[asterisk-bugs] [Asterisk 0015099]: 1.6.1/DAHDI: No outgoing analogue calls possible

[Asterisk Bug Tracker]

The following issue has been SUBMITTED. 
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https://issues.asterisk.org/view.php?id=15099 
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Reported By:                reinerotto
Assigned To:                
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Project:                    Asterisk
Issue ID:                   15099
Category:                   Channels/chan_dahdi
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     new
Asterisk Version:           1.6.1.0 
Regression:                 No 
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
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Date Submitted:             2009-05-13 15:34 CDT
Last Modified:              2009-05-13 15:34 CDT
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Summary:                    1.6.1/DAHDI: No outgoing analogue calls possible
Description: 
Hi,

I can call from analogue phone to SIP-phone, I can receive inbound calls
on analogue phone, but can not do outbound analog calls.

That outgoing calls do not work, only happens when dahdi+asterisk are
loaded during boot-up on my OpenSuSE 11.1 system. When asterisk+dahdi are
started by hand, no problem !

So, first of all I disabled "Parallel Execution" of start-up scripts for
the services during boot, to start all services in serial. SuSE by default
starts services, which do not depend upon each others, in parallel within
the same run-level.
Second, I made shure, that dahdi is started "far erlier" than asterisk.
Third, I used the "Last Weapon of a desperate Programmer" (who suspects a
timing problem; I am a programming-dinosaur myself, never needed to use it
permanently): I introduced some time-delay in /etc/init.d/asterisk. Below
you see this in more detail. 
And now it works.

But that means, there is still a race-condition/timing issue/load-problem
etc. during start-up of dahdi and asterisk after boot. Which I also had
using asterisk/zaptel, but did not report. 


asterisk -vvvvvvvvvr, when trying to do analogue call from phone to PSTN:
.......
    -- Starting simple switch on 'DAHDI/2-1'
    -- Executing [470555 at to_analog_phone:1] Dial("DAHDI/2-1",
"dahdi/1/470555,10,tTm") in new stack
[May 13 19:10:10] WARNING[6461]: app_dial.c:1518 dial_exec_full: Unable to
create channel of type 'dahdi' (cause 0 - Unknown)
  == Everyone is busy/congested at this time (1:0/0/1)
    -- Executing [470555 at to_analog_phone:2] VoiceMail("DAHDI/2-1", "100")
in new stack
    -- <DAHDI/2-1> Playing 'vm-intro.gsm' (language 'ru')
  == Spawn extension (to_analog_phone, 470555, 2) exited non-zero on
'DAHDI/2-1'
    -- Hungup 'DAHDI/2-1'

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Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2009-05-13 15:34 reinerotto     New Issue                                    
2009-05-13 15:34 reinerotto     Asterisk Version          => 1.6.1.0         
2009-05-13 15:34 reinerotto     Regression                => No              
2009-05-13 15:34 reinerotto     SVN Branch (only for SVN checkouts, not tarball
releases) => N/A             
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