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1 votes
1 answers
39 views
dpkg-shlibdeps: missing symbols in libasteriskpj.so.2 [asterisk]
I’m trying to build Asterisk packages for Ubuntu 22.04 LTS, and I’m encountering the same issue with both version 22.4.x and 22.5.0. During the build process, I get the following warnings: dpkg-shlibdeps: warning: symbol __ast_free used by debian/asterisk/usr/lib/x86_64-linux-gnu/libasteriskpj.so.2...
I’m trying to build Asterisk packages for Ubuntu 22.04 LTS, and I’m encountering the same issue with both version 22.4.x and 22.5.0. During the build process, I get the following warnings: dpkg-shlibdeps: warning: symbol __ast_free used by debian/asterisk/usr/lib/x86_64-linux-gnu/libasteriskpj.so.2 found in none of the libraries dpkg-shlibdeps: warning: symbol ast_pjproject_max_log_level used by debian/asterisk/usr/lib/x86_64-linux-gnu/libasteriskpj.so.2 found in none of the libraries dpkg-shlibdeps: warning: symbol __ast_repl_malloc used by debian/asterisk/usr/lib/x86_64-linux-gnu/libasteriskpj.so.2 found in none of the libraries dpkg-shlibdeps: warning: symbol ast_option_pjproject_log_level used by debian/asterisk/usr/lib/x86_64-linux-gnu/libasteriskpj.so.2 found in none of the libraries The .deb package builds successfully, but I'm not sure if the library will function correctly. I did find these symbols in the Asterisk binary. For simplicity, I’m building a single “all-in-one” package, as this makes local deployment easier for me.
Qmails (13 rep)
Jul 18, 2025, 11:54 AM • Last activity: Jul 18, 2025, 12:16 PM
1 votes
2 answers
1514 views
Issue in Asterisk Performance
I am running FreePBX on DigitalOcean VM. I am facing some call dropping issue in Asterisk. When looking at "htop", "top" and FreePBX GUI, I see different results of CPU utilization and can't understand if my VM is OK or needs attention w.r.t CPU. Load average shows nearly idle system while CPU utili...
I am running FreePBX on DigitalOcean VM. I am facing some call dropping issue in Asterisk. When looking at "htop", "top" and FreePBX GUI, I see different results of CPU utilization and can't understand if my VM is OK or needs attention w.r.t CPU. Load average shows nearly idle system while CPU utilization of Asterisk process is pretty high. enter image description here enter image description here enter image description here Could someone please advise me on this? 74% CPU on a process yet idle system on load average. Can this be an issue causing call dropping on Asterisk?
Jahanzeb Ali (33 rep)
Aug 26, 2014, 07:13 AM • Last activity: May 21, 2025, 01:37 PM
0 votes
1 answers
2739 views
Asterisk v13 on Kali Linux: No RTP engine was found. Do you have one loaded?
I installed Asterisk and tried to make a call with zoiper but I get an error [call failure 401 forbidden] and Asterisk return this message : Connected to Asterisk UNKNOWN__and_probably_unsupported currently running on kali (pid = 14877) [Jan 22 17:57:00] ERROR[14937][C-00000007]: rtp_engine.c:401 as...
I installed Asterisk and tried to make a call with zoiper but I get an error [call failure 401 forbidden] and Asterisk return this message : Connected to Asterisk UNKNOWN__and_probably_unsupported currently running on kali (pid = 14877) [Jan 22 17:57:00] ERROR[C-00000007]: rtp_engine.c:401 ast_rtp_instance_new: No RTP engine was found. Do you have one loaded? [Jan 22 17:57:00] NOTICE[C-00000007]: chan_sip.c:25550 handle_request_invite: Failed to authenticate device ;tag=9a473a54 I opened the menuselect and selected res_rtp_asterisk. When I try to reinstall (recompile!) Asterisk, I do ./configure This says it's ok! But when I put make or make install, I get this error: CC="cc" CXX="g++" LD="" AR="" RANLIB="" CFLAGS="" LDFLAGS="" make -C menuselect CONFIGURE_SILENT="--silent" makeopts make: Entering directory `/etc/asterisk/asterisk/menuselect' make: `makeopts' is up to date. make: Leaving directory `/etc/asterisk/asterisk/menuselect' [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c: In function ‘ice_create’: res_rtp_asterisk.c:2421:4: error: too many arguments to function ‘pj_ice_sess_create’ In file included from /usr/include/pjnath.h:23:0, from res_rtp_asterisk.c:53: /usr/include/pjnath/ice_session.h:736:22: note: declared here make: *** [res_rtp_asterisk.o] Error 1 make: *** [res] Error
Y. Dabbous (67 rep)
Jan 23, 2016, 12:55 PM • Last activity: May 13, 2025, 01:07 PM
2 votes
3 answers
3909 views
Debian increase ulimit for Asterisk
I've been facing an issue with Asterisk 13.11.2 on Debian 8 where it's crashing after reaching the limit of open files bridge_channel.c: Can't create pipe! Try increasing max file descriptors with ulimit -n I have managed to increase the limit from 65536 to 150000 using the `/etc/security/limits.con...
