GOOD NEWS Incredible PBX for RasPi3

[Waffull]

Nevermind... It was a partition size problem. I had expanded the partition, but not the logical root one. All seems to be running now that it has plenty of space.

Thanks. I ran the script, and similar to @Jebs2k above, it was a fresh incrediblepbx 13-13 install.. now asterisk/mysql won't start now. I'm on a RasPi 3B+. I decided to try another reboot and this is what it reported:

root@incrediblepbx:~# reboot
Exception: SQLSTATE[HY000] [2002] Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2)::SQLSTATE[HY000] [2002] Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) in file /var/www/html/admin/libraries/utility.functions.php on line 204
Stack trace:
1. Exception->() /var/www/html/admin/libraries/utility.functions.php:204
2. die_freepbx() /var/www/html/admin/libraries/BMO/Database.class.php:136
3. PDOException->() /var/www/html/admin/libraries/BMO/Database.class.php:133
4. PDO->__construct() /var/www/html/admin/libraries/BMO/Database.class.php:133
5. FreePBX\Database->__construct() /var/www/html/admin/libraries/BMO/FreePBX.class.php:69
6. FreePBX->__construct() /var/www/html/admin/bootstrap.php:139
7. require_once() /etc/freepbx.conf:9
8. include_once() /usr/local/sbin/fwconsole:12
[ ok ] Stopping mysql (via systemctl): mysql.service.
[....] Stopping apache2 (via systemctl): apache2.serviceWarning: Unit file of apache2.service changed on disk, 'systemctl daemon-reload' recommended.
. ok
​

 

[Waffull]

Is http://incrediblepbx.com/gvsip-naf-raspi.tar.gz being kept up to date with the latest commits from https://github.com/naf419/asterisk/tree/gvsip?

Just wondering if we redownload and run through the install again, will those changes be applied?

Thank you,
Marc

Minor tweak to the re-register script this morning. If you've already installed and are experiencing issues with failed incoming calls, download and untar the tarball to /tmp and

cp -p /tmp/gvsip-register /root/.

 

[wardmundy]

WiFi wasn't working so had to do some reading around
https://www.raspberrypi.org/documentation/configuration/wireless/wireless-cli.md
https://wiki.archlinux.org/index.php/dhcpcd#10-wpa_supplicant
https://manpages.debian.org/jessie/wpasupplicant/wpa_supplicant.conf.5.en.html
Then tried a few commands:

 wpa_cli -i wlan0 reconfigure
Failed to connect to non-global ctrl_ifname: wlan0  error: No such file or directory

File /etc/wpa_supplicant/wpa_supplicant.conf was pointing to non existant ctrl_interface=DIR

File /etc/wpa_supplicant/wpa_supplicant.conf
country=GB
ctrl_interface=DIR=/var/run/wpa_supplicant GROUP=netdev
update_config=1

So to create the directory:

mkdir /var/run/wpa_supplicant
chown root:netdev /var/run/wpa_supplicant
chmod 775 /var/run/wpa_supplicant

Tip (from the ArchLinux writeup), if wanting new IP for eth0 and/or wlan0 find the lease file here

 ls -al /var/lib/dhcpcd5/dhcpcd*
-rw-r--r-- 1 root root 278 Jul  3 00:30 /var/lib/dhcpcd5/dhcpcd-eth0.lease
-rw-r--r-- 1 root root 278 Jul  3 00:25 /var/lib/dhcpcd5/dhcpcd-wlan0.lease

then rename or delete the lease file
mv /var/lib/dhcpcd5/dhcpcd-wlan0.lease /var/lib/dhcpcd5/dhcpcd-wlan0.lease.old
mv /var/lib/dhcpcd5/dhcpcd-eth0.lease /var/lib/dhcpcd5/dhcpcd-eth0.lease.old

