@comtech wrote:
SNG 7
FreePBX 14/Asterisk 13Hello,
I am hoping to build a custom context where I collect some variables (done), write these variables to a text file (done) and then send the text file contents to a physical fax machine for an end user to consume.
I know I can use the faxsend dialplan command to actually send the fax, but how do I get the document “fax ready” (from .txt to .tiff?)?
Does anyone have any pointers or ideas on how this might be accomplished?
Posts: 2
Participants: 1
@fearx wrote:
When I enter the GUI, it displays the message “Can not connect to Asterisk”. When I start the asterisk manually, the GUI stops responding. How can I solve this? PLEASE ITS URGENT!
Edit: After a while, FreePBX starts the Asterisk, and then, GUI crashs again…
Posts: 4
Participants: 2
@josephchrz wrote:
Hello i have small problem. me and my friend share a freepbx server. He has a grandstream fxo analog pots box with 4 lines on it all in a hunt group. and i only have one analog line with a pci fxo card from samgoma. Problem is when my line is tied up and in use no one else can call me. my friend said i can use one of his lines Sense he doesn’t use them all. Problem is how can i port one of his lines to mine so if my line is tied up it can go to that second line and ring in? Can someone please help me. Thank you.
Posts: 1
Participants: 1
@rnmixon wrote:
We are running FreePBX distro "10.13.66-22 ", connected to a Digium SIP trunk.
Twice recently (middle of May and today) we’ve had a larger number of calls (193 in May and 32 today) that rang in the office - they would appear to hang up when someone answered or when they answer no one is on the line.
The CDR records show the individual calls for each time the phones rang - looks like they would choose an option on our IVR and route to a ring group. Most of the calls show a duration of 20 seconds or less, but 8 show a duration of just less than 6 minutes and one has a duration of 9:02 minutes. All are proportionally spaced across the overall duration - 8:23:06 to 9:20:05.
But when I extract all the /var/log/asterisk/full records for the first calls call id (all with e.g. “[C-00002520]” as the third field) it turns out it’s one long call.
I opened a ticket with Digium before but they were not able to shed much light on what was happening. I’ve opened a new ticket for today’s event but thought maybe it’s not specifically related to the Digium SIP Trunk.
Our main concern would be that a hacker was trying to route calls back out our trunk somehow - but looking at the CDRs this does not appear to be the case.
Before I start listing logs and other detail - does this pattern match a know problem with configuration or other issue?
If not what info is needed to comment further?
Thank you much for any insight - Richard
Posts: 1
Participants: 1
@gbrose wrote:
FreePBX 14 on Centos 7.5
Grandstream GXP2000I have two inbound routes and two extensions (200, 400) supporting two phone numbers. Both are configured essentially identically, but only one can receive calls. The other (200) fails immediately with
ERROR[14481] res_pjsip.c: Unable to retrieve PJSIP transport ‘udp,tcp,ws,wss’
This is the only log message when the call fails.
I can call out on both lines with no problem – when calling my cell phone, the caller ID on shows the correct originating phone number.
Log of registration messages:
VERBOSE[726] res_pjsip_registrar.c: Added contact ‘sip:200@192.168.71.5:5062’ to AOR ‘200’ with expiration of 3600 seconds
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 200/sip:200@192.168.71.5:5062 has been created
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Endpoint 200 is now Reachable
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 200/sip:200@192.168.71.5:5062 is now Reachable. RTT: 43.316 msec
VERBOSE[726] res_pjsip_registrar.c: Removed contact ‘sip:400@192.168.71.5:5064’ from AOR ‘400’ due to request
VERBOSE[9906] res_pjsip/pjsip_configuration.c: Contact 400/sip:400@192.168.71.5:5064 has been deleted
VERBOSE[726] res_pjsip_registrar.c: Added contact ‘sip:400@192.168.71.5:5064’ to AOR ‘400’ with expiration of 600 seconds
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 400/sip:400@192.168.71.5:5064 has been created
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Endpoint 400 is now Reachable
VERBOSE[20580] res_pjsip/pjsip_configuration.c: Contact 400/sip:400@192.168.71.5:5064 is now Reachable. RTT: 42.140 msecThe
pjsip.endpoint.conf:[400]
type=endpoint
aors=400
auth=400-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=g729,ulaw,alaw,g726,g723,g722,g719
context=from-internal
callerid=Home <400>
dtmf_mode=rfc4733
mailboxes=400@device
mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=no
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en[200]
type=endpoint
aors=200
auth=200-auth
tos_audio=ef
tos_video=af41
cos_audio=5
cos_video=4
allow=g729,ulaw,alaw,g726,g723,g722,g719
context=from-internal
callerid=Office <200>
dtmf_mode=rfc4733
mailboxes=200@device
mwi_subscribe_replaces_unsolicited=yes
aggregate_mwi=yes
use_avpf=no
rtcp_mux=no
ice_support=no
media_use_received_transport=no
trust_id_inbound=yes
media_encryption=no
timers=yes
media_encryption_optimistic=no
send_pai=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes
language=en[anonymous]
type=endpoint
context=from-sip-external
allow=all
transport=udp,tcp,ws,wss[dpma_endpoint]
type=endpoint
context=dpma-invalidI have deleted and recreated the OSS Endpoint configuration.
