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Summary:ASTERISK-25169: No audio from voicemail app with v13.4.0 on Grandstream GXP20XX phones
Reporter:Idon (snax)Labels:
Date Opened:2015-06-17 12:44:03Date Closed:2015-09-18 12:35:31
Priority:MajorRegression?Yes
Status:Closed/CompleteComponents:Applications/app_voicemail
Versions:13.4.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:Ubuntu 14.04.2 LTS x64 Vmware ESXi 5.5 Guest memory allocation: 1GBAttachments:( 0) extensions.conf
( 1) grandstream.cap
( 2) jitsi.cap
( 3) modules.conf
( 4) pjsip.conf
( 5) voicemail.conf
Description: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.
Comments:By: Rusty Newton (rnewton) 2015-06-17 13:54:19.095-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


By: Rusty Newton (rnewton) 2015-06-17 13:56:02.027-0500

As my previous comment mentions, we could certainly use more information.

Expect that we need to reproduce the issue in order to investigate it. That means we require the debug mentioned and your configuration files (only what is necessary to reproduce the issue). That could be sip.conf or pjsip.conf, dialplan such as extensions.conf and maybe voicemail.conf.

You should build a minimal test scenario where you can reproduce the issue and then post the necessary files here and describe how to reproduce.
By: Rusty Newton (rnewton) 2015-06-17 13:57:04.260-0500

In addition to the debug and configuration, please provide a packet capture (viewable in wireshark) that correlates to the debug collected.

Use 'send back' or 'enter feedback' to send the issue back to us when ready.
Thanks!

By: Idon (snax) 2015-06-30 12:21:23.369-0500

Will be posting the additional info within the next week.  Been busy with other things, including my computer dying.
By: Asterisk Team (asteriskteam) 2015-07-15 12:00:18.317-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
By: Rusty Newton (rnewton) 2015-07-16 16:54:09.563-0500

This was auto-closed. Despite the reporter not responding I'm going to take a closer look at this.
By: Idon (snax) 2015-07-21 12:22:56.318-0500

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


By: Idon (snax) 2015-07-21 12:26:03.600-0500

PJSIP Configuration File
By: Idon (snax) 2015-07-21 12:26:39.771-0500

Dialplan Configuration File
By: Idon (snax) 2015-07-21 12:27:09.708-0500

Modules Configuration File
By: Idon (snax) 2015-07-21 12:27:38.991-0500

Voicemail Configuration File
By: Idon (snax) 2015-07-21 12:40:19.684-0500

Steps to Reproduce:
1. Setup an Asterisk 13.4.0 server as per the detailed build and configuration steps provided in the prior comment
2. Configure a Grandstream GXP20XX phone and another non-Grandstream phone (can be a softphone) to register with that server using extension 100 and password 'secret'
3. Dial the voicemail at extension 1000, using either phone.
4. Note that with the non-Grandstream phone you hear the voicemail prompts; however, with the aforementioned Grandstream model(s), you hear nothing -- even though the Asterisk console indicates that the prompts are being played.
5. Compare this with Asterisk 13.2.0, which does not exhibit this issue.

By: Idon (snax) 2015-07-21 12:44:50.838-0500

Sorry.  That last post had a bunch of trivial typos -- in case you were wondering, "WTH?"  I've edited the post to correct those.
By: Idon (snax) 2015-07-21 15:11:34.002-0500

tcpdump capture of Jitsi voicemail session that works fine.
By: Idon (snax) 2015-07-21 15:12:35.678-0500

tcpdump capture of Grandstream voicemail session that does not work.
By: Idon (snax) 2015-07-21 15:21:21.612-0500

By the way, the captures were with the following command line on the Asterisk host:
tcpdump -n "(not ("broadcast or multicast")) and (udp dst port 5060 or src port 5060 or dst portrange 10000-20000 or src portrange 10000-20000)" -w /tmp/filename.cap
By: Idon (snax) 2015-07-21 17:27:20.968-0500

tcpdump capture of Grandstream voicemail session that does not work
By: Idon (snax) 2015-09-10 14:47:49.098-0500

Is anyone looking into this?  Asterisk 13.5.0 has been released and there has been no effort to resolve this.
By: Rusty Newton (rnewton) 2015-09-11 08:35:12.965-0500

I'll take a look at this today. I think it slipped through the cracks due to the "reopened" status. Thanks for providing thorough details on how to reproduce that should be helpful.
By: Rusty Newton (rnewton) 2015-09-11 09:14:37.531-0500

Howdy I setup your configuration files and followed your guidance on reproducing. I'm unable to reproduce the issue in 13.5.0 with a Grandstream GXP2120 or a Digium D40. I was able to hear all prompts and voicemail recordings playback fine.

I'll see if I can locate a 20XX model and try again..
By: Rusty Newton (rnewton) 2015-09-11 09:27:59.249-0500

I may not have one available. Would it possible that you have a 2120 that you could try to verify you can or cannot reproduce the issue with that model?

What firmware versions are you running on the 2000 and 2020 ?
By: Idon (snax) 2015-09-11 11:19:01.047-0500

This is very strange.  Following your replies, I decided to try again and now I get the prompts.

For Grandstreams, all we have are 2000, 2020, and GXV3175 (which is a crappy phone we don't use).  The 20XX phones are running firmware version 1.2.5.3 and 1.2.4.3, respectively.

It is possible that the issue has somehow resolved itself, but I'm not sure of an exact cause.  There were some network changes, so who knows.  The fact that only Grandstream phones were affected makes it difficult to guess and probably suggest quality problems with their software.

I will continue to test over the next few days and let you know, if the issue is truly resolved.

Thanks

By: Rusty Newton (rnewton) 2015-09-11 16:20:20.181-0500

Thanks for letting us know. I'll watch for your update and in the meantime see if I can locate another model.
By: Idon (snax) 2015-09-18 12:28:46.978-0500

Should be fine to close this out.  It appears it was some sort of Grandstream breaking network issue.

Thanks