Re: [asterisk-users] Asterisk stops responding
Am 23.01.2011 18:38, schrieb Carlos Chavez: On Sat, 22 Jan 2011 19:47:43 -0500, Mark Deneen wrote On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez wrote: On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote On 22 Jan 2011, at 18:02, Carlos Chavez wrote: Cannot allocate memory Have you tried looking at memory? S The server has 8gb of ram and 8gb of swap. Free indicates that there are at least two free gb of memory and swap remains at 0 use. Just asking the obvious, but, x86-64? How big is the asterisk process? 4650 root 15 0 3153M 132M 10420 S 0.0 1.6 3:24.40 /usr/sbin/asterisk -f -vvvg -c 2.6.18-194.32.1.el5 #1 SMP Wed Jan 5 17:52:25 EST 2011 x86_64 x86_64 x86_64 GNU/Linux Plase show us the header of the used ps-command. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops responding
On Sat, 22 Jan 2011 19:47:43 -0500, Mark Deneen wrote > On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez > wrote: > > On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote > >> On 22 Jan 2011, at 18:02, Carlos Chavez wrote: > >> > Cannot allocate memory > >> > >> Have you tried looking at memory? > >> > >> S > >> > > The server has 8gb of ram and 8gb of swap. Free indicates that there > > are > > at least two free gb of memory and swap remains at 0 use. > > Just asking the obvious, but, x86-64? How big is the asterisk process? > 4650 root 15 0 3153M 132M 10420 S 0.0 1.6 3:24.40 /usr/sbin/asterisk -f -vvvg -c 2.6.18-194.32.1.el5 #1 SMP Wed Jan 5 17:52:25 EST 2011 x86_64 x86_64 x86_64 GNU/Linux -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops responding
On Sat, Jan 22, 2011 at 6:53 PM, Carlos Chavez wrote: > On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote >> On 22 Jan 2011, at 18:02, Carlos Chavez wrote: >> > Cannot allocate memory >> >> Have you tried looking at memory? >> >> S >> > The server has 8gb of ram and 8gb of swap. Free indicates that there are > at least two free gb of memory and swap remains at 0 use. Just asking the obvious, but, x86-64? How big is the asterisk process? -M -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops responding
On Sat, 22 Jan 2011 20:51:54 +, Steve Howes wrote > On 22 Jan 2011, at 18:02, Carlos Chavez wrote: > > Cannot allocate memory > > Have you tried looking at memory? > > S > The server has 8gb of ram and 8gb of swap. Free indicates that there are at least two free gb of memory and swap remains at 0 use. -- Carlos Chavez Director de Tecnología Telecomunicaciones Abiertas de México S.A. de C.V. Tel: +52-55-91169161 Ext 2001 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops responding
On 22 Jan 2011, at 18:02, Carlos Chavez wrote: > Cannot allocate memory Have you tried looking at memory? S -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops responding
>-Original Message- >From: asterisk-users-boun...@lists.digium.com >[mailto:asterisk-users-boun...@lists.digium.com] On >Behalf Of Carlos Chavez >Sent: Saturday, January 15, 2011 2:02 AM >To: Asterisk >Subject: [asterisk-users] Asterisk stops responding > > I am having a problem with an Asterisk 1.6.2.15 server that runs a small >call center with Queuemetrics. In the past month we've had this problem 3 >times. > > The problem is that Asterisk simply stops responding. No calls in or out >and you cannot even get to the CLI. The process seems to be running but there >is simple no activity. All I see in the log files is: It might be this - I had something similar in behavior though I don't know if I ever got the same error message: https://issues.asterisk.org/view.php?id=18031 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Asterisk stops responding
On Jan 15, 2011, at 2:01 AM, Carlos Chavez wrote: > The problem is that Asterisk simply stops responding. No calls in or out > and you cannot even get to the CLI. The process seems to be running but there > is simple no activity. All I see in the log files is: > > [Jan 14 16:30:46] WARNING[20747] pbx.c: Failed to create new channel thread > [Jan 14 16:30:46] WARNING[20747] chan_sip.c: Failed to start PBX :( > [Jan 14 16:30:47] WARNING[20745] pbx.c: Failed to create new channel thread > [Jan 14 16:30:47] WARNING[20745] chan_dahdi.c: Unable to start PBX on > DAHDI/29-1 > > After restarting Asterisk everything is back to normal. The time between > the first failure and the second was almost a month, between the second and > third a few days. Carlos, What is in the logs immediately preceding the warning you have posted here? Scan up a number of lines (more if you have a very verbose installation, like FreePBX) and see if anything pops out at you. Basically, you want to figure out what was happening on the server at the time of the crash? Incoming fax? Hangup of a Dahdi channel? Incoming Dahdi call, etc. That will likely point you in the right direction. Tom -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [asterisk-users] Asterisk stops responding to SIP/ZAP
The problem has pretty much been there from the beginning. I may
re-arrange cards and see if it happens on one particular channel or if
the problem moves with cards.
