similar to: [PATCH v2] xen-netback: add a pseudo pps rate limit

time_after_eq() only works if the delta is < MAX_ULONG/2. For a 32bit Dom0, if netfront sends packets at a very low rate, the time between subsequent calls to tx_credit_exceeded() may exceed MAX_ULONG/2 and the test for timer_after_eq() will be incorrect. Credit will not be replenished and the guest may become unable to send packets (e.g., if prior to the long gap, all credit was exhausted).

On Tue, 2013-10-15 at 10:44 +0800, jianhai luan wrote: > On 2013-10-14 19:19, Wei Liu wrote: > > On Sat, Oct 12, 2013 at 04:53:18PM +0800, jianhai luan wrote: > >> Hi Ian, > >> I meet the DomU''s network interface hung issue recently, and have > >> been working on the issue from that time. I find that DomU''s network > >> interface,

Hi, First a confession - The box in question is a 1.2.35 box, so this may be solved in a newer version as I know the JB code is all hugely changed, but... It may be worth checking into. Scenario: - IAX outbound call from Asterisk, which rings okay. - Remote end sends ANSWER, which we immediately ACK. - The ANSWER control packet gets put into the JB (that's how I read the code) - The remote

I have this in sip show history for a particular channel marked as dead (should be removed) in sip show channels: 1. TxReqRel INVITE / 102 INVITE 2. Rx SIP/2.0 / 102 INVITE 3. CancelDestroy 4. Rx SIP/2.0 / 102 INVITE 5. CancelDestroy 6. Unhold SIP/2.0 7. Rx SIP/2.0 / 102 INVITE 8. CancelDestroy 9. Unhold SIP/2.0 10. Rx

Hi all, i have the following setup PSTN -> routing server (asterisk 1.6.2.11) -> IAX -> callcenter asterisk 1.6.2.9 -> SIP -> agent Does work quit fine - then agent does have the abibility to transfer a call to a third party - the agent can initiate the transfer over a web interface - it does generate a asterisk manager atxfer request... So agent does initiate transfer - call

Netback and netfront only use one event channel to do tx / rx notification. This may cause unnecessary wake-up of process routines. This patch adds a new feature called feautre-split-event-channel to netback, enabling it to handle Tx and Rx event separately. Netback will use tx_irq to notify guest for tx completion, rx_irq for rx notification. If frontend doesn''t support this feature,

We''ve been hitting the following issue on a variety of hosts and recent Xen/dom0 version combinations. Here''s an excerpt from our latest: Xen: 4.1.4 (xenbits @ 23432) Dom0: 3.7.1-x86_64 BUG: unable to handle kernel NULL pointer dereference at 000000000000001c IP: [<ffffffff8141a301>] evtchn_from_irq+0x11/0x40 PGD 0 Oops: 0000 [#1] SMP Modules linked in: ebt_comment

Hi all, I am trying to connect to a softphone application using an Iax channel on Asterisk 1.4.30. I can do outbound calls, from softphone to asterisk, but not inbound from asterisk to softphone. I get the following Debug: ---------------------------------------------------------------------- ---------------------------------------------------------------------- Tx-Frame Retry[000] -- OSeqno:

Tune xen_netbk_count_requests to not touch working array beyond limit, so that we can make working array size constant. Signed-off-by: Wei Liu <wei.liu2@citrix.com> --- drivers/net/xen-netback/netback.c | 26 +++++++++++++++++++++----- 1 file changed, 21 insertions(+), 5 deletions(-) diff --git a/drivers/net/xen-netback/netback.c b/drivers/net/xen-netback/netback.c index

A buggy or malicious frontend should not be able to confuse netback. If we spot anything which is not as it should be then shutdown the device and don''t try to continue with the ring in a potentially hostile state. Well behaved and non-hostile frontends will not be penalised. As well as making the existing checks for such errors fatal also add a new check that ensures that there

hallo, i am really confused how nativ briging is working with asterisk, i use a asterisk server as central server and register another asterisk and an iaxcomm client to the server, all three have public ips on the internet. somtimes, when i call from iaxcomm to my asterisk, the calls go peer to peer (i can see it with tcpdump) but sometimes the get routed through the central asterisk server

I'm taking inbound calls on an * server, then transferring them to a second * server where they join a MeetMe conference. If I have 'notransfer=yes' set on the first * server it works fine, but if I allow the transfer (call then shifts to be between the DID provider and the second server), the call is dropped 3-5 minutes later. There is no firewall on my end, and the two

Hi all, this is the long awaited patch series to add xen device model support in qemu; the main author is Anthony Perard. Developing this series we tried to come up with the cleanest possible solution from the qemu point of view, limiting the amount of changes to common code as much as possible. The end result still requires a couple of hooks in piix_pci but overall the impact should be very

The Asterisk Development Team has announced the release of Asterisk 1.6.1.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.18 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

The Asterisk Development Team has announced the release of Asterisk 1.6.1.18. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.1.18 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by

The Asterisk Development Team has announced the release of Asterisk 1.6.2.6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.6 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

The Asterisk Development Team has announced the release of Asterisk 1.6.2.6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk/ The release of Asterisk 1.6.2.6 resolves several issues reported by the community, and would have not been possible without your participation. Thank you! The following are a few of the issues resolved by community

hi list, after connecting 3 asterisk servers via IAX in a line (+ 1 client at each end), i noticed that call path optimization happens only one time, i.e. only one node/leg in the path can be reduced. Does anybody know if this is the intended behaviour or if it's a bug? Can anyone confirm my observations? It seems that the first node, that sends the TXREQ Message, is "optimized".

Another checksum offload problem was reported on xen-users, when using a domU as a firewall: http://lists.xensource.com/archives/html/xen-users/2006-04/msg01150.html It also fails without VLANs. The path from dom0->domU with ip_summed==CHECKSUM_HW/proto_csum_blank==1 is broken. - skb_checksum_setup() assumes that a checksum will definitely be calculated in dev_queue_xmit(), but the

I am having some problems getting TelIax service to work with *. Outbound calls work just fine. When I try an inbound call the phone rings and there is no audio. Upon further investigation "iax2 show channels" indicates that the codec is "unknown" The provider confirmed that they are set for ulaw and so am I. Does anyone have an idea what could be causing the codecs to