[asterisk-bugs] [JIRA] (ASTERISK-25249) Features code not working for called party when Local channels are involved

Etienne Lessard (JIRA) noreply at issues.asterisk.org
Tue Jul 14 10:33:33 CDT 2015


Etienne Lessard created ASTERISK-25249:
------------------------------------------

             Summary: Features code not working for called party when Local channels are involved
                 Key: ASTERISK-25249
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25249
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Features
    Affects Versions: 13.4.0
            Reporter: Etienne Lessard


In Asterisk 13, features that are activated through DTMF codes are not working for the called party when Local channels are involved.

Given I have the following extensions.conf:
{noformat}
[internal]

exten = 102,1,Dial(SIP/bob)

exten = 112,1,Dial(SIP/bob,,ht)

exten = 122,1,Dial(Local/102 at internal,,ht)

exten = 132,1,Dial(Local/102 at internal/n,,ht)

exten = 142,1,Dial(Local/112 at internal)
{noformat}
And I have the following sip.conf:
{noformat}
[general]
directmedia = no
dtmfmode = rfc2833
language = fr_FR

[alice]
host = dynamic
context = internal
callerid = "Alice" <101>
secret = S2BFNK
type = friend
setvar = TRANSFER_CONTEXT=internal

[bob]
host = dynamic
context = internal
callerid = "Bob" <102>
secret = 82NWM0
type = friend
setvar = TRANSFER_CONTEXT=internal
{noformat}
And I have the following features.conf:
{noformat}
[general]

[featuremap]
disconnect = *0
blindxfer = *1
atxfer = *2
automon = *3
{noformat}
When Alice dials 122
And Bob answers
And Bob press *0
Then the calls it not hung up (expected: the call is hung up)

See full.log for the associated asterisk log output.

This is a changed in behaviour from asterisk 11.

I've taken a look at some other option to make feature codes work when Local channel are involved:
* it  you add the /n modifier for the Local channel (extension 132 in my example), then feature code for the called party "works", but if you try an attended transfer for example, then the pbx-transfer sound will be played in language "en" instead of "fr_FR", and the TRANSFER_CONTEXT variable used is not the TRANSFER_CONTEXT of the called party but the TRANSFER_CONTEXT of the calling party.
* if you put the "ht" option in the Dial executed by the Local channel (extension 142 in my example), then feature code for the called party works, but this is not a practical solution in more complex dialplan

In my example, I've used the Dial application, but it also affects other applications, like the Queue application for example.



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list