Thanks for the response.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Yuan LIU
Sent: Saturday, April 21, 2007 12:52 AM
To: asterisk-users@lists.digium.com
Subject: RE: [asterisk-users] Asterisk stops responding to SIP/ZAP
>From: "Ken Williams" <[EMAIL PROTECTED]>
>Date: Fri, 20 Apr 2007 07:27:05 -0600
>
>About once a week or so my Asterisk box stops responding to all phones.
>I can pull up the console, do whatever I want at the CLI but the only
>way to get things working again is to restart Asterisk altogether.
>
>I finally cranked verbose & debugging way up (and watched my log files
>go from 1mb/day to 100mb/day), but below I believe contains my problem.
>The next line is 1.5 minutes later where I restart Asterisk.
As a general troubleshooting procedure, you want to ask yourself if you
have made any changes before it stopped working. If not, and especially
if you can restart and get it working again, I'd suspect some hardware
failure.
(Assuming the problem is reproduceable - I had times when TDM card
stopped working with no trace of error.) Try installing on another box.
Yuan Liu
>SIP/701 is a Grandstream GXP-2000 phone (we have about 30 of them in
>place here). Zap/3-1 is a Digium TDM400.
>
>I can't quite figure out where my problem is, is it the initial
>exception, is it not getting hung up completely, does it have to do
>with the call limit on the SIP channel, perhaps 'no provider found'
>statements?
>
>Any help would be appreciated, I have a relatively simple dial-plan, I
>can send over relevant bits of it if necessary.
>
>Thanks,
>Ken
>
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Exception on 12, channel 3
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Got event On hook(1) on
>channel 3 (index 0) [Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled
>echo cancellation on channel 3 [Apr 19 13:51:13] DEBUG[27722]
>channel.c: Didn't get a frame from
>channel: Zap/3-1
>[Apr 19 13:51:13] DEBUG[27722] channel.c: Bridge stops bridging
>channels SIP/701-08ee6120 and Zap/3-1 [Apr 19 13:51:13] DEBUG[27722]
>channel.c: Hanging up channel 'Zap/3-1'
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: zt_hangup(Zap/3-1) [Apr 19
>13:51:13] DEBUG[27722] chan_zap.c: Hangup: channel: 3 index = 0, normal
>= 12, callwait = -1, thirdcall = -1 [Apr 19 13:51:13] DEBUG[27722]
>chan_zap.c: disabled echo cancellation on channel 3 [Apr 19 13:51:13]
>DEBUG[27722] chan_zap.c: Set option TDD MODE, value:
>OFF(0) on Zap/3-1
>[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Updated conferencing on 3,
>with 0 conference users [Apr 19 13:51:13] VERBOSE[27722] logger.c: --
>Hungup 'Zap/3-1'
>[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
>change to be queued on device/channel Zap/3-1 [Apr 19 13:51:13]
>DEBUG[27722] pbx.c: Spawn extension
>(from-internal,201,2) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
>(from-internal, 201, 2) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] DEBUG[27722] pbx.c: Launching 'Hangup'
>[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Executing
>[EMAIL PROTECTED]:1] Hangup("SIP/701-08ee6120", "") in new stack [Apr 19
>13:51:13] DEBUG[27722] pbx.c: Spawn extension
>(from-internal,h,1) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
>(from-internal, h, 1) exited non-zero on 'SIP/701-08ee6120'
>[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel
>'SIP/701-08ee6120'
>[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Hangup call
>SIP/701-08ee6120, SIP callid [EMAIL PROTECTED]) [Apr 19
>13:51:13] DEBUG[27722] chan_sip.c: Updating call counter for incoming
>call [Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Call from peer '701'
>removed from call limit 6 [Apr 19 13:51:13] DEBUG[27722] devicestate.c:
>Notification of state change to be queued on device/channel SIP/701
>[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
>change to be queued on device/channel SIP/701-08ee6120 [Apr 19
>13:51:13] DEBUG[20432] devicestate.c: No provider found, checking
>channel drivers for Zap - 3 [Apr 19 13:51:13] DEBUG[20432]
>devicestate.c: Changing state for Zap/3 - state 0 (Unknown) [Apr 19
>13:51:13] DEBUG[20432] devicestate.c: No provider found, checking
>channel drivers for SIP - 701 [Apr 19 13:51:13] DEBUG[20432]
>chan_sip.c: Checking device state for peer 701 [Apr 19 13:51:1
RE: [asterisk-users] Asterisk stops responding to SIP/ZAP
From: "Ken Williams" <[EMAIL PROTECTED]>
Date: Fri, 20 Apr 2007 07:27:05 -0600
About once a week or so my Asterisk box stops responding to all phones.
