similar to: Sep 28 17:52:28 WARNING[163850]: chan_sip.c:673 retrans_pkt: Maximum retries exceeded on call 2bda648025dbf8c52fd293515d98d2c2@216.252.176.45 for seqno 102 (Non-critical Request)

Hello List, I'm trying to setup MGCP channel with a Centile Media Hub box. My Centile box has 4 ports and I got no dial tone. Can somebody help with this isuue? This is my mgcp.conf and extensions.conf Thanks Daniel. ; MGCP Configuration for Asterisk ; [general] port = 2427 bindaddr = 192.168.11.20 disallow=all allow=g729 allow=alaw allow=ulaw [192.168.11.200] context=MGCP

I've seen other request info on this config before. I've got the system up and running, Zap/Modem has been working for several days, no errors/warnings. Simply added info for broadvoice and I can dial into my system via the remote, but dial out from my box renders the following: I get one ring on the remote line, then on the console chan_sip.c:675 retans_pkt: Maximum retries exceeded

Hey, I'm getting an odd message in my logs, and have'nt been able to find much information on it: Jan 24 00:22:39 WARNING[-1137431632]: chan_sip.c:486 retrans_pkt: Maximum retries exceeded on call 6010532c6fedf9be383872e07e4be70c@192.168.1.2 for seqno 102 (Request) I'm running asterisk with a Cisco 7960G If anyone know's why i'd get this.....Any help would be appreciated!

Hello, This is the scenario i want to setup: Cisco ATA 186 -----------> SER -------------> Asterisk I want the Cisco ATA to register to Asterisk through SER. when the Cisco ATA place a call, SER querry a data base (MySQL or else), and if there is a valid Account for the ATA, the call go to Asterisk. Did someone know how to set SER to work like this with Asterisk? which version of SER

We have an interesting issue: One of our providers has two softswitches. Calls coming from the first one are handled fine by asterisk, calls coming from the second one and going through the first one are euhm... dropped half a second into the RTP stream. I have opened a ticket at Digium for it: http://bugs.digium.com/view.php?id=10449 The output of "sip debug" is funny from line

Hello all, Is X-lite soft phone support G.729 ? I actually use it but there is no G.729 support. Anyone know where to have it? Regards. Daniel. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050209/8cdbeeec/attachment.htm

I have a situation here where a user has an AAstra 480i phone, which function corectly. The phone is behing a nat-router (a linksys wrv200 for it's VPN point to point facility). The phone is plugued in a port wich has qos enabled. And when the user places a call, sometimes (not always), we get this in the console : [May 8 13:41:55] WARNING[5804]: chan_sip.c:1948 retrans_pkt: Maximum

I've being using the same non-nat config for a 7960 for about a year with no issues. I have just upgraded cvs head from the 16th of July to todays. About a minute after I make a call I get the message chan_sip.c:1132 retrans_pkt: Maximum retries exceeded on transmission 000b46a0-8661007f-5c457e72-2b0dba2c@192.168.123.20 for seqno 102 (Critical Response) I don't get this for other

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Has anyone who's gotten this message managed to figure it out and fix it? I've been scouring the mailing list for clues but I'm still no closer. I have 2 asterisk servers, old and new. I'm trying to switch to the new server. I am using a 2.4 kernel on the new and a 2.6 on the old. I am running 0.9.0 on the old and my sip phones work fine, on the new Ihave tried 0.9.0,

Hi, I just upgraded my entire environment from Tinc 1.0.24 (or prior) to the most recent version from git (1.1pre10+), using ED25519. Sometimes, especially after some failed authentication attempts or timed out authentications, I get the following log messages: 2014-08-11 21:51:10 tinc.NNN[3376]: Connection with XXX (1.2.3.4 port 1) activated 2014-08-11 21:51:10 tinc.NNN[3376]:

This smells like https://github.com/gsliepen/tinc/pull/104 - are you sure you're using latest HEAD? How often does this occur? Does it correlate with other events such as nodes joining or leaving? Does it occur more often if you reduce the value of KeyExpire? On 22 December 2015 at 16:10, Florent B <florent at coppint.com> wrote: > Hi, > > With latest Tinc 1.1 git code, I have

On Mon, Feb 22, 2016 at 11:46:28AM +0100, Florent B wrote: > I still have this "bug"... any idea of what's going wrong ? There are some situations where keys are being renegotiated that can cause two peers to be very briefly out of sync. If that happens when there is VPN traffic, it could cause that message to pop up. If it only happens once in a while and you don't notice

If the fence context has notifies enabled, each of the fences' FENCE_FLAG_SIGNALED_BIT will be updated from the interrupt. We can rely on this status for reporting the current fence_is_signaled() and so avoid an expensive uncached read. Signed-off-by: Chris Wilson <chris at chris-wilson.co.uk> Cc: Ben Skeggs <bskeggs at redhat.com> Cc: dri-devel at lists.freedesktop.org Cc:

From: Chia-I Wu <olvaffe at gmail.com> [ Upstream commit efe2bf965522bf0796d413b47a2abbf81d471d6f ] This is motivated by having meaningful ftrace events, but it also fixes use cases where dma_fence_is_later is called, such as in sync_file_merge. In other drivers, fence creation and cmdbuf submission normally happen atomically, mutex_lock(); fence = dma_fence_create(...,

From: Chia-I Wu <olvaffe at gmail.com> [ Upstream commit efe2bf965522bf0796d413b47a2abbf81d471d6f ] This is motivated by having meaningful ftrace events, but it also fixes use cases where dma_fence_is_later is called, such as in sync_file_merge. In other drivers, fence creation and cmdbuf submission normally happen atomically, mutex_lock(); fence = dma_fence_create(...,

Hello to all, I have a strange behavior of my asterisk box. I'm running asterisk with asterisk-oh323 channel driver and everything works very well. But after few hours, my asterisk stop running and I have to restart it by typing "asterisk -vvvc". Most of the time I connect to my asterisk with a remote host so I don't know exactly which error causes my box to stop, but I found on

Dear developers, the CRAN incoming queue will be closed from Sep 1 to Sep 9. Hence package submissions are only possible before and after that period. Best, Uwe Ligges (for the CRAN team)

Hey, I just googled my error and many things came up. I followed the leads and read the ?read.delim page; I tried changing header = TRUE, and row.names = TRUE-- but I've still been having trouble fixing it, so I would greatly appreciate any help you can provide. Here is my code: rm(list=ls()) source("../../functions.R") uncurated <-

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