[asterisk-bugs] [JIRA] (ASTERISK-25169) No audio from voicemail app with v13.4.0 on Grandstream GXP20XX phones

[Idon (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-25169?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=226983#comment-226983 ] 

Idon commented on ASTERISK-25169:
---------------------------------

Sorry about not getting this done earlier.  Had a rough couple of weeks.  Here goes:

Install Ubuntu 14.04.2 LTS Server and Apply Updates:
uname -a
Linux proto 3.16.0-43-generic #58~14.04.1-Ubuntu SMP Mon Jun 22 10:21:20 UTC 2015 x86_64 x86_64 x86_64 GNU/Linux


Become Root:
sudo -i


Install Build Environment and Essential Libraries
apt-get install build-essential

apt-get install git-core pkg-config subversion sqlite autoconf automake libtool libxml2-dev libncurses5-dev unixodbc unixodbc-dev libasound2-dev libogg-dev libvorbis-dev libneon27-dev libsrtp0-dev libspandsp-dev libmyodbc uuid uuid-dev sqlite3 libsqlite3-dev libssl-dev libgnutls-dev libsrtp0-dev bison flex libcurl4-openssl-dev libsnmp-dev libspeex-dev libspeexdsp-dev libgsm1-dev libmp3lame-dev libldap2-dev libltdl-dev liburiparser-dev libxslt1-dev libmysqlclient-dev libpq-dev sox liblua5.2-dev binutils-dev libedit-dev libpopt-dev lua5.2 libusb-dev


Download Required Sources:
cd /usr/src

wget http://www.digip.org/jansson/releases/jansson-2.7.tar.gz
tar xvf jansson-2.7.tar.gz

wget http://www.pjsip.org/release/2.4/pjproject-2.4.tar.bz2
tar xvf pjproject-2.4.tar.bz2

wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-13-current.tar.gz
tar xvf asterisk-13-current.tar.gz


Build and Install Jansson Library:
cd /usr/src/jansson-2.7
./configure --libdir=/usr/lib
make & make install


Build and Install PJSIP:
cd /usr/src/pjproject-2.4
./configure --enable-shared --prefix=/usr --libdir=/usr/lib --disable-video --disable-sound --disable-resample --disable-opencore-amr --with-external-speex --with-external-srtp --with-external-gsm CFLAGS='-O2 -DNDEBUG'
make dep
make
make install

ldconfig

ldconfig -p | grep pj


Build and Install Asterisk:
cd /usr/src/asterisk-13.4.0
./configure --prefix=/opt/asterisk
make
make install
make samples
make config


Set Lower Permissions for Asterisk:
groupadd --system asterisk
adduser --system --gid $(expr "$(cat /etc/group |grep asterisk)" : 'asterisk.*:\([0-9]*\):') --no-create-home asterisk
chown -R asterisk:asterisk /opt/asterisk


Add Asterisk to System Path:
In the /etc/profile.d/ directory, create a file called asterisk.sh with the following content:
export PATH="$PATH:/opt/asterisk/sbin"

Reload profile:
. /etc/profile


Place the Four Configuration Files, Attached to This bug Report, into:
cd /opt/asterisk/etc/asterisk


Start Asterisk:
service asterisk start



> No audio from voicemail app with v13.4.0 on Grandstream GXP20XX phones
> ----------------------------------------------------------------------
>
>                 Key: ASTERISK-25169
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-25169
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Applications/app_voicemail
>    Affects Versions: 13.4.0
>         Environment: Ubuntu 14.04.2 LTS x64
> Vmware ESXi 5.5
> Guest memory allocation: 1GB
>            Reporter: Idon
>            Assignee: Rusty Newton
>
> When either calling into the voicemail application directly or being transferred to voicemail (i.e. when called party is unavailable), none of the voicemail prompts can be heard.  Looking at the Asterisk console, it is evident that it is being processed, but there is no audio.
> This is a regression introduced in Asterisk 13.4.0 and, so far, appears to affects only Grandstream phones.  The specific models we have that I've tested against are GXP2000 and GXP2020.
> So far, no other IP phone (Snom and Digium) or softphone (Jitsi) I've tested against has an issue with voicemail and 13.4.0.
> Reverting to 13.2.0 resolves the issue.
> I'm not sure what else may be required to help track down the issue.  I tried debugging RTP in the console and I do see communication between the phone and Asterisk; therefore, Asterisk must be sending some malformed bit of data that's causing Asterisk to ignore the stream.
> Let me know if there's some additional bit of info that's needed.



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