Try invoke /sbin/wpa_supplicant directly with

  /sbin/wpa_supplicant -P /var/run/wpa_supplicant.wlan0.pid -i wlan0 \
 -D nl80211,wext -c /etc/wpa_supplicant/wpa_supplicant.conf
Successfully initialized wpa_supplicant
wlan0: Trying to associate with 18:90:d8:ad:8a:87 (SSID='SKY355F7' freq=2462 MHz)
wlan0: Associated with 18:90:d8:ad:8a:87
wlan0: WPA: Key negotiation completed with 18:90:d8:ad:8a:87 [PTK=CCMP GTK=CCMP]
wlan0: CTRL-EVENT-CONNECTED - Connection to 18:90:d8:ad:8a:87 completed [id=0 id_str=SKY355F7-PBX.A13.G12]
wlan0: CTRL-EVENT-REGDOM-CHANGE init=COUNTRY_IE type=COUNTRY alpha2=GB
^C (did a ctrl+C on hang to terminate)
wlan0: CTRL-EVENT-DISCONNECTED bssid=18:90:d8:ad:8a:87 reason=3 locally_generated=1
wlan0: CTRL-EVENT-TERMINATING

 ifup wlan0
pbxstatus

OK SOLVED
@wardmundy please add the missing DIR

 ls -al  /var/run/wpa_supplican*
-rw-r--r-- 1 root root    6 Jul  3 00:25 /var/run/wpa_supplicant.wlan0.pid

/var/run/wpa_supplicant:
total 0
drwxr-x---  2 root netdev  60 Jul  3 00:25 .
drwxr-xr-x 24 root root   900 Jul  3 00:33 ..
srwxrwx---  1 root netdev   0 Jul  3 00:25 wlan0

 ls -ald /var/run/wpa_supplicant
drwxr-x--- 2 root netdev 60 Jul  3 01:30 /var/run/wpa_supplicant

I built a fresh RasPi the other day, and all of these files and directories were already in place. Were you using the latest build and where did you get it??

 

[wardmundy]

Is http://incrediblepbx.com/gvsip-naf-raspi.tar.gz being kept up to date with the latest commits from https://github.com/naf419/asterisk/tree/gvsip?

Just wondering if we redownload and run through the install again, will those changes be applied?

Thank you,
Marc

Absolutely. And please do install it again. Many, many changes both by NAF, me, and others. Seems to be completely stable now without any refreshes of the registration. One of the major problems was the the username in pjsip_custom.conf for your GV account MUST be unique, and we were using gv1 for everybody. Now it's a combination of random numbers coupled with your GV 10-digit phone number. Really bad design on Google's end, but there's not much we can do about it.

For example, if you discover someone else's phone number or GV acct name and they happen to be registering with that, you could register with the same username and blow their trunk out of the water. As more and more people switch to GVSIP, this could pose a serious problem without a little security on the chosen usernames.

 

[Waffull]

Other than the security issue, sounds like great news. I'll install again. I had reverted back to using the direct connections to GV on my Obi devices for the stability... Looking forward to being able to swtich back to pbx.

BTW, will this script work with RASPBX or only IncrediblePBX?

Thanks,
Marc

Absolutely. And please do install it again. Many, many changes both by NAF, me, and others. Seems to be completely stable now without any refreshes of the registration. One of the major problems was the the username in pjsip_custom.conf for your GV account MUST be unique, and we were using gv1 for everybody. Now it's a combination of random numbers coupled with your GV 10-digit phone number. Really bad design on Google's end, but there's not much we can do about it.

For example, if you discover someone else's phone number or GV acct name and they happen to be registering with that, you could register with the same username and blow their trunk out of the water. As more and more people switch to GVSIP, this could pose a serious problem without a little security on the chosen usernames.

Only works with Incredible PBX for the RasPi.

 

[wardmundy]

Let’s Celebrate: Google Voice GVSIP Comes to Raspberry Pi

 

[Seattle]

I have problem with Configuring Google Voice with Incredible PBX. It was fresh install last night. It is showing:Requested TLS. How to fix it?