I can pull up the console, do whatever I want at the CLI but the only
way to get things working again is to restart Asterisk altogether.
I finally cranked verbose & debugging way up (and watched my log files
go from 1mb/day to 100mb/day), but below I believe contains my problem.
The next line is 1.5 minutes later where I restart Asterisk.
As a general troubleshooting procedure, you want to ask yourself if you have
made any changes before it stopped working. If not, and especially if you
can restart and get it working again, I'd suspect some hardware failure.
(Assuming the problem is reproduceable - I had times when TDM card stopped
working with no trace of error.) Try installing on another box.
Yuan Liu
SIP/701 is a Grandstream GXP-2000 phone (we have about 30 of them in
place here). Zap/3-1 is a Digium TDM400.
I can't quite figure out where my problem is, is it the initial
exception, is it not getting hung up completely, does it have to do with
the call limit on the SIP channel, perhaps 'no provider found'
statements?
Any help would be appreciated, I have a relatively simple dial-plan, I
can send over relevant bits of it if necessary.
Thanks,
Ken
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Exception on 12, channel 3
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Got event On hook(1) on
channel 3 (index 0)
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on
channel 3
[Apr 19 13:51:13] DEBUG[27722] channel.c: Didn't get a frame from
channel: Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] channel.c: Bridge stops bridging channels
SIP/701-08ee6120 and Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel 'Zap/3-1'
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: zt_hangup(Zap/3-1)
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Hangup: channel: 3 index = 0,
normal = 12, callwait = -1, thirdcall = -1
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: disabled echo cancellation on
channel 3
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Set option TDD MODE, value:
OFF(0) on Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] chan_zap.c: Updated conferencing on 3,
with 0 conference users
[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Hungup 'Zap/3-1'
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel Zap/3-1
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension
(from-internal,201,2) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
(from-internal, 201, 2) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Launching 'Hangup'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: -- Executing
[EMAIL PROTECTED]:1] Hangup("SIP/701-08ee6120", "") in new stack
[Apr 19 13:51:13] DEBUG[27722] pbx.c: Spawn extension
(from-internal,h,1) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] VERBOSE[27722] logger.c: == Spawn extension
(from-internal, h, 1) exited non-zero on 'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] channel.c: Hanging up channel
'SIP/701-08ee6120'
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Hangup call SIP/701-08ee6120,
SIP callid [EMAIL PROTECTED])
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Updating call counter for
incoming call
[Apr 19 13:51:13] DEBUG[27722] chan_sip.c: Call from peer '701' removed
from call limit 6
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel SIP/701
[Apr 19 13:51:13] DEBUG[27722] devicestate.c: Notification of state
change to be queued on device/channel SIP/701-08ee6120
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for Zap - 3
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for Zap/3 -
state 0 (Unknown)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701
- state 1 (Not in use)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: Changing state for SIP/701
- state 1 (Not in use)
[Apr 19 13:51:13] DEBUG[20432] devicestate.c: No provider found,
checking channel drivers for SIP - 701
[Apr 19 13:51:13] DEBUG[20432] chan_sip.c: Checking device state for
peer 701
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Re: [Asterisk-Users] Asterisk stops responding when internet is down
Jay Milk wrote: Michael Strelnikov wrote: I just never used one. Is BIND good enough? dnsmasqd is quick and easy. All the joys of DNS caching without the pain of configuring a full-blown bind. Unless, of course, you do this sort of thing every day ;-) ___ --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 The software that this fellow is referring to is available at: http://thekelleys.org.uk/dnsmasq/doc.html It's widely used on Linux based routers too, so you use your router as the DNS server and it proxies to your ISP... and caches information. It works VERY well, but I've never personally tested it with helping with the DNS issue in Asterisk so if someone does then please post your results so others will know! -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
Michael Strelnikov wrote: I just never used one. Is BIND good enough? dnsmasqd is quick and easy. All the joys of DNS caching without the pain of configuring a full-blown bind. Unless, of course, you do this sort of thing every day ;-) ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
I just never used one. Is BIND good enough?On 4/12/06, Olle E Johansson <[EMAIL PROTECTED]> wrote: 12 apr 2006 kl. 08.46 skrev Michael Strelnikov:> What caching DNS do you recommend?>Anyone you feel comfortable running./O___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Best regards,Michael Strelnikov ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
