[asterisk-bugs] [JIRA] (ASTERISK-20440) [patch] No ringback towards SLAstation on outbound trunk call.

Matt Jordan (JIRA) noreply at issues.asterisk.org
Mon Nov 19 08:40:45 CST 2012


    [ https://issues.asterisk.org/jira/browse/ASTERISK-20440?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199846#comment-199846 ] 

Matt Jordan edited comment on ASTERISK-20440 at 11/19/12 8:40 AM:
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A few comments on the patch:
# Asterisk coding guidelines [specify that variables should not be declared mid-block|https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines#CodingGuidelines-Declarationoffunctionsandvariables] - so your declaration of {{last_state}} and {{current_state}} should be moved up to the top of {{dial_trunk}}
# You need to introduce a 'break' after the assignment of done.  Previously, this would fall through to the 'break' statement of the {{AST_DIAL_RESULT_PROCEEDING}} case - now, it falls through into your state change in {{AST_DIAL_RESULT_TRYING}}.  This means you'll assign a state of {{AST_CONTROL_PROGRESS}} when a channel has hung up - there's even a chance it will attempt to indicate that to the channel when its hung up (which would be bad).  This leads to:
# Move the {{if (done) break;}} check before your indication.  That way, if something has hung up, you're guaranteed to bail out before you attempt to indicate to the channel.  That makes the code a bit more robust in the face of future changes.

                
      was (Author: mjordan):
    Two comments on the patch:
# Asterisk coding guidelines [specify that variables should not be declared mid-block|https://wiki.asterisk.org/wiki/display/AST/Coding+Guidelines#CodingGuidelines-Declarationoffunctionsandvariables] - so your declaration of {{last_state}} and {{current_state}} should be moved up to the top of {{dial_trunk}}
# You need to introduce a 'break' after the assignment of done.  Previously, this would fall through to the 'break' statement of the {{AST_DIAL_RESULT_PROCEEDING}} case - now, it falls through into your state change in {{AST_DIAL_RESULT_TRYING}}.  This means you'll assign a state of {{AST_CONTROL_PROGRESS}} when a channel has hung up - there's even a chance it will attempt to indicate that to the channel when its hung up (which would be bad).  This leads to:
# Move the {{if (done) break;}} check before your indication.  That way, if something has hung up, you're guaranteed to bail out before you attempt to indicate to the channel.  That makes the code a bit more robust in the face of future changes.

                  
> [patch] No ringback towards SLAstation on outbound trunk call.
> --------------------------------------------------------------
>
>                 Key: ASTERISK-20440
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20440
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_meetme, Applications/SLA
>    Affects Versions: 1.8.15.0
>         Environment: Astlinux
>            Reporter: dkerr
>            Assignee: dkerr
>         Attachments: asterisk-trunk-bugid20440v2.patch, Direct Outbound.txt, DISA outbound.txt, extensions.conf, Simulated Inbound.txt, Simulated Outbound Direct.txt, Simulated Outbound DISA.txt, sip.conf, sla.conf, SLAstation log.txt
>
>
> This is similar to ASTERISK-11549 except that the problem is with calls in the other direction. If I make an outbound call from a SLA phone using the SLAStation then there is no ringtone audible to the phone that originates the call.  Both SLA stations and SLA trunks are SIP channels and for outbound I have tried both the DISA() method of connecting to a trunk (get dialtone from DISA and then enter a number to call) and directly connecting (using Dial() application instead of DISA()).
> I have turned on debug and verbose in the asterisk CLI and can observe the SIP messages and dialplan execution.  A SIP RINGING message is received by asterisk from the trunk, but the message is not passed on to the SLA station.  I have set progressinband=yes in the sip.conf and this made no difference.

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