici wrote:

On 08/29/2018 09:42 AM, Carlos Rojas wrote: > Hi > > Probably somebody is trying to hack your system, you should block that > ip on your firewall. > > Regards > > On Wed, Aug 29, 2018 at 9:34 AM, sean darcy <seandarcy2 at gmail.com > <mailto:seandarcy2 at gmail.com>> wrote: > > I'm getting invites to very high ports every 30 seconds from a

I wonder if I could have that patch, maybe I could add it to my fail2ban regexp and if you have the correct regexp, I would apperciate that as well. Thanks. On Wed, 29 Aug 2018 19:18:29 -0400, Telium Support Group wrote: > > Depending on log trolling (Asterisk security log) misses a lot, and also depends on the SIP/PJSIP folks to not change message structure (which has already happened

On 08/29/2018 11:59 AM, Telium Support Group wrote: > Block a single IP is the wrong approach (whack-a-mole). You should consider a more comprehensive approach to securing your VoIP environment. Have a look at this wiki: > > https://www.voip-info.org/asterisk-security/ > > > > -----Original Message----- > From: asterisk-users [mailto:asterisk-users-bounces at

On Wed, Aug 29, 2018 at 6:20 PM Telium Support Group <support at telium.ca> wrote: > Depending on log trolling (Asterisk security log) misses a lot, and also > depends on the SIP/PJSIP folks to not change message structure (which has > already happened numerous time). If you are comfortable hacking > chan_sip.c you may prefer to get the same messages from the AMI. It still

I'm getting invites to very high ports every 30 seconds from a particular ip address: Retransmitting #10 (NAT) to 5.199.133.128:52734: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 0.0.0.0:52734;branch=z9hG4bK1207255353;received=5.199.133.128;rport=52734 From: <sip:37120116780191250 at 67.80.191.250>;tag=1872048972 To: <sip:3712011972592181418 at 67.80.191.250>;tag=as3a52e748

Still trying to upgrade to 1.6.0.9 for 1.4. It worked - it worked all day yesterday, but this morning: -- Executing [646xxxyyyy at longdistance:1] Answer("SIP/172-08276a60", "") in new stack .......... -- Executing [646xxxyyy at longdistance:6] Dial("SIP/172-08276a60", ""DAHDI/g2"/1646xxxyyyy") in new stack May 27 09:56:57]

Hi. So, I applied the patch, works, but I could not figure out a fail2ban regex which will hit that line, have you got one I can use? Thanks. On Thu, 30 Aug 2018 11:03:08 -0400, sean darcy wrote: > > On 08/29/2018 09:33 PM, John Covici wrote: > > OK, Thanks. I have a couple of questions -- the line numbers do not > > match exactly, so can you tell me a couple of lines before

We're getting a new server. I'm considering installing 64bit fedora rather than the 32bit we use now. Is 64 bit a problem with asterisk? Any issues we should expect? sean

I just upgraded to 1.6.2.rc-1 after running betas 2 and 3 with no problems and while everything seems fine i get these message at startup and than all is well. Should I be worried or do i need to let the team know about this? Also, is not finding "/dev/dahdi/transcode" a problem I should be worried about? And lastly conf2ael always segfaults when I try to run it. it did run once

The Asterisk Development Team has announced the release of Asterisk 1.6.0.9. Asterisk 1.6.0.9 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves a merge issue from trunk to 1.6.0.7 that caused memory to be freed that should not be. In trunk, pkt->data is an ast_str, but in 1.6.0, it is allocated in the same chunk of memory as the sip_pkt.

The Asterisk Development Team has announced the release of Asterisk 1.6.0.9. Asterisk 1.6.0.9 is available for immediate download at http://downloads.digium.com/pub/asterisk/ This release resolves a merge issue from trunk to 1.6.0.7 that caused memory to be freed that should not be. In trunk, pkt->data is an ast_str, but in 1.6.0, it is allocated in the same chunk of memory as the sip_pkt.

Guest receive mergeable rx buffer, it can merge scatter rx buffer into a big buffer and then copy to user space. In addition, it also use iovec to replace buf in struct virtio_vsock_pkt, keep tx and rx consistency. The only difference is now tx still uses a segment of continuous physical memory to implement. Signed-off-by: Yiwen Jiang <jiangyiwen at huawei.com> --- drivers/vhost/vsock.c

Guest receive mergeable rx buffer, it can merge scatter rx buffer into a big buffer and then copy to user space. In addition, it also use iovec to replace buf in struct virtio_vsock_pkt, keep tx and rx consistency. The only difference is now tx still uses a segment of continuous physical memory to implement. Signed-off-by: Yiwen Jiang <jiangyiwen at huawei.com> --- drivers/vhost/vsock.c

Hi Michael, On 2018/12/12 23:31, Michael S. Tsirkin wrote: > On Wed, Dec 12, 2018 at 05:31:39PM +0800, jiangyiwen wrote: >> Guest receive mergeable rx buffer, it can merge >> scatter rx buffer into a big buffer and then copy >> to user space. >> >> In addition, it also use iovec to replace buf in struct >> virtio_vsock_pkt, keep tx and rx consistency. The

Hi Michael, On 2018/12/12 23:31, Michael S. Tsirkin wrote: > On Wed, Dec 12, 2018 at 05:31:39PM +0800, jiangyiwen wrote: >> Guest receive mergeable rx buffer, it can merge >> scatter rx buffer into a big buffer and then copy >> to user space. >> >> In addition, it also use iovec to replace buf in struct >> virtio_vsock_pkt, keep tx and rx consistency. The

Guest receive mergeable rx buffer, it can merge scatter rx buffer into a big buffer and then copy to user space. Signed-off-by: Yiwen Jiang <jiangyiwen at huawei.com> --- include/linux/virtio_vsock.h | 9 ++++ net/vmw_vsock/virtio_transport.c | 75 +++++++++++++++++++++++++++++---- net/vmw_vsock/virtio_transport_common.c | 59 ++++++++++++++++++++++---- 3 files changed,

Guest receive mergeable rx buffer, it can merge scatter rx buffer into a big buffer and then copy to user space. Signed-off-by: Yiwen Jiang <jiangyiwen at huawei.com> --- include/linux/virtio_vsock.h | 9 ++++ net/vmw_vsock/virtio_transport.c | 75 +++++++++++++++++++++++++++++---- net/vmw_vsock/virtio_transport_common.c | 59 ++++++++++++++++++++++---- 3 files changed,

On Fri, May 10, 2019 at 02:58:36PM +0200, Stefano Garzarella wrote: > Since virtio-vsock was introduced, the buffers filled by the host > and pushed to the guest using the vring, are directly queued in > a per-socket list avoiding to copy it. > These buffers are preallocated by the guest with a fixed > size (4 KB). > > The maximum amount of memory used by each socket should