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Summary:ASTERISK-25253: confbridge volume options and other volume controls such as func_volume don't work
Reporter:Dmitriy Serov (Demon)Labels:
Date Opened:2015-07-15 12:21:46Date Closed:2015-07-22 20:04:18
Priority:BlockerRegression?Yes
Status:Closed/CompleteComponents:Applications/app_confbridge
Versions:SVN 13.4.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Attachments:
Description:app_confbridge, some user menu actions don't work:
*4=decrease_listening_volume
4=decrease_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*9=increase_talking_volume
9=increase_talking_volume

regression
[Edit by Rusty - A few notes]
* Tested in GIT-13-1aafadf (built on 2015-7-15)
* Tested with chan_sip and chan_pjsip, issue occurs with both.
* A few non-volume options work, such as toggle_mute and leave_conference
Comments:By: Asterisk Team (asteriskteam) 2015-07-15 12:21:47.523-0500

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].
By: Rusty Newton (rnewton) 2015-07-16 18:40:34.655-0500

Debug output when activating toggle_mute:

{noformat}
[Jul 16 18:36:34] DEBUG[30935]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '127.0.1.1:0' into...
[Jul 16 18:36:34] DEBUG[30935]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '127.0.1.1' and port '0'.
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: res_rtp_asterisk.c:3549 create_dtmf_frame: Creating BEGIN DTMF Frame: 49 (1), at 10.24.18.16:4046
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: res_rtp_asterisk.c:3549 create_dtmf_frame: Creating END DTMF Frame: 49 (1), at 10.24.18.16:4046
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: bridge_channel.c:1528 bridge_channel_feature_digit_add: DTMF feature string on 0x7faf28009118(SIP/ALICE-00000005) is now '1'
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: bridge_channel.c:1606 ast_bridge_channel_feature_digit: DTMF feature hook 0x7faf28000920 matched DTMF string '1' on 0x7faf28009118(SIP/ALICE-00000005)
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: res_rtp_asterisk.c:2852 ast_rtp_update_source: Setting the marker bit due to a source update
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: app_confbridge.c:1040 conf_update_user_mute: User SIP/ALICE-00000005 is muted: user:1 system:0.
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: channel.c:5490 set_format: Channel SIP/ALICE-00000005 setting write format path: gsm -> ulaw
[Jul 16 18:36:35] DEBUG[30986][C-00000005]: channel.c:3402 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second
   -- <SIP/ALICE-00000005> Playing 'conf-muted.gsm' (language 'en')
[Jul 16 18:36:36] DEBUG[30929]: acl.c:958 ast_find_ourip: Not an IPv4 nor IPv6 address, cannot get port.
[Jul 16 18:36:36] DEBUG[30929]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'newtonr-laptop' into...
{noformat}
Debug output when attempting to activate increase_listening_volume:

{noformat}
[Jul 16 18:37:51] DEBUG[30935]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '10.24.18.138' and port '4039'.
[Jul 16 18:37:51] DEBUG[30935]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '127.0.1.1:0' into...
[Jul 16 18:37:51] DEBUG[30935]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '127.0.1.1' and port '0'.
[Jul 16 18:37:53] DEBUG[30986][C-00000005]: res_rtp_asterisk.c:3549 create_dtmf_frame: Creating BEGIN DTMF Frame: 54 (6), at 10.24.18.16:4046
[Jul 16 18:37:53] DEBUG[30883]: threadpool.c:508 grow: Increasing threadpool stasis-core's size by 1
[Jul 16 18:37:53] DEBUG[30986][C-00000005]: res_rtp_asterisk.c:3549 create_dtmf_frame: Creating END DTMF Frame: 54 (6), at 10.24.18.16:4046
[Jul 16 18:37:53] DEBUG[30986][C-00000005]: bridge_channel.c:1528 bridge_channel_feature_digit_add: DTMF feature string on 0x7faf28009118(SIP/ALICE-00000005) is now '6'
[Jul 16 18:37:53] DEBUG[30986][C-00000005]: bridge_channel.c:1606 ast_bridge_channel_feature_digit: DTMF feature hook 0x7faf280077a0 matched DTMF string '6' on 0x7faf28009118(SIP/ALICE-00000005)
[Jul 16 18:37:53] DEBUG[30986][C-00000005]: res_rtp_asterisk.c:2852 ast_rtp_update_source: Setting the marker bit due to a source update
[Jul 16 18:37:53] DEBUG[30986][C-00000005]: res_rtp_asterisk.c:2852 ast_rtp_update_source: Setting the marker bit due to a source update
[Jul 16 18:37:53] DEBUG[30929]: acl.c:958 ast_find_ourip: Not an IPv4 nor IPv6 address, cannot get port.
[Jul 16 18:37:53] DEBUG[30929]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'newtonr-laptop' into...
[Jul 16 18:37:53] DEBUG[30929]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'newtonr-laptop' and port ''.
{noformat}

The user menu looks like:

{noformat}
[sample_user_menu]
type=menu
*=playback_and_continue(conf-usermenu)
*1=toggle_mute
1=toggle_mute
*4=decrease_listening_volume
4=decrease_listening_volume
*6=increase_listening_volume
6=increase_listening_volume
*7=decrease_talking_volume
7=decrease_talking_volume
*8=leave_conference
8=leave_conference
*9=increase_talking_volume
9=increase_talking_volume
{noformat}
By: Rusty Newton (rnewton) 2015-07-17 13:02:03.570-0500

The issue is a little more broad than we thought originally. It affects func_volume as well.. possibly other features that control volume on a call. Updated the summary to reflect this.