 

[Waffull]

Only works with Incredible PBX for the RasPi.

Thanks for the reply. I'll see if I can migrate my settings from RASPBX over to IncrediblePBX.

That said, I now have IncrediblePBX up and running and ran the naf-raspi script again.

Incoming calls seem to be working, but I do get the following error with the inbound calls:

[2018-07-04 15:28:07] ERROR[2750]: pjproject:0 <?>: icess0x6df0002 ...Error sending STUN request: Network is unreachable
[2018-07-04 15:28:08] ERROR[1934]: pjproject:0 <?>: icess0x6df0002 ..Error sending STUN request: Network is unreachable

Outbound calls produce the following errors:

[2018-07-04 15:33:53] ERROR[3191]: pjproject:0 <?>: sip_msg ...Header with no vptr encountered!! Current buffer: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.2.154:5061;rport;branch=z9hG4bKPj8a742e5e-9a26-41bd-aeae-70e4e3efd402;alias
From: "701" <sip:[email protected]>;tag=dea43cb4-a3b1-4aeb-9375-80cf24da521c
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: 0def6df8-1e76-41fe-a342-69c7a61e8711
CSeq: 978 INVITE

[2018-07-04 15:33:53] ERROR[3191]: pjproject:0 <?>: icess0x75b1bca ......Error sending STUN request: Network is unreachable
[2018-07-04 15:33:53] ERROR[1934]: pjproject:0 <?>: icess0x75b1bca ..Error sending STUN request: Network is unreachable

Both inbound and outgoing calls are currently working, despite the errors.

 

[Waffull]

What would need changing to get this to print temp in farenheit?

Thanks.

pbxstatus mod for Raspberry Pi users:

Adds CPU temperature and chan_dongle details: name, provider, signal level and states.

replace /usr/local/sbin/pbxstatus with the code below:

#!/bin/bash
DN="\Z1\ZrDN\Z0\ZR";
UP="\Zr\Z2UP\Z0\ZR";

if hostname -I > /etc/hostip; then
    _IP="Private IP: `cat /etc/hostip`"
  if [ -f "/etc/pbx/.phone" ]; then
    _PHONE="Ph: `cat /etc/pbx/.phone`"
  else
    _PHONE=" "
  fi
    _PUBLIC="Public  IP: `curl -s -S --user-agent "Mozilla/4.0" http://myip.incrediblepbx.com | awk 'NR==2'` $_PHONE"
else
   _IP=""
   _PUBLIC=""
fi
version=`cat /etc/pbx/.version`
astversion=`asterisk -rx "core show version" | grep -m 2 ^ | tail -1 | cut -f 2 -d " "`
fpbxversion=`cat /var/www/html/admin/modules/core/module.xml | grep version | head -n 1 | cut -f 2 -d ">" | cut -f 1 -d "<"`
OSrelease=`cat /etc/os-release | grep "PRETTY_NAME" | cut -f 2 -d "=" | sed "s|\"||g" | sed "s|(Core)|Core|"`
#OSrelease=" `cat /etc/os-release | cut -f 1 -d "(" | sed 's|Scientific Linux release|Scientific Linux|'`"

test=`cat /etc/os-release | grep "Scientific Linux"`
if [[ -z $test ]]; then
 TITLE="Incredible PBX $version for Raspberry Pi 2/3"
else
 TITLE="  Incredible PBX $version for Scientific Linux"
fi

