Incoming calls can't hear us but we can her them. This occurs ramdomly. We have tried letting it ring twice before answer
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Incoming calls
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Grandstream GXP-2140 can't rebuild config
Hello,
Are you trying to provision the phone ? What do you mean when you say save and rebuild ? Have you tried configuring ( registering ) manually ? Also, we would be glad to help you if you open a ticket with Grandstream Help-desk.
Thank you
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Bug or Design? Files erased automatically
Using PIAF Incredible PBX 13-12.2
I'm not sure if ths a FreePBX or PIAF issue
I have some scripts that play messages located in voicemail/default/(some extension number)/greet/.
If I populate these greet folders with voice files to play (wav) and then make a change using the GUI and then when I apply of configuration (red button), all my wav files are erased, but the folder is not. Actually, if the files are located in any folder below the voicemail folder does the same thing.
In the meanwhile, I've moved my wav files to sounds fold where there is no issues.
But for the life of me I do not understand why files would be erased without warning.
Bart
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Need an Gigaset IP Phone. Suggestions?
Hey,
I have an asterisk system setup on a remote server ( it's not behind NAT). My office router is a symmetric NAT. I have a TURN server setup remotely and I am using Zoiper softphones for extensions right now. (All extensions are behind NAT) I am looking to replace one extension by an IP phone. I see that Gigaset is the only provider with cordless IP phones. Digium or Sangoma don't provide ?
Can someone suggest a good Gigaset model that is good with freePBX ?
Thanks
Dinesh
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Debian8 fresh install Error Class Userman not found
I tried to downgrade php had issues. http://superuser.com/questions/913792/php-5-3-10-on-debian-jessie after searching a bit online I am unable to find anyone who is able to get past dependencies problems with libapache2-mod-php5.
So i re installed back to the 5.6 php . At this point I am going to try freepbx12
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VPN client option Not showing up for Yealink in EPM
After the latest set of updates I can no longer select the VPN client option under EPM Extension mapping for yealink phones. All that shows up in the dropdown is "none". Previously I could select the VPN client I created under Sys admin PRO.
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Bug or Design? Files erased automatically
Does this bug accurately describe what you are seeing? What version of Freepbx?
http://issues.freepbx.org/browse/FREEPBX-11080
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retrieve_conf failed, config not applied
Hi guys,
I'm experimenting a retrieve_conf failed error
It seems to follow a Digium Update from module admin.
I waslooking to get back to the previous version, but the module seems not to integrate this function
Would you have a recommandation to solde the problem ?
Thanks !
[root@localhost ~]# /var/lib/asterisk/bin/retrieve_conf
Whoops\Exception\ErrorException: Invalid argument supplied for foreach() in file /var/www/html/admin/modules/digium_phones/conf/res_digium_phone_devices.php on line 193
Stack trace:
1. Whoops\Exception\ErrorException->() /var/www/html/admin/modules/digium_phones/conf/res_digium_phone_devices.php:193
2. Whoops\Run->handleError() /var/www/html/admin/modules/digium_phones/conf/res_digium_phone_devices.php:193
3. res_digium_phone_devices() /var/www/html/admin/modules/digium_phones/functions.inc.php:435
4. digium_phones_conf->generateConf() /var/www/html/admin/libraries/BMO/FileHooks.class.php:65
5. FreePBX\FileHooks->processOldHooks() /var/www/html/admin/libraries/BMO/FileHooks.class.php:24
6. FreePBX\FileHooks->processFileHooks() /var/lib/asterisk/bin/retrieve_conf:823↧
Is Base station required for using Gigaset handsets?
Hey,
I'm quite new to VOIP ... so, this question might seem basic
I have been using softphones for a while with my asterisk system. I have decided to move to IP phones and started reading which would be good. I liked the idea of cordless phones, and decided gigaset would be a good option.
But, I see that they have a base station with them. Would a base station be required when we want a handset. Is it not possible to register a IP phone like a softphone ?