12 apr 2006 kl. 08.46 skrev Michael Strelnikov: What caching DNS do you recommend? Anyone you feel comfortable running. /O ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
What caching DNS do you recommend?On 4/12/06, Olle E Johansson <[EMAIL PROTECTED]> wrote: 11 apr 2006 kl. 14.28 skrev Michael Strelnikov:> I do have that line. I also have all my phones defined by IP> address. But all providers are defined by names.>> On 4/10/06, Michiel van Baak < [EMAIL PROTECTED]> wrote:On> 22:14, Mon 10 Apr 06, Michael Strelnikov wrote:> > Hi,> >> >My * refuses SIP registrations when internet is down. All is > returing at> > the moment when outside connection is up. What is wrong?>> Try to set srvlookup=no in your sip.conf> Or put all the phone ip's in the servers /etc/hosts>> This is clearly a resolving issue >This has to do with the current DNS implementation in asterisk, whichis not very asynchronus. we are workingon fixing this. While waiting for that solution (hopefully in therelease after 1.4) I would guess that running a local caching DNS server on your LAN would help. Asterisk will then get aDNS reply, even if it says "sorry, have no answer".Sending DNS queries, not getting any response, kills Asterisk./O ---* Olle E. Johansson - [EMAIL PROTECTED] * MeetAsterisk European Tourhttp://www.meetasterisk.com* Asterisk Training http://edvina.net/training/___--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 -- Best regards,Michael Strelnikov ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
11 apr 2006 kl. 14.28 skrev Michael Strelnikov: I do have that line. I also have all my phones defined by IP address. But all providers are defined by names. On 4/10/06, Michiel van Baak < [EMAIL PROTECTED]> wrote:On 22:14, Mon 10 Apr 06, Michael Strelnikov wrote: > Hi, > >My * refuses SIP registrations when internet is down. All is returing at > the moment when outside connection is up. What is wrong? Try to set srvlookup=no in your sip.conf Or put all the phone ip's in the servers /etc/hosts This is clearly a resolving issue This has to do with the current DNS implementation in asterisk, which is not very asynchronus. we are working on fixing this. While waiting for that solution (hopefully in the release after 1.4) I would guess that running a local caching DNS server on your LAN would help. Asterisk will then get a DNS reply, even if it says "sorry, have no answer". Sending DNS queries, not getting any response, kills Asterisk. /O --- * Olle E. Johansson - [EMAIL PROTECTED] * MeetAsterisk European Tour http://www.meetasterisk.com * Asterisk Training http://edvina.net/training/ ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
Title: Re: [Asterisk-Users] Asterisk stops responding when internet is down Asterisk is sensitive when it comes to DNS lookups. If the DNS server configured on your Asterisk server is not reachable, Asterisk may block while waiting for a result. This can cause chan_sip to hang and not allow phones to register, or calls to be placed. One solution is to run a DNS server on the same machine, and cache results or use IP addresses instead of hostnames. -- Joshua Colp Software Developer Digium P - 256-428-6066 C - 506-878-0147 [EMAIL PROTECTED] ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
I do have that line. I also have all my phones defined by IP address. But all providers are defined by names.On 4/10/06, Michiel van Baak < [EMAIL PROTECTED]> wrote:On 22:14, Mon 10 Apr 06, Michael Strelnikov wrote: > Hi,>>My * refuses SIP registrations when internet is down. All is returing at> the moment when outside connection is up. What is wrong?Try to set srvlookup=no in your sip.conf Or put all the phone ip's in the servers /etc/hostsThis is clearly a resolving issue--Michiel van Baakhttp://michiel.vanbaak.info [EMAIL PROTECTED]GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D"Why is it drug addicts and computer afficionados are both called users?" ___--Bandwidth and Colocation provided by Easynews.com --Asterisk-Users mailing listTo UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users-- Best regards,Michael Strelnikov ___ --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
Re: [Asterisk-Users] Asterisk stops responding when internet is down
On 22:14, Mon 10 Apr 06, Michael Strelnikov wrote: > Hi, > >My * refuses SIP registrations when internet is down. All is returing at > the moment when outside connection is up. What is wrong? Try to set srvlookup=no in your sip.conf Or put all the phone ip's in the servers /etc/hosts This is clearly a resolving issue -- Michiel van Baak http://michiel.vanbaak.info [EMAIL PROTECTED] GnuPG key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x7E0B9A2D "Why is it drug addicts and computer afficionados are both called users?" ___ --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
Re: [Asterisk-Users] Asterisk Stops Responding
