Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Noah Miller

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.


The best way to deal with a bug from an enduser standpoint:  Do what
you can to come up with a good description for the bug, when and how
it occurs (even if randomly).  If you can generate any info from the
logger that's good, if not, fine.  Then start a new bug report at
bugs.digium.com (you have to sign up for a user account, and then use
the "Report Issue" link), and include all this info.  The key is just
to include as much information as you possibly can about the bug.  A
bug that is this serious will probably be resolved within a very short
amount of time.

- Noah
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RE: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Jason Adams
On Fri, Dec 29, 2006 at 09:12:24AM -0500, Jason Adams wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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. Please
enable '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
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Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Julio Tejera




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.


I just installed AsteriskNOW (with * 1.4) it occurs to me on call
queues completelly randomly ..:o( on SIP channels.., any one have
tested it using IAX2 ?

jat

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Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Ira

At 08:29 AM 12/29/2006, you wrote:

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. Please
enable 'full' in logger.conf and set (core) verbosity and debug to some
decent value.

What channels do you have configured?



I asked this question on the 25th but no one even noticed, he asked 
the same question worded differently and he got a fair number of 
me-to's and this response. I finally had to go back to 1.2.14 so 
Asterisk would stay up more than 4 or 5 calls at a time. In my case 
it's All SIP inside with outside split 1/4 SIP, 1/4 IAX and the rest 
via a Digium TDM-04.  My symptoms with 1.4 were I start Asterisk, I 
run Asterisk -vr or whatever it was to get a console and about 4 or 5 
calls later the phone system would stop working and I'd be sitting at 
a Linux prompt. Typing Asterisk again would allow it to run for 
another few calls.  If I had any idea what to do to help you, I would 
have done it, but like many of the people here, we can make Asterisk 
do what we need but I've no idea what development might need to 
troubleshoot my problem and your blithely stating set verbosity to 
some decent value has absolutely no meaning to me.


I promise I want to help you solve this and I know how difficult it 
can be to fix something that might be caused by my unexpected 
configuration choices or accidently missing Linux component that 1.2 
didn't care about.  This is the first Linux box I've ever touched, 
all it does is run Asterisk. I'm thankful for MC and my previous 
familiarity with NC so I can navigate the box without too many 
issues, but if you want me to help you, and I assume this goes for 
others in my situation, I'll need a bit more direction than you just 
given.  Maybe this needs to be on the wikki so you can just point or 
so we can just search or maybe it is and you'll make me feel stupid, 
no matter, all that really matters to me is that the problem get 
fixed. I'm willing to put 1.4 up again and make calls till it crashes 
if I had a hope that I'd have a chance of helping you.  But I need 
your help to get there.


Ira 


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Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Ira

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 


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Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Antonio José dos Santos Brandão

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. Brandão


On 12/28/06, Jason Adams <[EMAIL PROTECTED]> 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 | [EMAIL PROTECTED]

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Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Tzafrir Cohen
On Fri, Dec 29, 2006 at 09:12:24AM -0500, Jason Adams wrote:
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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. Please
enable 'full' in logger.conf and set (core) verbosity and debug to some
decent value.

What channels do you have configured?

-- 
   Tzafrir Cohen   
icq#16849755jabber:[EMAIL PROTECTED]
+972-50-7952406   mailto:[EMAIL PROTECTED]   
http://www.xorcom.com  iax:[EMAIL PROTECTED]/tzafrir
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RE: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread Jason Adams
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] 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 <[EMAIL PROTECTED]> 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?

 - Jason
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Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread David Thomas

On 12/28/06, Jason Adams <[EMAIL PROTECTED]> 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
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Re: [asterisk-users] 1.4 Random disconnects

2006-12-29 Thread J. Iddings
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?
>  

> 

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[asterisk-users] 1.4 Random disconnects

2006-12-28 Thread Jason Adams
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 | [EMAIL PROTECTED] mailto:[EMAIL PROTECTED]>

 
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