test=`asterisk -rx "cdr show status" | grep Enabled`
if [[ -z $test ]]; then
 asterisk="$DN"
else
 asterisk="$UP"
fi
test=`service apache2 status | grep running`
if [[ -z $test ]]; then
 apache="$DN"
else
 apache="$UP"
fi
test=`ps aux | grep -c mysqld`
if [ "$test" -lt "2" ]; then
 mysql="$DN"
else
 mysql="$UP"
fi
test=`service webmin status | grep -c running`
if [[ -z $test ]]; then
 webmin="$DN"
else
 webmin="$UP"
fi
test=`service ssh status | grep -c running`
if [[ -z $test ]]; then
 ssh="$DN"
else
 ssh="$UP"
fi
test=`asterisk -rx "module show like sccp" | grep -c Running`
if [[ -z $test ]]; then
 sccp="$DN"
else
 sccp="$UP"
fi
test=`ps aux | grep -c sendmail`
if [ "$test" -lt "2" ]; then
 sendmail="$DN"
else
 sendmail="$UP"
fi
test=`ps aux | grep -c faxgetty`
if [ "$test" -lt "2" ]; then
 faxgetty="$DN"
else
 faxgetty="$UP"
fi
test=`ps aux | grep -c iaxmodem`
if [ "$test" -lt "2" ]; then
 iaxmodem="$DN"
else
 iaxmodem="$UP"
fi
test=`service fail2ban status | grep running`
if [[ -z "$test" ]]; then
 fail2ban="$DN"
else
 fail2ban="$UP"
fi
test=`ps aux | grep -c ntpd`
if [ "$test" -lt "2" ]; then
 ntpd="$DN"
else
 ntpd="$UP"
fi
test=`iptables -nL | grep DROP`
if [[ -z $test ]]; then
 IPtables="$DN"
else
 IPtables="$UP"
fi
#IPPORT=`ifconfig | head -1 | cut -f 1 -d ":"`
IPPORT=`ifconfig | grep inet | head -1 | cut -f 2 -d ":" | cut -f 1 -d " "`
#IPPORT='eth0'
test=`/sbin/ifconfig | grep -c inet`
if [ "$test" -lt "2" ]; then
 eth0="$DN"
else
 eth0="$UP"
fi

test=`cat /etc/pbx/.version`
if [[ "$test" < "13-12.17" ]]; then
 gvoauth="$UP"
else
 gvoauth="$UP"
fi
test=`ps aux | grep -c hfaxd`
if [ "$test" -lt "2" ]; then
 hfax="$DN"
else
 hfax="$UP"
fi
test=`ps aux | grep -c nrservice`
if [ "$test" -lt "2" ]; then
 nrclient="$DN"
else
 nrclient="$UP"
fi
test=`asterisk -rx "module show like chan_dongle" | grep -c Running`
if [[ -z $test ]]; then
 chan_dongle="$DN"
else
 chan_dongle="$UP"
fi

revision=`cat /proc/cpuinfo | grep Revision | cut -f 2 -d ":"`
if [ "$revision" = " a21041" ]; then
  model="2B"
elif [ "$revision" = " a01041" ]; then
  model="2B"
elif [ "$revision" = " a02082" ]; then
  model="3B"
elif [ "$revision" = " a22042" ]; then
  model="2B"
elif [ "$revision" = " a01040" ]; then
  model="2B"
elif [ "$revision" = " a02082" ]; then
  model="3B"
elif [ "$revision" = " a22082" ]; then
  model="3B"
elif [ "$revision" = " a32082" ]; then
  model="3B"
elif [ "$revision" = " 9000c1" ]; then
  model="ZeroW"
elif [ "${revision:0:3}" = " 00" ]; then
  model="1"
elif [ "$revision" = " 900092" ]; then
  model="0"
else
  model="3"
fi
TITLE="Incredible PBX $version for Raspberry Pi $model"


RAM=`free | tr -s " " | grep -m 2 ^ | tail -1 | cut -f 4 -d " "`
if [ -z "$RAM" ]; then
 RAM="`free | grep -m 2 ^ | tail -1 | cut -f 18 -d " "`"
 if [ -z "$RAM" ]; then
  RAM=" `free | grep -m 2 ^ | tail -1 | cut -f 19 -d " "`"
 fi
fi
DISK=`df -h | tr -s ' ' |grep " /" | head -n 1 | cut -f 4 -d " "`
#DISK=`df -h | grep -m 2 ^ | tail -1 | cut -f 15 -d " "`
if [ -z "$DISK" ]; then
 DISK=" `df -h | grep -m 2 ^ | tail -1 | cut -f 9 -d " "`"
fi
if [ "${#DISK}" -eq "4" ]; then
 DISK=" $DISK"
fi