1) 8/29 2) The Agents are not used to * to hang up. I think it works until they hangup the phone, and then when they pick up again to login there is nothing (Dialtone, dtmf,...) 3) Can't hard hang up, see #2 :) 4) masqueraded? Not sure what you mean here. We use only Zap devices. No, VOIP. 5) nope. From my paste before. 25 and 26 are inbound/remote calls. 52, 54, 64, 65, 66 are all agents/local. The local phones would not work. But not ALL of them. It has only happened on agent phones. Everyone else in the building seems to work fine. Zap/66-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/54-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/26-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/25-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/52-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/65-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/64-1 (local 7100 1 ) Up AgentLogin (Empty) On Wednesday, September 3, 2003, at 10:27 AM, TC wrote: John 1) Is this from current CVS ??? 2) does the agent notice by the fact that they can't do a * to hang up the channel, in fact all dtmf is not recognized 3) if you do a hard hang up the agent line does it stay up 4) Does it only happend when the call is masqueraded to the agent line 5) if the remote hangs up the channel does the agent line come free If you ans yes to these items, I beleive I have duplicated this in testing last night with a config I was testing last night ... or is this a system wide deadlock ?? Can you do any other * functions outside of queues and agents, like dial an extension etc -Original Message- From: John Congdon <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Date: September 3, 2003 5:43 AM Subject: [Asterisk-Users] Asterisk Stops Responding This is getting to be a big problem. I am hoping it is something I have setup wrong somewhere... Various channels just freeze. It always appears to be the agents phones only. They will come to me and say the phones are down again. This morning here is what I see. I can not do STOP NOW. Just returns to the CLI prompt. I have to kill it. Notice that I try to hangup the channels and nothing happens. Any suggestions? ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Stops Responding
If you have the recent version of asterisk you propably should contact IRC or Mark (kram) to discover the new bug and fix it. It might take about 15 minutes . if you have time and you can find him. regards Martin On Wed, 3 Sep 2003, John Congdon wrote: > This is getting to be a big problem. I am hoping it is something > I have setup wrong somewhere... > > Various channels just freeze. It always appears to be the agents > phones only. They will come to me and say the phones are down again. > > This morning here is what I see. I can not do STOP NOW. Just returns > to > the CLI prompt. I have to kill it. Notice that I try to hangup the > channels and > nothing happens. > > Any suggestions? > > > > pbx*CLI> show channels > Channel (ContextExtensionPri ) State Appl. > Data > Zap/66-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/54-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/26-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/25-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/52-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/65-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/64-1 (local 7100 1 ) Up AgentLogin > (Empty) > 7 active channel(s) > pbx*CLI> soft hangup Za > Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 > pbx*CLI> soft hangup Zap/ > Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 > pbx*CLI> soft hangup Zap/ > Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 > pbx*CLI> soft hangup Zap/66-1 > Requested Hangup on channel 'Zap/66-1' > pbx*CLI> soft hangup Zap/65-1 > Requested Hangup on channel 'Zap/65-1' > pbx*CLI> soft hangup Zap/64-1 > Requested Hangup on channel 'Zap/64-1' > pbx*CLI> soft hangup Zap/54-1 > Requested Hangup on channel 'Zap/54-1' > pbx*CLI> soft hangup Zap/52-1 > Requested Hangup on channel 'Zap/52-1' > pbx*CLI> soft hangup Zap/25-1 > Requested Hangup on channel 'Zap/25-1' > pbx*CLI> soft hangup Zap/26-1 > Requested Hangup on channel 'Zap/26-1' > pbx*CLI> show channels > Channel (ContextExtensionPri ) State Appl. > Data > Zap/66-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/54-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/26-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/25-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/52-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/65-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/64-1 (local 7100 1 ) Up AgentLogin > (Empty) > 7 active channel(s) > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk Stops Responding
My last restart was 8 days ago. I'm now running: 'asterisk -vgc > /var/log/asterisk.log' from screen 0 and 'tail -f /var/log/asterisk.log' from screen 1 I have not had a crash since 8/25 and have run about 6580.18 minutes through the system over my PRI. Hardware: VALINUX 1220 PIII 866, 384Mb, 2x20 GHZ T100P cdr_mysql is being used and is 8/25 cvs. I do not load the ;driver=aopen in modem.conf Are you using the uniqueid feature from within cdr_mysql? I am, but I hacked it to work. I'm sure there is a place for me turn the uniqueid feature but I did not know where to turn it on from so I removed the if statement and force the insert of the uniquie id. Maybe that has something to do with it. Regards, TL -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of John Congdon Sent: Wednesday, September 03, 2003 8:40 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk Stops Responding This is getting to be a big problem. I am hoping it is something I have setup wrong somewhere... Various channels just freeze. It always appears to be the agents phones only. They will come to me and say the phones are down again. This morning here is what I see. I can not do STOP NOW. Just returns to the CLI prompt. I have to kill it. Notice that I try to hangup the channels and nothing happens. Any suggestions? pbx*CLI> show channels Channel (ContextExtensionPri ) State Appl. Data Zap/66-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/54-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/26-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/25-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/52-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/65-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/64-1 (local 7100 1 ) Up AgentLogin (Empty) 7 active channel(s) pbx*CLI> soft hangup Za Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 pbx*CLI> soft hangup Zap/ Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 pbx*CLI> soft hangup Zap/ Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 pbx*CLI> soft hangup Zap/66-1 Requested Hangup on channel 'Zap/66-1' pbx*CLI> soft hangup Zap/65-1 Requested Hangup on channel 'Zap/65-1' pbx*CLI> soft hangup Zap/64-1 Requested Hangup on channel 'Zap/64-1' pbx*CLI> soft hangup Zap/54-1 Requested Hangup on channel 'Zap/54-1' pbx*CLI> soft hangup Zap/52-1 Requested Hangup on channel 'Zap/52-1' pbx*CLI> soft hangup Zap/25-1 Requested Hangup on channel 'Zap/25-1' pbx*CLI> soft hangup Zap/26-1 Requested Hangup on channel 'Zap/26-1' pbx*CLI> show channels Channel (ContextExtensionPri ) State Appl. Data Zap/66-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/54-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/26-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/25-1 (macro-enqueue s105 ) Up Queue PillNetwork|t||pillnetwork Zap/52-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/65-1 (local 7100 1 ) Up AgentLogin (Empty) Zap/64-1 (local 7100 1 ) Up AgentLogin (Empty) 7 active channel(s) ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Stops Responding
Same thing here. I thought we had it stable until it froze up again. We are disabling more and more modules and features until we can figure out which one is the culprit. On Wednesday 03 September 2003 07:39, John Congdon wrote: > This is getting to be a big problem. I am hoping it is something > I have setup wrong somewhere... > > Various channels just freeze. It always appears to be the agents > phones only. They will come to me and say the phones are down again. > > This morning here is what I see. I can not do STOP NOW. Just returns > to > the CLI prompt. I have to kill it. Notice that I try to hangup the > channels and > nothing happens. > > Any suggestions? > > > > pbx*CLI> show channels > Channel (ContextExtensionPri ) State Appl. > Data > Zap/66-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/54-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/26-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/25-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/52-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/65-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/64-1 (local 7100 1 ) Up AgentLogin > (Empty) > 7 active channel(s) > pbx*CLI> soft hangup Za > Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 > pbx*CLI> soft hangup Zap/ > Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 > pbx*CLI> soft hangup Zap/ > Zap/25-1 Zap/26-1 Zap/52-1 Zap/54-1 Zap/64-1 Zap/65-1 Zap/66-1 > pbx*CLI> soft hangup Zap/66-1 > Requested Hangup on channel 'Zap/66-1' > pbx*CLI> soft hangup Zap/65-1 > Requested Hangup on channel 'Zap/65-1' > pbx*CLI> soft hangup Zap/64-1 > Requested Hangup on channel 'Zap/64-1' > pbx*CLI> soft hangup Zap/54-1 > Requested Hangup on channel 'Zap/54-1' > pbx*CLI> soft hangup Zap/52-1 > Requested Hangup on channel 'Zap/52-1' > pbx*CLI> soft hangup Zap/25-1 > Requested Hangup on channel 'Zap/25-1' > pbx*CLI> soft hangup Zap/26-1 > Requested Hangup on channel 'Zap/26-1' > pbx*CLI> show channels > Channel (ContextExtensionPri ) State Appl. > Data > Zap/66-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/54-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/26-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/25-1 (macro-enqueue s105 ) Up Queue > PillNetwork|t||pillnetwork > Zap/52-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/65-1 (local 7100 1 ) Up AgentLogin > (Empty) > Zap/64-1 (local 7100 1 ) Up AgentLogin > (Empty) > 7 active channel(s) > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk Stops Responding
John 1) Is this from current CVS ??? 2) does the agent notice by the fact that they can't do a * to hang up the channel, in fact all dtmf is not recognized 3) if you do a hard hang up the agent line does it stay up 4) Does it only happend when the call is masqueraded to the agent line 5) if the remote hangs up the channel does the agent line come free If you ans yes to these items, I beleive I have duplicated this in testing last night with a config I was testing last night ... or is this a system wide deadlock ?? Can you do any other * functions outside of queues and agents, like dial an extension etc -Original Message- From: John Congdon <[EMAIL PROTECTED]> To: [EMAIL PROTECTED] <[EMAIL PROTECTED]> Date: September 3, 2003 5:43 AM Subject: [Asterisk-Users] Asterisk Stops Responding >This is getting to be a big problem. I am hoping it is something >I have setup wrong somewhere... > >Various channels just freeze. It always appears to be the agents >phones only. They will come to me and say the phones are down again. > >This morning here is what I see. I can not do STOP NOW. Just returns >to >the CLI prompt. I have to kill it. Notice that I try to hangup the >channels and >nothing happens. > >Any suggestions? > > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stops responding