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retrieve_conf failed, config not applied
In order to pass php problems, i had to comment parts of the php scripts :
I think it's not the way to do it
res_digium_phone_devices.php
foreach ($queue['entries'] as $entry) {
if ($entry['deviceid'] == $deviceid) {
$doutput[] = "application=queue-{$queueid}-{$deviceid}";
}
}
res_digium_phone_applications.php
foreach ($conf->digium_phones->get_queues() as $queueid=>$queue) { foreach($queue['entries'] as $entry) { if ($entry['deviceid'] == null) { continue; } $output[] = "[queue-{$queueid}-{$entry['deviceid']}]"; $output[] = "type=application"; $output[] = "application=queue"; $output[] = "queue={$queueid}"; $output[] = "permission={$entry['permission']}"; if ($entry['member'] == null) { $output[] = "member=false"; } else { if ($entry['location'] != null) { $output[] = "location={$entry['location']}"; } /* Try to find the toggle feature code and use that */ $fcc = new featurecode('queues', 'que_toggle'); $toggle = $fcc->getCodeActive(); unset($fcc); if ($toggle != "") { $output[] = "login_exten={$toggle}{$queueid}@ext-queues"; $output[] = "logout_exten={$toggle}{$queueid}@ext-queues"; } else if ($amp_conf['GENERATE_LEGACY_QUEUE_CODES']) { $output[] = "login_exten={$queueid}*@ext-queues"; $output[] = "logout_exten={$queueid}**@ext-queues"; } } $output[] = ""; } }
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Is Base station required for using Gigaset handsets?
There are DECT cordless ip phones and wifi phones. Dect cordless phones need the base station which is what actually registers to the PBX. Typically, one base station can service multiple handsets. Wifi phones are cordless and autonomous and will register directly to the PBX using wifi infrastructure.
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Problem with delay in call setup
I had a similar issue that was a result of specifying a STUN server in the Asterisk SIP Settings. I didn't have the network and firewall (firewall device not related to the FreePBX Firewall Module) configured properly, so it was creating this delay. As soon as I removed the STUN Server setting and left it blank, everything was fine.
I have not yet re-enabled the STUN Server settings, just haven't had time to identify the proper firewall rules needed.
I hope that helps.
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Need an Gigaset IP Phone. Suggestions?
There are lots of Wireless IP phones available, I recommend DECT if it fits your needs but Wifi is also available. Here is a starting point.
http://www.voipsupply.com/voip-phones/cordless
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2 of 4 lines on external SIP phone keep dropping
Just upgraded to 66-7. I'll follow up in a few days to report.
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Debian8 fresh install Error Class Userman not found
Ok the freepbx 12 worked with version 11 asterisk
I have it functional
here is what I deleted and dropped and ran for reference if you tried on jessie 8 to run freepbx13 and latest asterisk. hopefully help someone.
i will try again in a few months
So i re-got the dependencies 9since there is a few differences
dropped the databases
removed a few directories (will cause issues if you do not!)
downloaded the latest freepbx12 and asterisk 11
apt-get install -y build-essential linux-headers-uname -r` openssh-server apache2 mysql-server\
mysql-client bison flex php5 php5-curl php5-cli php5-mysql php-pear php-db php5-gd curl sox\
libncurses5-dev libssl-dev libmysqlclient-dev mpg123 libxml2-dev libnewt-dev sqlite3\
libsqlite3-dev pkg-config automake libtool autoconf git subversion unixodbc-dev uuid uuid-dev\
libasound2-dev libogg-dev libvorbis-dev libcurl4-openssl-dev libical-dev libneon27-dev libsrtp0-dev\
libspandsp-dev libmyodbc
wget http://downloads.asterisk.org/pub/telephony/certified-asterisk/certified-asterisk-11.6-current.tar.gz
tar xvfz certified-asterisk-11.6-current.tar.gz
wget http://mirror.freepbx.org/modules/packages/freepbx/freepbx-12.0-latest.tgz
tar xvfz freepbx-12.0-latest.tgz
mysql -uroot -ppass -e "drop database asteriskcdrdb;"
mysql -uroot -ppass -e "drop database asterisk;"
rm -rf /var/www/html
rm -rf /etc/asterisk
rm -rf /etc/amportal.conf
rm -rf /etc/freepbx.conf
rm -rf /usr/lib/asterisk/modules
rm -rf /var/lib/asterisk
cd /usr/src
cd dahdi-linux-complete-*
make all && make install && make config
cd /usr/src
cd pjproject-2.4
CFLAGS='-DPJ_HAS_IPV6=1' ./configure --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr
make dep && make && make install
cd /usr/src
cd jansson-*
autoreconf -i
./configure && make && make install
cd /usr/src
cd /usr/src/certified-asterisk-11.6-cert11*
./configure
contrib/scripts/get_mp3_source.sh
make menuselect
make && make install && make config
ldconfig
mysqladmin -uroot -ppass create asterisk
mysqladmin -uroot -ppass create asteriskcdrdb
mysql -uroot -ppass -e "GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY 'pass'"