I have deleted and recreated the extension.Any idea what the problem is/what else to try.?
Thanks.
Posts: 1
Participants: 1
@sektek wrote:
I have a FreePBX distro that the call history in UCP for all user’s extensions are showing calls from August 29th, 2017 and older. Nothing new.
I have completely removed the UCP module, and reinstalled it. All other modules and RPMs are up-to-date. PBX Version 14.0.3.6 and System: 12.7.5-1805-2.sng7
Any ideas on how to fix?
Posts: 1
Participants: 1
@Logistry wrote:
connectivity - firewall - enable firewall, got the below error
Exception thrown with message “SQLSTATE[HY000] [2002] Connection refused::SQLSTATE[HY000] [2002] Connection refused”
Stacktrace: #7 Exception in /var/www/html/admin/libraries/utility.functions.php:204 #6 die_freepbx in /var/www/html/admin/libraries/BMO/Database.class.php:142 #5 PDOException in /var/www/html/admin/libraries/BMO/Database.class.php:137 #4 PDO:__construct in /var/www/html/admin/libraries/BMO/Database.class.php:137 #3 FreePBX\Database:__construct in /var/www/html/admin/libraries/BMO/FreePBX.class.php:71 #2 FreePBX:__construct in /var/www/html/admin/bootstrap.php:153 #1 require_once in /etc/freepbx.conf:11 #0 include_once in /var/www/html/admin/config.php:100see attached full pictures of the error.
PBX Firmware 12.7.5-1805-3.sng7
Firewall 13.0.56
FreePBX GUI 14.0.3.6If anyone with more knowledge than me who can please take look at these errors and tell me whats wrong here.
Is this a bug?Appreciate your responses
Cheers!
Posts: 1
Participants: 1
@kspare wrote:
I have a customer coming on with us from an old Key system.
They had a button that allow them to basically dial ##* and wait for them to dial the extension.
Effectively sending a call directly to someones voicemail, but they only have to type in their ext.
I’m trying to make this work on a polycom vvx411 and FPBX 14.
I’m ok with telling them they will just have to put ##* but thought i’d ask if there is a way to make this a macro.
Posts: 2
Participants: 2
@aeciolemos wrote:
Hi everyone,
I have run into a problem today and I cannot seem to find the solution. After I ran a yum update and updated all the modules, I get an error trying to Apply Config:Reload failed because retrieve_conf encountered an error: 1
exit: 1
Unable to continue. Only variables should be assigned by reference in /var/www/html/admin/modules/core/functions.inc.php on line 5454
#0 /var/www/html/admin/modules/core/functions.inc.php(5454): Whoops\Run->handleError(2048, ‘Only variables …’, ‘/var/www/html/a…’, 5454, Array)
#1 /var/www/html/admin/modules/core/functions.inc.php(2239): general_generate_indications()
#2 /var/www/html/admin/libraries/BMO/DialplanHooks.class.php(95): core_do_get_config(‘asterisk’)
#3 /var/lib/asterisk/bin/retrieve_conf(860): FreePBX\DialplanHooks->processHooks(‘asterisk’, Array)
#4 {main}I am running
PBX Firmware:12.7.5-1805-3.sng7
FreePBX 14.0.3.6
This server was installed from the distroBefore this error in the CORE module, I got the same error for DAHDI Config module and Phone Apps module.