I have seen similar problems with using Asterisk as a voip UA (though not as bad or predictable as you.) The *8# bug was causing the bulk of my problems. http://bugs.digium.com/bug_view_page.php?bug_id=116 David Harris wrote: Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't have any digium cards installed. I use SIP only. I use cisco IP phones and cisco sip voice gateways. I have experienced the issue while using ztdummy and zaprtc as well as neither. The lockup occurs once every 2 hours or so during heavy use, which is like 50 calls an hour and people checking their voicemail and such. During off hours like the weekend it usually doesn't even crash at all even though we do have some call volume. probably like 20 calls all weekend. CLI commands such as "sip show peers", "show channels", "show modules", still return results. Thanks, David Harris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stops responding
I guess a field has been added to the cdr_mysql since my version. The uniqueid field. Do you think this could be causing this type of issue. I could probably test it. I bet you could make one call (or Agent Login), and once you hung up and it tried writing to mysql, it failed and that channel was no longer any good. John On Thursday, August 28, 2003, at 01:59 PM, John Congdon wrote: I had the same problem, and yes I am using cdr_mysql. What should be done? John On Thursday, August 28, 2003, at 12:44 PM, Martin Pycko wrote: Are you using cdr_mysql module ? (storing CDRs in mysql ?) regards Martin On Thu, 28 Aug 2003, David Harris wrote: Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't have any digium cards installed. I use SIP only. I use cisco IP phones and cisco sip voice gateways. I have experienced the issue while using ztdummy and zaprtc as well as neither. The lockup occurs once every 2 hours or so during heavy use, which is like 50 calls an hour and people checking their voicemail and such. During off hours like the weekend it usually doesn't even crash at all even though we do have some call volume. probably like 20 calls all weekend. CLI commands such as "sip show peers", "show channels", "show modules", still return results. Thanks, David Harris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stops responding
not to be picky, but what does that have to do with cdr_mysql? John On Thursday, August 28, 2003, at 02:48 PM, Martin Pycko wrote: Try to cvs update or edit /etc/asterisk/modems.conf and comment out driver=aopen line. regards Martin On Thu, 28 Aug 2003, John Congdon wrote: I had the same problem, and yes I am using cdr_mysql. What should be done? John On Thursday, August 28, 2003, at 12:44 PM, Martin Pycko wrote: Are you using cdr_mysql module ? (storing CDRs in mysql ?) regards Martin On Thu, 28 Aug 2003, David Harris wrote: Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't have any digium cards installed. I use SIP only. I use cisco IP phones and cisco sip voice gateways. I have experienced the issue while using ztdummy and zaprtc as well as neither. The lockup occurs once every 2 hours or so during heavy use, which is like 50 calls an hour and people checking their voicemail and such. During off hours like the weekend it usually doesn't even crash at all even though we do have some call volume. probably like 20 calls all weekend. CLI commands such as "sip show peers", "show channels", "show modules", still return results. Thanks, David Harris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stops responding
Try to cvs update or edit /etc/asterisk/modems.conf and comment out driver=aopen line. regards Martin On Thu, 28 Aug 2003, John Congdon wrote: > I had the same problem, and yes I am using cdr_mysql. > > What should be done? > > John > > > On Thursday, August 28, 2003, at 12:44 PM, Martin Pycko wrote: > > > Are you using cdr_mysql module ? (storing CDRs in mysql ?) > > > > regards > > Martin > > > > On Thu, 28 Aug 2003, David Harris wrote: > > > >> Anyone have any thoughts on why versions of asterisk I try (4 so far) > >> after CVS-07/18/03 always end up locking up on me... which means no > >> sip > >> clients can register/re-register and if I type "reload" or "stop now" > >> at > >> the cli it just returns and does nothing. > >> > >> I have experienced this same issue on three separate boxes. Two > >> running > >> RedHat 9 and one running Redhat 8. > >> > >> I don't have any digium cards installed. I use SIP only. I use cisco > >> IP phones and cisco sip voice gateways. I have experienced the issue > >> while using ztdummy and zaprtc as well as neither. > >> > >> The lockup occurs once every 2 hours or so during heavy use, which is > >> like 50 calls an hour and people checking their voicemail and such. > >> During off hours like the weekend it usually doesn't even crash at all > >> even though we do have some call volume. probably like 20 calls all > >> weekend. > >> > >> CLI commands such as "sip show peers", "show channels", "show > >> modules", > >> still return results. > >> > >> Thanks, > >> > >> David Harris > >> > >> ___ > >> Asterisk-Users mailing list > >> [EMAIL PROTECTED] > >> http://lists.digium.com/mailman/listinfo/asterisk-users > >> > > > > ___ > > Asterisk-Users mailing list > > [EMAIL PROTECTED] > > http://lists.digium.com/mailman/listinfo/asterisk-users > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stops responding
