Hi, We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that asterisk stops running and I need to restart in order for us to receive calls. We receive our calls via a local sip provider over a dedicated T-1. We never have had an issue before until the upgrade to 1.4. It seems like asterisk gets hung up on a certain call and dumps. Anyone else noticing anything like this? Thanks, Jason Jason Adams Sumo Systems 4694 Cemetery Road Suite 310 Hilliard, OH 43026 Phone | 614.433.9906 ext: 102 Fax | 614.433.9931 E-mail | jadams@sumosystems.net <blocked::mailto:jadams@sumosystems.net> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20061228/4161b3fe/attachment.htm
I'm experiencing the same exact issue. I haven't found a solution for it however. Jeff On Thu, 2006-12-28 at 18:28 -0500, Jason Adams wrote:> Hi, > > We just upgraded to 1.4 and I'm noticing weird issues. I have noticed > that asterisk stops running and I need to restart in order for us to > receive calls. We receive our calls via a local sip provider over a > dedicated T-1. We never have had an issue before until the upgrade to > 1.4. It seems like asterisk gets hung up on a certain call and dumps. > Anyone else noticing anything like this? >>
On 12/28/06, Jason Adams <jadams@sumosystems.net> wrote:> > Hi, > > We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that > asterisk stops running and I need to restart in order for us to receive > calls. We receive our calls via a local sip provider over a dedicated T-1. > We never have had an issue before until the upgrade to 1.4. It seems like > asterisk gets hung up on a certain call and dumps. Anyone else noticing > anything like this?Yes, same thing here. This seems to be the only problem we have with 1.4. We are using only SIP connections. David
-----Original Message----- From: asterisk-users-bounces@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David Thomas Sent: Friday, December 29, 2006 8:18 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] 1.4 Random disconnects On 12/28/06, Jason Adams <jadams@sumosystems.net> wrote:> > Hi, > > We just upgraded to 1.4 and I'm noticing weird issues. I have noticed> that asterisk stops running and I need to restart in order for us to > receive calls. We receive our calls via a local sip provider over adedicated T-1.> We never have had an issue before until the upgrade to 1.4. It seems > like asterisk gets hung up on a certain call and dumps. Anyone else > noticing anything like this?> Yes, same thing here. This seems to be the only problem we have with1.4. We are using only SIP connections.>DavidI'm wondering if this is a bug? How do I go about getting all the proper info to submit a bug? Has anyone come up with a solution? - Jason
Antonio José dos Santos Brandão
2006-Dec-29 09:58 UTC
[asterisk-users] 1.4 Random disconnects
Testing 1.4 here i got the same issue. Running tcpdump figure out that packets are sent from the sip provider or ATA to asterisk1.4 machine but asterisk doesn't reply. At really, nothing apears at /var/log/asterisk/full and looks like the sockets aren't open. After a "stop now" and restart all works again. I can't reproduce the bug, it occurs time to time. Not a very load server. Just 2 channels was running. -- Antonio J. S. Brand?o On 12/28/06, Jason Adams <jadams@sumosystems.net> wrote:> > > > Hi, > > We just upgraded to 1.4 and I'm noticing weird issues. I have noticed that > asterisk stops running and I need to restart in order for us to receive > calls. We receive our calls via a local sip provider over a dedicated T-1. > We never have had an issue before until the upgrade to 1.4. It seems like > asterisk gets hung up on a certain call and dumps. Anyone else noticing > anything like this? > > Thanks, > Jason > > Jason Adams > Sumo Systems > 4694 Cemetery Road > Suite 310 > Hilliard, OH 43026 > Phone | 614.433.9906 ext: 102 > Fax | 614.433.9931 > E-mail | jadams@sumosystems.net > > _______________________________________________ > --Bandwidth and Colocation provided by Easynews.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > >
At 03:28 PM 12/28/2006, you wrote:>We just upgraded to 1.4 and I'm noticing weird issues. I have >noticed that asterisk stops running and I need to restart in order >for us to receive calls. We receive our calls via a local sip >provider over a dedicated T-1. We never have had an issue before >until the upgrade to 1.4. It seems like asterisk gets hung up on a >certain call and dumps. Anyone else noticing anything like this? >You're not alone. Ira
On Fri, Dec 29, 2006 at 09:12:24AM -0500, Jason Adams wrote:> -----Original Message----- > From: asterisk-users-bounces@lists.digium.com > [mailto:asterisk-users-bounces@lists.digium.com] On Behalf Of David > Thomas > Sent: Friday, December 29, 2006 8:18 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] 1.4 Random disconnects > > On 12/28/06, Jason Adams <jadams@sumosystems.net> wrote: > > > > Hi, > > > > We just upgraded to 1.4 and I'm noticing weird issues. I have > > noticed > > > that asterisk stops running and I need to restart in order for us to> > receive calls. We receive our calls via a local sip provider over a > dedicated T-1. > > We never have had an issue before until the upgrade to 1.4. It > > seems like asterisk gets hung up on a certain call and dumps. > > Anyone else noticing anything like this? > > > Yes, same thing here. This seems to be the only problem we have with > 1.4. We are using only SIP connections. > > >David > > I'm wondering if this is a bug? How do I go about getting all the > proper info to submit a bug? Has anyone come up with a solution? > > > Solution to what? > > > What exactly are the steps required to reproduce the problem? > > >All I saw in this thread is some random reports of disconnects. Pleaseenable 'full' in logger.conf and set (core) verbosity and debug to some decent value.> > > What channels do you have configured? > >Obviously I'm not the only one with this problem. We are using sip channels only; from our provider and internally between peers. I have set the logger.conf to 'full' and set the core verbosity and haven't noticed anything unusual so far. Although we haven't had a dropped call yet. I will continue to watch the logs and see what happens. As far as reproducing the problem it's hard to stay. I'm not sure at this point how to reproduce the issue. Sometimes it's on outbound calls (Long distance) other times it's on inbound calls. - Jason