NOW=`date`

#Temperature
TEMP=`/opt/vc/bin/vcgencmd measure_temp | cut -d '=' -f2 | sed 's/..$//'`

TEMPINT=`printf '%.*f\n' 0 $TEMP`

if ((TEMPINT <= 40)); then
  TEMP="4\ZB$TEMP°C"
elif ((TEMPINT >= 40 && TEMPINT <= 50)); then
  TEMP="2\ZB$TEMP°C"
elif ((TEMPINT >= 50 && TEMPINT <= 60)); then
  TEMP="3$TEMP°C!"
elif ((TEMPINT >= 60)); then
  TEMP="1$TEMP°C!!"
fi

#chan_dongle
devs=0
IFS=$'\n'
dongles=`asterisk -rx "dongle show devices" | sed -n '1!p'`
for don in $dongles
 do
 devs=$((devs+1))
 DEV=`echo $don | cut -c 1-13`
 RSTATE=`echo $don | cut -c 20-30 | sed 's/ //g'`
 PROV=`echo $don | cut -c 49-59 | sed 's/[^ ]\+/\L\u&/g'`
 RSSI=`echo $don | cut -c 31-32 | sed 's/ //g'`
 
if ((RSSI <= 2)); then
  RSSI="\Z1OOOO"
elif ((RSSI >= 2 && RSSI <= 9)); then
  RSSI="\Z1●OOO"
elif ((RSSI >= 10 && RSSI <= 14)); then
  RSSI="\Z3●●OO"
elif ((RSSI >= 15 && RSSI <= 19)); then
  RSSI="\Z6●●●O"
elif ((RSSI >= 20)); then
  RSSI="\Z2●●●●"
fi

PROV="$PROV$RSSI "
 
  case "$RSTATE" in
   Free)
    state="\Z2Free";;
   Dialing)
    state="\Z4Dialing";;
   Outgoing)
    state="\Z6Outgoing call";;
   Notconnec)
    state="\Z1Not Connected";;
   Incoming)
    state="\Z5Incoming call";;
  esac

 cdmsg="$cdmsg$DEV$PROV\Zr$state\ZR\Z0\n"
 headers="\nDevice       Provider    Ɏ   State\n\ZB"
done
unset IFS
#end of chan_dongle

MSGLEN=`echo ${#TEMP}`

TOP="\n\ZbAsterisk\ZB: $asterisk      \ZbApache\ZB: $apache \Zb MariaDB\ZB: $mysql\n\ZbSendMail\ZB: $sendmail    \ZbIPtables\ZB: $IPtables      \ZbSSH\ZB: $ssh\n\ZbLAN port\ZB: $eth0    \ZbFail2Ban\ZB: $fail2ban   \ZbWebmin\ZB: $webmin\n\ZbGV OAUTH\ZB: $gvoauth  \Zb Chan-SCCP\ZB: $sccp   \ZbNR VPN\ZB: $nrclient\n\ZbFaxGetty\ZB: $faxgetty  \Zb IAX Modem\ZB: $iaxmodem  \ZbHylaFax\ZB: $hfax\n"
#DNG="\Z0\ZbChDongle\ZB: $chan_dongle     \ZbDevices:  $devs\n$cdmsg\Z$TEMP\n\Zr\Z4\Zb"
if [[ $MSGLEN > 7 ]]; then
 DNG="\Z0\ZbTemp \Z$TEMP \Z0\ZbChan_Dongle\ZB: $chan_dongle  \ZbDevices:  $devs\n$cdmsg\n\Zr\Z4\Zb"
else
 DNG="\Z0\ZbTemp: \Z$TEMP \Z0\ZbChan_Dongle\ZB: $chan_dongle  \ZbDevices:  $devs\n$cdmsg\n\Zr\Z4\Zb"
fi
MID="RAM\ZB:$RAM  Raspbian (Jessie)  \ZbDisk\ZB:$DISK\ZR\Z0\n\n\Z5\Zb"
BOT="Asterisk $astversion  +  GUI ver. $fpbxversion\ZB\Z0\n\n\ZR\Zb$_IP\n\n$_PUBLIC\ZB\n\n\Zb\Z4System Time: $NOW"


