[asterisk-users] SIP/IAX peers UNREACHABLE and audio loss

[Rajeev Natarajan]

Well, we have add similar issues - do you use a media gateway /.IP Phones /
softphones as your extensions?

We were running Audiocodes and for some reason (I suspect a poor ethernet
switch), when there are more than 15 people using the line, Audiocodes will
not respond to a qualify and asterisk will drop the call. Turned off qualify
(removed qualify=yes) and <still keeping fingers crossed> things seem fine.

Rajeev

On 3/23/07, Edoardo Serra <edoardo.serra at webrainstorm.it> wrote:
>
> Hi all,
>         I'm having a problem with some Asterisk servers interconnected
> with
> each other using IAX (I also tried with SIP without solving the problem)
>
> Sometimes, with apparently no reason, some peers become UNREACHABLE
> (I have qualify=yes in iax.conf) and REACHABLE again as soon as
> another qualify test is made.
>
> Our users are also complaining about audio loss during their calls,
> apparently randomly, everything goes ok for days and bad for another few
> days.
>
> I strongly believe the 2 problems are strictly related because in the
> logs I see REACHABLE / UNREACHABLE messages only for certains days
> without regularity.
> The days in wich i see a lot of messages are exactly the days with
> most of complaint about audio loss
>
> I just noticed that timestamps of the logs (REACHABLE / UNREACHABLE)
> are quite always during business hours, this makes me think at somewhat
> related to load (cpu load, badwidth load, calls load, etc...)
>
> But, looking at hardware specs of our lan, servers and average load I
> don't think they are over-stressed.
>
> Our servers are all:
> 2 x Intel(R) Xeon(TM) CPU 3.20GHz
> 1 GB RAM
> 2 x IDE HDDs Software RAID 1
> Asterisk 1.2.13 with res_perl
> Gentoo Linux
> Some of them has a Sangoma card connected with an E1
>
> Most ot these are on the same LAN, interconnected with a 1 GB switch
> (I don't think it should be a bandwidth problem).
>
> Load averages of these server is varying from 0.5 to 1.0
> (I guess it should be ok)
>
> On each server we don't have more than 50 concurrent calls
> (bridged SIP <-> IAX2 or IAX2 <-> ZAP)
>
> Used codec is mostly G729
>
> Sometimes on asterisk cli i see some messages like
> "Avoided initial deadlock for '0x9fd130', 10 retries!"
> I don't know if it could be somehow related.
>
> Someone of you can point me in the right direction ?
>
> Tnx in advance
>
> Regards
>
> Ing. Edoardo Serra
> WeBRainstorm S.r.l.
>
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