I removed those modules but I can’t remove the CORE, obviously. I tried reinstalling PHP as well.
Any help will be greatly appreciated. Let me know if I failed to provide important information.
Thank you
Posts: 1
Participants: 1
@jasonp wrote:
I am wanting to put 1 extension on a trunk from a different provider to test and see if it resolves some issues we are having that i believe are from our provider. If I set up an outbound route using the extension number in the route CID, will that limit it to the one extension?
Posts: 2
Participants: 2
@tomilin wrote:
HI,
Please help me to solve error.
Upgraded all the modules to the latest. However, now on the " System Status Page", the following error is being shown…
Cronmanager encountered 1 Errors
The following commands failed with the listed error
/var/lib/asterisk/bin/clean_calltracking.php (127)[root@sip ~]# cat /var/lib/asterisk/bin/clean_calltracking.php
cat: /var/lib/asterisk/bin/clean_calltracking.php: No such file or directoryFreePBX 14.0.3.6
0: Linux sip 3.10.0-693.17.1.el7.x86_64 #1 SMP Thu Jan 25 20:13:58 UTC 2018 x86_64 x86_64 x86_64 GNU/Linux
Posts: 1
Participants: 1
@ownroom000 wrote:
Hey,guys. I set my freepbx service in a small orange pi (192.168.10.129). Everything running fine on Intranet (two client: PC-192.168.10.210 and phone both can hear each other on pjsip mode). However the special situation is, my router can’t get a external ip. The wan ip was like 10.x.x.x or 100.x.x.x. So i used a NAT traverse software called frp-(ttps://github.com/fatedier/frp),basically you can use this software to forward any port to a vps which with a static ip. so successfully forward 5060,10001-10041,80,22 to my vps. And then i changed the sip address to my vps’s ip (both phone and computer). and i used my mobile phone (cellular data) to call my PC (wifi connection), ringing is okay but no voice heard. The strange thing is when i connect my phone to wifi everything works fine I can hear the voice. after that i used wireshark to catch the packet. no audio packet- (ttps://drive.google.com/open?id=1nJ105lmQfHMH4j2zHWGY-LrRqHCDeZVp) /// works fine packet-(ttps://drive.google.com/open?id=13txmSD9vhHAx_7pdjlIn7fH3n4VTbj3f) Can you help me solve this problem? or give me a hint whats the possible issue?
Posts: 1
Participants: 1
@shivani wrote:
Hello,
We recently migrated from Freepbx 6.12.65 to 10.13.66. SIP phone calls work fine. However FAX machines are not able to send FAX anymore. I did captures on the old version and the new version and when I compare the packet captures on the two versions the only difference I see is that version 6.12.65 was sending FAX machines IP address in the SDP to the SIP provider. However with the new version 10.13.66 the server is sending its own NATted public IP address as the “connection ID” in the SDP packets for media. I believe this is the reason why its not working. The FAX call connects but then after some time it drops (Freepbx sends a bye). Does anyone know how to make Freepbx not send its own IP address as the connection ID? I checked the SIP server and extension settings and the “can reinvite” option is set to yes.
I have also attached a picture of wireshark showing the comparison of the two versions packet capture. Yellow one is the one working and it sends the FAX machine’s IP address as the connection Information. Red one is the one not working and it sends Freepbx NAT IP in the connection information. I believe this is the issue as this is the only difference I see between the two.
Please let me know how to troubleshoot or resolve this issue.
Thank you.
Shivani
Posts: 1
Participants: 1
@charchar wrote:
Newbie to FreePBX and would appreciate some help.
I am trying to have multiple extensions rings at once when someone calls the department. For example: If someone calls department A, extensions 323, 322 and 321 will ring at once.
I’m thinking a ring group would be the best option here but am not entirely sure.
Thanks for the help!
Posts: 1
Participants: 1
@drummerjoe wrote:
We’ve implemented a few FreePBX servers, but have never run into this issue before. When an extension has Follow Me disabled, there is no problem calling that extension from another extension. However, when Follow Me is enabled, we get a busy signal and the logs show this:
[2018-07-06 15:31:14] WARNING[6562][C-00000003] pbx.c: Channel ‘SIP/1-00000003’ sent to invalid extension but no invalid handler: context,exten,priority=followme-check,2,1
Everything is set to default - this is a brand new installation of FreePBX on the latest distro (12.7.5-1805-3.sng7).