MSGLEN=`echo ${#MID}`

if [ "$MSGLEN" -gt "95" ]; then
# MID2=`echo $MID| sed -e 's/  / /g'`
 MID2=$MID
 MID=`echo $MID2| sed -e 's/Disk/ Disk/g'`
 MID2=`echo $MID| sed -e 's/: /:/g'`
fi

MSG="$TOP$DNG$MID$BOT"

lines=$((22 + devs))

dialog --no-lines --colors --title "$TITLE" --msgbox "$MSG" $lines 47

clear

 

[mrfeh]

Let’s Celebrate: Google Voice GVSIP Comes to Raspberry Pi

I went through this procedure today, but things are not working for me.

Everything compiled without error, and I rebooted the server. I don't see any changes to my google voice account (there is no obitalk device), and incoming calls do not reach my server. I also don't see any new trunks or outbound routes in FreePBX; should I?

When I try to place an outgoing call, here is the asterisk logging (destination phone number removed):

== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
> 0xb6324220 -- Strict RTP learning after remote address set to: 192.168.2.21:5004
-- Executing [6085------@from-internal:1] Set("SIP/1024-00000003", "CALLERID(dnid)=6085------") in new stack
-- Executing [6085------@from-internal:2] Set("SIP/1024-00000003", "CALLERID(dnid)=16085------") in new stack
-- Executing [6085------7@from-internal:3] Goto("SIP/1024-00000003", "16085------,1") in new stack
-- Goto (from-internal,16085------,1)
-- Executing [16085------@from-internal:1] Set("SIP/1024-00000003", "CHANNEL(accountcode)=Google Voice") in new stack
-- Executing [1608------@from-internal:2] Dial("SIP/1024-00000003", "PJSIP/16085------@gvsip,,r") in new stack
[2018-07-04 18:53:17] ERROR[3286]: chan_pjsip.c:2215 request: Unable to create PJSIP channel - endpoint 'gvsip' was not found
[2018-07-04 18:53:17] WARNING[3285][C-00000002]: app_dial.c:2527 dial_exec_full: Unable to create channel of type 'PJSIP' (cause 3 - No route to destination)
== Everyone is busy/congested at this time (1:0/0/1)
-- Executing [16085------@from-internal:3] Progress("SIP/1024-00000003", "") in new stack
-- Executing [16085------@from-internal:4] Wait("SIP/1024-00000003", "1") in new stack
> 0xb6324220 -- Strict RTP switching to RTP target address 192.168.2.21:5004 as source
-- Executing [16085------@from-internal:5] Playback("SIP/1024-00000003", "silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer") in new stack
-- Playing 'silence/1.ulaw' (language 'en')
-- Playing 'cannot-complete-as-dialed.ulaw' (language 'en')
-- Playing 'check-number-dial-again.ulaw' (language 'en')


By the way, I'm running this on a pi B+. If this is only supposed to work on a pi 3, please let me know.

