[asterisk-bugs] [JIRA] (ASTERISK-27896) tests/channels/SIP/SDP_attribute_passthrough: Requires codec_speex.

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27896?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=243684#comment-243684 ] 

Asterisk Team commented on ASTERISK-27896:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

> tests/channels/SIP/SDP_attribute_passthrough: Requires codec_speex.
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-27896
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27896
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Tests/testsuite
>    Affects Versions: GIT
>            Reporter: Alexander Traud
>            Severity: Minor
>              Labels: patch
>         Attachments: test_speex.patch
>
>
> Without the module codec_speex, the Asterisk Test Suite fails on one of the tests{code}WARNING[24788]: asterisk.sipp:524 processEnded: Resolving remote host '127.0.0.1'... Done.
> WARNING[24788]: asterisk.sipp:524 processEnded: Aborting call on an unexpected CANCEL for call: 07ee49b90e93cf527571e91c6d4a7100 at 127.0.0.1:5060.
> WARNING[24788]: asterisk.sipp:628 __scenario_callback: SIPp Scenario phone_B_speex.xml Failed [1]
> WARNING[24788]: asterisk.sipp:637 __evaluate_scenario_results: SIPp Scenario phone_B_speex.xml Failed
> WARNING[24788]: asterisk.sipp:524 processEnded: Resolving remote host '127.0.0.1'... Done.
> WARNING[24788]: asterisk.sipp:524 processEnded: Aborting call on unexpected message for Call-Id '1-24866 at 127.0.0.2': while expecting '180' (index 2), received 'SIP/2.0 603 Declined
> Via: SIP/2.0/UDP 127.0.0.2:5065;branch=z9hG4bK-24866-1-0;received=127.0.0.2
> From: test1 <sip:phoneA at 127.0.0.2:5065>;tag=1
> To: test <sip:test at 127.0.0.1:5060>;tag=as547b0540
> Call-ID: 1-24866 at 127.0.0.2
> CSeq: 1 INVITE
> Server: Asterisk PBX UNKNOWN__and_probably_unsupported
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Content-Length: 0'.
> WARNING[24788]: asterisk.sipp:628 __scenario_callback: SIPp Scenario phone_A_speex.xml Failed [1]
> WARNING[24788]: asterisk.sipp:637 __evaluate_scenario_results: SIPp Scenario phone_A_speex.xml Failed
> Test ['tests/channels/SIP/SDP_attribute_passthrough/run-test'] failed{code}*Workaround* (Debian/Ubuntu)
> {code}sudo apt install libspeexdsp-dev
> ./configure --enable-dev-mode
> make
> sudo make install{code}*Note*
> The attached patch fixes this by listing this dependency.



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