Hi, We have experienced the exact same problem. Pure SIP environment and yes...we store CDRs in MySQL. Version is: CVS-08/22/03 Gazing at the console I was able to determine the exact time Asterisk froze. Even with DEBGUG on it did not show anything important. The moment it freezes is when a call from Phone1 tries to connect to a SIP Provider like Iconnect: Phone1Our SIP Server---Our AsteriskSIP Provider It was by no means 100% reproducible. Maybe 1 out of 10 calls caused the trouble. A bad symptom would be that the command "show sip channels" would show several calls, even though they had hungup a long time ago. Troubleshooting revealed that the BYE message was not being sent by our SIP Server to the Asterisk server upon hangup. We rectified this and we no longer see those phantom SIP Channels and Aterisk has not froze for about a week. Regards, Andres http://www.telesip.net On Thursday 28 August 2003 11:08, David Harris wrote: > Anyone have any thoughts on why versions of asterisk I try (4 so far) > after CVS-07/18/03 always end up locking up on me... which means no sip > clients can register/re-register and if I type "reload" or "stop now" at > the cli it just returns and does nothing. > > I have experienced this same issue on three separate boxes. Two running > RedHat 9 and one running Redhat 8. > > I don't have any digium cards installed. I use SIP only. I use cisco > IP phones and cisco sip voice gateways. I have experienced the issue > while using ztdummy and zaprtc as well as neither. > > The lockup occurs once every 2 hours or so during heavy use, which is > like 50 calls an hour and people checking their voicemail and such. > During off hours like the weekend it usually doesn't even crash at all > even though we do have some call volume. probably like 20 calls all > weekend. > > CLI commands such as "sip show peers", "show channels", "show modules", > still return results. > > Thanks, > > David Harris > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stops responding
I had the same problem, and yes I am using cdr_mysql. What should be done? John On Thursday, August 28, 2003, at 12:44 PM, Martin Pycko wrote: Are you using cdr_mysql module ? (storing CDRs in mysql ?) regards Martin On Thu, 28 Aug 2003, David Harris wrote: Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't have any digium cards installed. I use SIP only. I use cisco IP phones and cisco sip voice gateways. I have experienced the issue while using ztdummy and zaprtc as well as neither. The lockup occurs once every 2 hours or so during heavy use, which is like 50 calls an hour and people checking their voicemail and such. During off hours like the weekend it usually doesn't even crash at all even though we do have some call volume. probably like 20 calls all weekend. CLI commands such as "sip show peers", "show channels", "show modules", still return results. Thanks, David Harris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
RE: [Asterisk-Users] Asterisk stops responding
Hi David- As mentioned, there was a bug in the CDR SQL module that caused a problem similar to what you describe, however the symptom I experienced in this case was that the "reload" command would start, but then hang up during the reload of one the modules. I was still able to "stop now" correctly however. I'm also Running RH9. A recent 8/26? CVS replaced cdr_sql, and I think it fixed that particular problem. I have noticed sometimes that asterisk -r fails to connect to a running asterisk server after a several hour period of time - I am trying to get more data on this before reporting to the bug list. Maybe this is related to your problem? Cheers Scott Scott M. Stingel Emerging Voice Technology Inc. Palo Alto, California and London, England Email: [EMAIL PROTECTED] URL:www.evtmedia.com -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of David Harris Sent: Thursday, August 28, 2003 9:09 AM To: [EMAIL PROTECTED] Subject: [Asterisk-Users] Asterisk stops responding Anyone have any thoughts on why versions of asterisk I try (4 so far) after CVS-07/18/03 always end up locking up on me... which means no sip clients can register/re-register and if I type "reload" or "stop now" at the cli it just returns and does nothing. I have experienced this same issue on three separate boxes. Two running RedHat 9 and one running Redhat 8. I don't have any digium cards installed. I use SIP only. I use cisco IP phones and cisco sip voice gateways. I have experienced the issue while using ztdummy and zaprtc as well as neither. The lockup occurs once every 2 hours or so during heavy use, which is like 50 calls an hour and people checking their voicemail and such. During off hours like the weekend it usually doesn't even crash at all even though we do have some call volume. probably like 20 calls all weekend. CLI commands such as "sip show peers", "show channels", "show modules", still return results. Thanks, David Harris ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [Asterisk-Users] Asterisk stops responding
Are you using cdr_mysql module ? (storing CDRs in mysql ?) regards Martin On Thu, 28 Aug 2003, David Harris wrote: > Anyone have any thoughts on why versions of asterisk I try (4 so far) > after CVS-07/18/03 always end up locking up on me... which means no sip > clients can register/re-register and if I type "reload" or "stop now" at > the cli it just returns and does nothing. > > I have experienced this same issue on three separate boxes. Two running > RedHat 9 and one running Redhat 8. > > I don't have any digium cards installed. I use SIP only. I use cisco > IP phones and cisco sip voice gateways. I have experienced the issue > while using ztdummy and zaprtc as well as neither. > > The lockup occurs once every 2 hours or so during heavy use, which is > like 50 calls an hour and people checking their voicemail and such. > During off hours like the weekend it usually doesn't even crash at all > even though we do have some call volume. probably like 20 calls all > weekend. > > CLI commands such as "sip show peers", "show channels", "show modules", > still return results. > > Thanks, > > David Harris > > ___ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > ___ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