 

[Waffull]

I spoke too soon... Made a call this evening. After it rang 4 or 5 times, I hear a message that the call cannot be completed as dialed. This is what appeared is CLI for the whole call:

Connected to Asterisk 13.21.1 currently running on incrediblepbx (pid = 1780)
[2018-07-04 21:54:38] DEBUG[1909]: res_pjsip_outbound_registration.c:2374 handle_outgoing_request: Found matching outbound registration state
[2018-07-04 21:54:38] DEBUG[1909]: res_pjsip_outbound_registration.c:2385 handle_outgoing_request: Found service-route. Adding route header for <sip:AAYAYVANRNB2TDETNLSTHLN2NUHXL6VE4BMBKXMTS2PG7JGYPOKWMWTWLFVZD4L:5060;uri-econt=IUP534PMNDZSKTMTOH2FSTPONLQLDEHS4ZUOYXTHWMLTCCZ2QRYER33QEZ7JYNQ4XXVL3XS4324HCHWG4VZYINOWXLEOOTFVYPG3LDJY3IALJCWO2ZKEI3G5V5PMC5LKYFCHNR4KZ37J7J6X4BCJU3M4ZYPPHVSQ;lr>
[2018-07-04 21:54:38] DEBUG[1909]: res_pjsip_outbound_registration.c:2385 handle_outgoing_request: Found service-route. Adding route header for <sip:AA65NWTR7IEJJAY5A3EUBEFZYWVX6OD5YQQYE4UO3DPDCBICLSN4HT7MOBVVS24:5060;transport=udp;lr;uri-econt=3BSNJYK6L>
[2018-07-04 21:54:38] DEBUG[1909]: res_pjsip_outbound_registration.c:2412 handle_outgoing_request: Re-using outbound registration address of 64.9.243.108:5061
[2018-07-04 21:54:38] ERROR[1909]: pjproject:0 <?>: sip_msg ...Header with no vptr encountered!! Current buffer: INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.2.154:5061;rport;branch=z9hG4bKPje2cbe9e0-491f-4ae3-abf1-e01df0c16360;alias
From: "701" <sip:[email protected]>;tag=c6ff1637-53d7-4330-b6d4-aa1cfdd5b0a2
To: <sip:[email protected]>
Contact: <sip:[email protected]:5061;transport=TLS>
Call-ID: 02d687a7-46ab-4daf-ae4a-0a1a9b457ec5
CSeq: 168 INVITE

[2018-07-04 21:54:39] ERROR[1909]: pjproject:0 <?>: icess0x6dee167 ......Error sending STUN request: Network is unreachable
[2018-07-04 21:54:39] ERROR[1926]: pjproject:0 <?>: icess0x6dee167 ..Error sending STUN request: Network is unreachable
[2018-07-04 21:55:01] ERROR[1909]: pjproject:0 <?>: sip_msg ..Header with no vptr encountered!! Current buffer: ACK sip:[email protected] SIP/2.0
Via: SIP/2.0/TLS 192.168.2.154:5061;rport;branch=z9hG4bKPje2cbe9e0-491f-4ae3-abf1-e01df0c16360;alias
From: "701" <sip:[email protected]>;tag=c6ff1637-53d7-4330-b6d4-aa1cfdd5b0a2
To: <sip:[email protected]>;tag=1459650947
Call-ID: 02d687a7-46ab-4daf-ae4a-0a1a9b457ec5
CSeq: 168 ACK

Called # changed to x's for privacy reasons.

Thanks for the reply. I'll see if I can migrate my settings from RASPBX over to IncrediblePBX.

That said, I now have IncrediblePBX up and running and ran the naf-raspi script again.

Incoming calls seem to be working, but I do get the following error with the inbound calls:


Outbound calls produce the following errors:


Both inbound and outgoing calls are currently working, despite the errors.

 

[mrfeh]

Please see the post from naf on this page in response to my gvsip problems with IncrediblePBX:

http://www.dslreports.com/forum/r32028869-~time=1530782000

It appears as though the gvsip patches are not included in the download indicated on this page: http://nerdvittles.com/?p=26267

 

[Seattle]

I can confirm the the new gvsip works on Raspberry Pi B as well. It shows up as Obitalk devices on the Legacy google voice page. Google voice incoming call rings Incredible PBX IVR first and I could not find a way to turn it off even I setup to ring my extension (please let me know if you know how). Work around is to uncheck or turn off Obitalk on google voice and forward to my callcentric number, which registers on my Obi110.

