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benny+usenet at amorse...
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 Posted: Tue Mar 18, 2008 2:30 pm Post subject: [asterisk-users] Asterisk 1.4 reliability problems |
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"Steve Totaro" <stotaro at totarotechnologies.com> writes:
| Quote: | I will probably continue this train of thought (1.2.X is more
production ready) until these threads stop popping up on the list.
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I think you're being too kind to 1.2.x. It has numerous problems, most
especially with locking in chan_sip. 1.4.x is a HUGE improvement.
/Benny |
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astmattf at gmail.com
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| Posted: Tue Mar 18, 2008 3:13 pm Post subject: [asterisk-users] Asterisk 1.4 reliability problems |
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On 3/18/08, Benny Amorsen <benny+usenet at amorsen.dk> wrote:
| Quote: | "Steve Totaro" <stotaro at totarotechnologies.com> writes:
| Quote: | I will probably continue this train of thought (1.2.X is more
production ready) until these threads stop popping up on the list.
|
I think you're being too kind to 1.2.x. It has numerous problems, most
especially with locking in chan_sip. 1.4.x is a HUGE improvement.
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Who uses chan_sip? Long live IAX! 
But seriously, several of my clients use SIP exclusively, passing tens
of thousand of calls a day on Asterisk 1.2.X with no issues. I have
noticed that the load is slightly lower for SIP-only in 1.4, but I
have not noticed any stability issues revolving around SIP on 1.2.X.
MATT--- |
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matt at venturevoip.com
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| Posted: Tue Mar 18, 2008 8:41 pm Post subject: [asterisk-users] Asterisk 1.4 reliability problems |
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-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Have you tried disabling highpriority=yes in asterisk.conf?
- --
Kind Regards,
Matt Riddell
Director
_______________________________________________
http://www.venturevoip.com (Great new VoIP end to end solution)
http://www.venturevoip.com/news.php (Daily Asterisk News - html)
http://www.venturevoip.com/newrssfeed.php (Daily Asterisk News - rss)
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benny+usenet at amorse...
Guest
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| Posted: Wed Mar 19, 2008 5:11 pm Post subject: [asterisk-users] Asterisk 1.4 reliability problems |
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"Matt Florell" <astmattf at gmail.com> writes:
| Quote: | But seriously, several of my clients use SIP exclusively, passing tens
of thousand of calls a day on Asterisk 1.2.X with no issues. I have
noticed that the load is slightly lower for SIP-only in 1.4, but I
have not noticed any stability issues revolving around SIP on 1.2.X.
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No hung calls? Our 1.2.x customer PBX's are drowning in "channel.c:
Avoided deadlock for '0x91dbee8', 9 retries!". Of course you can just
ignore the hung calls if you want, but they mess up hint state and
prevent graceful restarts. 1.4.x fixes it.
/Benny |
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astmattf at gmail.com
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| Posted: Thu Mar 20, 2008 12:18 am Post subject: [asterisk-users] Asterisk 1.4 reliability problems |
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On 3/19/08, Benny Amorsen <benny+usenet at amorsen.dk> wrote:
| Quote: | "Matt Florell" <astmattf at gmail.com> writes:
| Quote: | But seriously, several of my clients use SIP exclusively, passing tens
of thousand of calls a day on Asterisk 1.2.X with no issues. I have
noticed that the load is slightly lower for SIP-only in 1.4, but I
have not noticed any stability issues revolving around SIP on 1.2.X.
|
No hung calls? Our 1.2.x customer PBX's are drowning in "channel.c:
Avoided deadlock for '0x91dbee8', 9 retries!". Of course you can just
ignore the hung calls if you want, but they mess up hint state and
prevent graceful restarts. 1.4.x fixes it.
|
I will say that we did notice some SIP issues with older 1.2 releases,
but on the current 1.2.24+ releases we really haven't had many
problems, and we do not have hung channels. I should mention that most
of these installations have all phones on a LAN and almost none of the
calls are native SIP-bridged since they go through meetme rooms which
might account for why we do not see problems like this.
As for 1.4.X we are moving closer to putting a live production machine
on it, just a few more weeks of testing like we have had for the last
month, and I should be convinced of it's stability.
MATT--- |
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bwentdg at pipeline.com
Guest
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| Posted: Thu Mar 20, 2008 12:29 am Post subject: [asterisk-users] Asterisk 1.4 reliability problems |
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You hit the nail on the head when you said "
, just a few more weeks of testing like we have had for the last
month"
Some of the code you see makes you think the "testing" was slightly more than a clean compile
Matt Florell wrote:
| Quote: | On 3/19/08, Benny Amorsen <benny+usenet at amorsen.dk> wrote:
| Quote: | "Matt Florell" <astmattf at gmail.com> writes:
| Quote: | But seriously, several of my clients use SIP exclusively, passing tens
of thousand of calls a day on Asterisk 1.2.X with no issues. I have
noticed that the load is slightly lower for SIP-only in 1.4, but I
have not noticed any stability issues revolving around SIP on 1.2.X.
|
No hung calls? Our 1.2.x customer PBX's are drowning in "channel.c:
Avoided deadlock for '0x91dbee8', 9 retries!". Of course you can just
ignore the hung calls if you want, but they mess up hint state and
prevent graceful restarts. 1.4.x fixes it.
|
I will say that we did notice some SIP issues with older 1.2 releases,
but on the current 1.2.24+ releases we really haven't had many
problems, and we do not have hung channels. I should mention that most
of these installations have all phones on a LAN and almost none of the
calls are native SIP-bridged since they go through meetme rooms which
might account for why we do not see problems like this.
As for 1.4.X we are moving closer to putting a live production machine
on it, just a few more weeks of testing like we have had for the last
month, and I should be convinced of it's stability.
MATT---
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