mysql -uroot -ppass -e "GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY 'pass'"
mysql -uroot -ppass -e "flush privileges;"
cd /usr/src/freepbx
./start_asterisk start
./install_amp --installdb --username=asteriskuser --password=pass`
lastly you need to
amportal chown
amportal a reload
amportal a ma refreshsignatures
amportal chown
they mention to do a amportal a ma installall but i always get errors when I do this so i avoid it and loginto the web interface and manually load the modules I want and I do this and process them and install some more if they depend on something and keep doing this then when I have the basics or what is nessary then i hit the apply button.. I will post a link on debian jessie
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Howto Debian Jessie 8 Asterisk 11 freepbx12
Load debian 8 , note I used this as a reference http://wiki.freepbx.org/display/FOP/Installing+FreePBX+12+on+Ubuntu+Server+14.04+LTS
apt-get install -y build-essential linux-headers-uname -r openssh-server apache2 mysql-server\
mysql-client bison flex php5 php5-curl php5-cli php5-mysql php-pear php-db php5-gd curl sox\
libncurses5-dev libssl-dev libmysqlclient-dev mpg123 libxml2-dev libnewt-dev sqlite3\
libsqlite3-dev pkg-config automake libtool autoconf git subversion unixodbc-dev uuid uuid-dev\
libasound2-dev libogg-dev libvorbis-dev libcurl4-openssl-dev libical-dev libneon27-dev libsrtp0-dev\
libspandsp-dev libmyodbc
///make sure to note your mysql root password you will need this later
cd /usr/src
wget http://downloads.asterisk.org/pub/telephony/dahdi-linux-complete/dahdi-linux-complete-current.tar.gz
wget http://downloads.asterisk.org/pub/telephony/libpri/libpri-1.4-current.tar.gz
wget http://downloads.asterisk.org/pub/telephony/asterisk/asterisk-12-current.tar.gz
git clone https://github.com/akheron/jansson.git
git clone https://github.com/asterisk/pjproject.git
cd /usr/src
wget http://mirror.freepbx.org/modules/packages/freepbx/freepbx-12.0-latest.tgz
tar xvfz freepbx-12.0-latest.tgz
cd /usr/src
tar xvfz dahdi-linux-complete-current.tar.gz
cd dahdi-linux-complete-*
make all && make install && make config
cd /usr/src
tar -xjvf pjproject-2.4.tar.bz2
cd pjproject-2.4
CFLAGS='-DPJ_HAS_IPV6=1' ./configure --enable-shared --disable-sound --disable-resample --disable-video --disable-opencore-amr
make dep && make && make install
cd /usr/src
tar vxfz jansson.tar.gz
cd jansson-*
autoreconf -i
./configure && make && make install
cd /usr/src
cd /usr/src/certified-asterisk-11.6-cert11*
./configure
contrib/scripts/get_mp3_source.sh
make menuselect
///select your sound files in the codecs you think you will use I use gsm and g729 and of course Ulaw so I load hold sounds and extra in these codecs
make && make install && make config
ldconfig
///rootmysqlpass is your mysql root password
///pass4astusr would be your asteriskuser password you want to use
mysqladmin -uroot -prootmysqlpass create asterisk
mysqladmin -uroot -prootmysqlpass create asteriskcdrdb
mysql -uroot -prootmysqlpass -e "GRANT ALL PRIVILEGES ON asterisk.* TO asteriskuser@localhost IDENTIFIED BY 'pass4astusr'"
mysql -uroot -prootmysqlpass -e "GRANT ALL PRIVILEGES ON asteriskcdrdb.* TO asteriskuser@localhost IDENTIFIED BY 'pass4astusr'"
mysql -uroot -prootmysqlpass -e "flush privileges;"
cd /usr/src/freepbx
./start_asterisk start
./install_amp --installdb --username=asteriskuser --password=astJku88
amportal chown
amportal a reload
amportal a ma refreshsignatures
amportal chown
now login to the http://yourip
and go update your modules in module admin leave the asterisk modules alone
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Incoming calls
We need more info, how are you getting your phone lines? What kind of phones are you using? Are they on a LAN or the public internet? If on the internet is the server behind a NAT?
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Yealink T-32G won't provision!
Running the latest FreePBX x64 v10.13.66-7 and EPM 13.0.19.1. My phones are not grabbing the .cfg file from my server. I did a factory reset on the phones, and I get nothing. I also have T46G phones on my reception desks, and they seem to be provisioning OK. Any ideas would be appreciated.
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Yealink T-32G won't provision!
What firmware on the phones?
Are you using tftp?
if you
cd /tftpboot/
ls
Do you see the .cfg for that MAC? in other words is it being generated?
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Multiple Extensions Answering A Call?
I'd solve it in a series of callfiles through AGI. Write a script that calls all your extensions and (when they pick up) connects them to the conference in question. It's not "simple", but it is straightforward.
Some of the connect technologies available implement "shared lines" (like Chan-SCCP-B and DAHDI) but I'm not sure you want to have all of the extensions be on a party line like that.
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