 

[wardmundy]

I can confirm the the new gvsip works on Raspberry Pi B as well. It shows up as Obitalk devices on the Legacy google voice page. Google voice incoming call rings Incredible PBX IVR first and I could not find a way to turn it off even I setup to ring my extension (please let me know if you know how). Work around is to uncheck or turn off Obitalk on google voice and forward to my callcentric number, which registers on my Obi110.

Have you read the tutorial?? It explains how to redirect incoming calls to anywhere you like:

To modify the Google Voice behavior for incoming calls, jump to the bottom of extensions_custom.conf. There you’ll find the [from-external-custom] context which controls the routing and processing of incoming calls to your Google Voice trunk. Several examples are provided. By default, the inbound calls are routed to the Demo IVR (3366). If your PBX has a Ring Group 777 and you’d prefer to send the calls there, simply change 3366 to 777. If you would prefer to send the calls to an extension, then comment out the Demo IVR line with a semicolon and uncomment the SIP/701 line while also replacing 701 with the extension desired. If you’d prefer to send incoming calls to a specific Asterisk application, an example is provided to route the calls to the NV Weather ZIP application. Then reload your Asterisk dialplan by issuing the following command: asterisk -rx "dialplan reload"

 

[mrfeh]

Please see the post from naf on this page in response to my gvsip problems with IncrediblePBX:

http://www.dslreports.com/forum/r32028869-~time=1530782000

It appears as though the gvsip patches are not included in the download indicated on this page: http://nerdvittles.com/?p=26267

Well, I started from scratch with a branch new IncrediblePBX installation, and I now have things working. Yay!

One small problem, if somebody could help - on outgoing calls, if the call is not answered in 20 seconds, I get the "your call cannot be completed as dialed" message. This is after the phone rings for 20 seconds.

Can somebody explain how to disable this apparent timer on outgoing calls? Thanks.

 

[wardmundy]

Well, I started from scratch with a branch new IncrediblePBX installation, and I now have things working. Yay!

One small problem, if somebody could help - on outgoing calls, if the call is not answered in 20 seconds, I get the "your call cannot be completed as dialed" message. This is after the phone rings for 20 seconds.

Can somebody explain how to disable this apparent timer on outgoing calls? Thanks.

You posted the same question on the DSLR Forum. Did you not like the answer there??

 

[mrfeh]

You posted the same question on the DSLR Forum. Did you not like the answer there??

Didn't know it had been responded to, but now that I've read the response, no, it does not provide a solution.

Do you know of one?

 

[Waffull]

Didn't know it had been responded to, but now that I've read the response, no, it does not provide a solution.

Do you know of one?

Are you sure, it worked for me. The wiki raf pointed me to in his answer also gives an explanation.

Make sure you reloaded the dialplan after you made the change.

 

[mrfeh]

Are you sure, it worked for me. The wiki raf pointed me to in his answer also gives an explanation.

Make sure you reloaded the dialplan after you made the change.

Could you please provide an example of what "remove the r from the dial command" means? Is that an edit to extensions_custom.conf, or something in the FreePBX interface?

I tried removing the ',,r' from the Dial command in extensions_custom.conf, and while that prevented the call from being taken down after 20 seconds, I no longer hear the destination phone ringing from the source phone; just silence until the voicemail message from the destination phone is heard.

Thanks.

 

[billsimon]

Could you please provide an example of what "remove the r from the dial command" means? Is that an edit to extensions_custom.conf, or something in the FreePBX interface?

I tried removing the ',,r' from the Dial command in extensions_custom.conf, and while that prevented the call from being taken down after 20 seconds, I no longer hear the destination phone ringing from the source phone; just silence until the voicemail message from the destination phone is heard.

Thanks.

Yes, you get one or the other.

PS: this was discussed fully on the other thread.