The follow me list shows the extension, and then an external number (followed by #). I’ve tested removing the external number and just having the extension, and I’ve tested without the extension and just the external number. All result in the same log entry as shown above.
Posts: 4
Participants: 3
@comtech wrote:
SNG7
FreePBX14/Asterisk13I am using customer inputs from an inbound call to generate a .txt file to send to another group via fax.
•Caller input captured in channel variables
•Channel variables written to ${UNIQUEID}.txt file
•${UNIQUEID}.txt file converted to ${UNIQUEID}.ps file (enscript)
https://www.gnu.org/software/enscript/•${UNIQUEID}.ps file converted to ${UNIQUEID}.tiff file (ghostscript)
https://www.gnu.org/software/ghostscript/•${UNIQUEID}.tiff file sent to fax machine using Faxsend.
This all works, but it looks really plan, just the text from the original .txt file. We were hoping to be able to add a logo to the top right to make the fax appear more professional.
Does anyone have any ideas on how we might be able to make this happen? Thanks for your help!
Posts: 1
Participants: 1
@slockner wrote:
i am working on trying to get GVSIP going… but i am stuck having issues even getting PJSIP installed even as as part of a standard build. using the guild located here… https://wiki.freepbx.org/display/FOP/Installing+FreePBX+14+on+CentOS+7
i am not seeing any errors displayed as i walk thru tall the steps.
howerver when ever i log in to asterisk and try a command like pjsip show endpoints it says no such command… however when i run the same command in my production freepbx the commands seem to work.
there is nothing about PjSIP in /etc/asterisk/modules.conf
i used CentOS-7-x86_64-Minimal-1804.iso that i just downloaded the other day.
i have probably attempted building this 20 times…
any help would be great… probably somethings simple i am over looking.
looks like it installed Asterisk 14.5.0 the last time i tried
Posts: 1
Participants: 1
@hiastar_alex wrote:
Hi all:
My customer wants to connect Sangoma software SBC with 3cx, then connect IMS 100rel with Sangoma SBC.
Can you succeed?Topology:
Posts: 1
Participants: 1
@Shawn0228 wrote:
Hey!
I upgraded the VPS that we are on to give us 100 GB of storage, vs. the 60 we were at previously. After upgrading the VPS, I booted an Ubuntu LiveCD, opened GParted, expanded the partition to fill the space, rebooted back into the FreePBX environment, and went from there.
Looking at system admin, it is reporting that I am still at 60GB total of storage, as is “df” (system admin “storage” screenshot attached)
Filesystem 1K-blocks Used Available Use% Mounted on /dev/mapper/SangomaVG-root 56767404 41513932 15253472 74% / devtmpfs 3992336 0 3992336 0% /dev tmpfs 4004716 0 4004716 0% /dev/shm tmpfs 4004716 8848 3995868 1% /run tmpfs 4004716 0 4004716 0% /sys/fs/cgroup /dev/vda1 1983056 201304 1662968 11% /boot tmpfs 800944 0 800944 0% /run/user/995 tmpfs 800944 0 800944 0% /run/user/0I feel like I’m missing a step, I just don’t know what. If needed, I can go back and provide a GParted GUI output to see if that helps at all.
Posts: 2
Participants: 2
@domosute wrote:
Hi,
I have a similar situation as in below thread, and I observe receiving “sorry the call cannot be completed” male voice message whenever SIP trunk vendor reply back with SIP 486 status.
I learned the message is played from far end call server whenever far-end line is filled up so it is not the issue of neither our PBX nor our SIP trunk provider. But the played message is not so clear to our internal user why the phone is terminated after the message.
So I am thinking somehow to override the played message to custom message so that our internal user understand the cause of issue is on far end, not our phone system. (instead of “sorry the cannot be completed” male voice, “Far end line is busy and can not reach out at this time”, etc.)
Any comment / advice would be highly appreciated!
Many thanks in advance,
Platform:
FreePBX: 13.0.195.1
Asterisk: 13.19.2
Posts: 3
Participants: 3
