asterisk users - May 2005 - RED ALARM on PRI channel takes Asterisk DOWN

have an asterisk box (P4, 1GB RAM, etc ) with a sangoma two E1 port card in
it. I have used a E1 cross cable to connect the two E1 ports together and
pass calls through them for testing purposes allo works fine.  also have
multiple voip phones connected to this setup,

what i noticed is that when i pull any one end of the E1 (breaking the E1
connection) I get multiple RED ALARMS on the zap channels I understand this
is ok and should happen if the E1 link breaks but my problem is that asterisk
stops doing a lot of other things too like i have extensions configured to
meetme, voicemail and other aplications they all stop working i get no
sound but when i call one hardphone to another it works i guess thats cause
the rtp streams go from phone to phone directly. but why does asterisk stop
passing anything through itself. Also if anyone else has experience this is
this only with sangoma or does it happen with asterisk too. I have a digium
card but its a single port and i dont have a real E1 link to test with. )

Few lines from the dump i get on the asterisk cli. the first few lines are
the red alarm and the rest of me trying to call up an extension

May  5 11:07:15 NOTICE[309]: chan_zap.c:7395 pri_dchannel: PRI got event:
Alarm (4) on Primary D-channel of span 2
May  5 11:07:15 WARNING[309]: chan_zap.c:1931 pri_find_dchan: No D-channels
available!  Using Primary on channel anyway 47!
    -- Executing Answer("SIP/1000-f92e", "") in new stack
    -- Executing MusicOnHold("SIP/1000-f92e", "") in new
stack
    -- Started music on hold, class 'default', on SIP/1000-f92e
  == Primary D-Channel on span 1 down
May  5 11:07:22 WARNING[308]: chan_zap.c:1931 pri_find_dchan: No D-channels
available!  Using Primary on channel anyway 16!
  == Primary D-Channel on span 2 down
May  5 11:07:22 WARNING[309]: chan_zap.c:1931 pri_find_dchan: No D-channels
available!  Using Primary on channel anyway 47!
    -- Stopped music on hold on SIP/1000-f92e
  == Spawn extension (default, 501, 2) exited non-zero on
'SIP/1000-f92e'

-- 
regards
Vikram

On Thu, 5 May 2005, Vikram Rangnekar wrote:
> what i noticed is that when i pull any one end of the E1 (breaking the E1
> connection) I get multiple RED ALARMS on the zap channels I understand this
> is ok and should happen if the E1 link breaks but my problem is that
asterisk
> stops doing a lot of other things too like i have extensions configured to
> meetme, voicemail and other aplications they all stop working i get no
> sound but when i call one hardphone to another it works i guess thats cause
> the rtp streams go from phone to phone directly. but why does asterisk stop
> passing anything through itself. Also if anyone else has experience this is
> this only with sangoma or does it happen with asterisk too. I have a digium
> card but its a single port and i dont have a real E1 link to test with. )
All the things mentioned above that stop working when the E1 is 
disconnected require zaptel clocking. Zaptel clocking is normally derived 
from the E1 clock, or may be generated internally by a zaptel card. 

My guess is that the Sangoma card does not switch to internal clocking 
when the external clocking is lost, thus depriving Asterisk of the zaptel 
clock source that drives a lot of the internal transmissions. 

You may be able to compensate by having the ztdummy module loaded. Doing 
so may cause problems. I have never tried having both real zaptel drivers 
loaded at the same time as ztdummy.

Peter

Sangoma is great prompt response to the problem, its fixed now
Use the latest stable latest stable 2.3.2-3



+++ Vikram Rangnekar [05/05/05 08:58 +0200]:
> 
>  have an asterisk box (P4, 1GB RAM, etc ) with a sangoma two E1 port card
in
> it. I have used a E1 cross cable to connect the two E1 ports together and
> pass calls through them for testing purposes allo works fine.  also have
> multiple voip phones connected to this setup,
> 
> what i noticed is that when i pull any one end of the E1 (breaking the E1
> connection) I get multiple RED ALARMS on the zap channels I understand this
> is ok and should happen if the E1 link breaks but my problem is that
asterisk
> stops doing a lot of other things too like i have extensions configured to
> meetme, voicemail and other aplications they all stop working i get no
> sound but when i call one hardphone to another it works i guess thats cause
> the rtp streams go from phone to phone directly. but why does asterisk stop
> passing anything through itself. Also if anyone else has experience this is
> this only with sangoma or does it happen with asterisk too. I have a digium
> card but its a single port and i dont have a real E1 link to test with. )
> 
> Few lines from the dump i get on the asterisk cli. the first few lines are
> the red alarm and the rest of me trying to call up an extension
> 
-- 
regards
Vikram (http://www.vicramresearch.com)