I've been facing an issue with Asterisk 13.11.2 on Debian 8 where it's crashing after reaching the limit of open files bridge_channel.c: Can't create pipe! Try increasing max file descriptors with ulimit -n I have managed to increase the limit from 65536 to 150000 using the /etc/security/limits.conf I have added the following: root soft nofile 150000 root hard nofile 150000 * soft nofile 150000 * hard nofile 150000 The result of ulimit -n is now 150000 When i try check the limit for the Asterisk process cat /proc/xxx/limits I still get the old limit! Limit Soft Limit Hard Limit Units Max cpu time unlimited unlimited seconds Max file size unlimited unlimited bytes Max data size unlimited unlimited bytes Max stack size 8388608 unlimited bytes Max core file size 0 unlimited bytes Max resident set unlimited unlimited bytes Max processes 31945 31945 processes Max open files 1024 4096 files Max locked memory 65536 65536 bytes Max address space unlimited unlimited bytes Max file locks unlimited unlimited locks Max pending signals 31945 31945 signals Max msgqueue size 819200 819200 bytes Max nice priority 0 0 Max realtime priority 0 0 Max realtime timeout unlimited unlimited us How to solve this?
TareKhoury (121 rep)
Oct 4, 2016, 08:54 AM • Last activity: Apr 9, 2025, 01:09 PM
0 votes
0 answers
17 views
Why i get this message in Statistic in Operators and Queue in WellTime that works with Asterisk
Please, help. I get this message in WellTime in Operators and Queue [![enter image description here][1]][1] [![enter image description here][2]][2] And this is the error message from first picture: mysql error: [1054: Unknown column 'from' in 'having clause'] in EXECUTE("select count(*) as count fro...
Please, help. I get this message in WellTime in Operators and Queue enter image description here enter image description here And this is the error message from first picture: mysql error: [1054: Unknown column 'from' in 'having clause'] in EXECUTE("select count(*) as count from( SELECT if(complete_cause='ABANDON' || complete_cause='EXITWITHTIMEOUT' || complete_cause='EXITEMPTY','@lost_calls',if( t1.agent='',concat('@',ext),t1.agent)) as agent_res, sum(1) as count_all, sum(if(t1.call_type='in',1,0)) as count_in, sum(if(t1.call_type in ('out', 'out_qcb'),1,0)) as count_out, sum(if(t1.call_type='ext',1,0)) as count_ext, sum(if(t1.call_type='in',to_complete,0)) as time_in, sum(if(t1.call_type in ('out', 'out_qcb'),to_complete,0)) as time_out, sum(if(t1.call_type='ext',to_complete,0)) as time_ext, sum(to_complete) as time_total, max(if(t1.call_type='in', 0, to_answer)) as max_answer FROM call t1 WHERE complete_cause != 'ABANDON' and complete_cause != 'EXITWITHTIMEOUT' and complete_cause != 'EXITEMPTY' AND t1.agent != '' AND (utime between 1730667600 and 1731618000 ) AND ( (1=2 OR ( t1.ext = '9165682305' ) OR ( t1.agent = '9165682305' ) OR ( t1.subscriber = '9165682305' ) ) ) and (1=1) GROUP BY agent_res HAVING 1=1 AND ( from LIKE ('%9165682305%') ) ) as result") Maybe it's because i redacted the asterisk extensions_custome.conf, here is the code from there [ivr-25-custom] exten => h,1,Noop(${CALLERID(dnid)}) exten => h,n,gotoif($["${CDR(dstchannel)}"=""]?:noop) exten => h,n,System(/opt/ast/conv.sh info@ast.ru "${CDR(src)}" "${CALLERID(name)}" "${DIALSTATUS}" "${FILE}" 2 "${CALLERID(dnid)}") exten => h,n(noop),Noop(bye) [ivr-105-custom] exten => h,1,Noop(${CALLERID(dnid)}) exten => h,n,gotoif($["${CDR(dstchannel)}"=""]?:noop) exten => h,n,System(/opt/ast/conv.sh info@ast.ru "${CDR(src)}" "${CALLERID(name)}" "${DIALSTATUS}" "${FILE}" 2 "${CALLERID(dnid)}") exten => h,n(noop),Noop(bye) [ivr-112-custom] exten => h,1,Noop(${CALLERID(dnid)}) exten => h,n,gotoif($["${CDR(dstchannel)}"=""]?:noop) exten => h,n,System(/opt/ast/conv.sh info@ast.ru "${CDR(src)}" "${CALLERID(name)}" "${DIALSTATUS}" "${FILE}" 2 "${CALLERID(dnid)}") exten => h,n(noop),Noop(bye) I really don't know why i get this message. I mean why this message is pop up. I don't do nothing in mysql and boom, this happens. Help me, please.
Илья Медведев (1 rep)
Nov 14, 2024, 04:08 PM
0 votes
1 answers
90 views
Asterisk pbx agi php script with imagick imaging on the fly not working
### Problem description: Generate an image with an ASTERISK AGI PHP script that is displayed on a phone (Yealnk) using `imagick`. ``` installed:php-imagick/oldstable,now 3.4.4+php8.0+3.4. ``` The AGI PHP script consists of two parts: 1 Generating an image containing caller information, such as telep...
### Problem description: Generate an image with an ASTERISK AGI PHP script that is displayed on a phone (Yealnk) using imagick.
installed:php-imagick/oldstable,now 3.4.4+php8.0+3.4.
The AGI PHP script consists of two parts: 1 Generating an image containing caller information, such as telephone number and a corresponding passport photo based on the telephone number. 2 Send this image to the phone. ### What works in the PHP script? 2: Send an existing image to the phone. If I use an existing image, it is sent to the phone and displayed without any problems. ### What doesn't work in the PHP script: 1: No image is generated with imagick. In the Asterisk CLI I see that the script is executed with a 0, so no error messages. ---- If I start the script as root in /usr/share/asterisk/agi-bin then everything works fine. Imagick generates an image and it is neatly delivered to the phone:
php script.php 01234567
But as soon as the script calls imagick it stops working, like this: $foto = new Imagick("/path/to/image/image.png"); ### What have I tried: 1 Used other image generators: gdimage and gmagick, but they have the same problem. 2 chown 777 script.php 3 Check whether Asterisk is passing the correct variables (that works fine) with the AGI command 4 Start PHP script in the CLI as another user (su otheruser) imagick does not work either. Can anyone point me in the right direction as to why imagick isn't generating images from an AGI in Asterisk? UPDATE: Ok, I've made some progress. I suspect it is a read and write permission case. When I write to /tmp, the script does not stop and indeed an image is written to /tmp. Reading that picture still causes problems. I continue searching.
Edderpet (1 rep)
Sep 13, 2023, 08:49 AM • Last activity: Sep 13, 2023, 01:52 PM
2 votes
2 answers
7226 views
Spam Messages cb crond[1288]: No configuration file found at /root/.esmtprc or /etc/esmtprc
Centos 8, fresh install with asterisk on it all log drains by one message 20-100 per sec ```cb crond[1288]: No configuration file found at /root/.esmtprc or /etc/esmtprc```
Centos 8, fresh install with asterisk on it all log drains by one message 20-100 per sec
crond: No configuration file found at /root/.esmtprc or /etc/esmtprc
Stremovskyy (611 rep)
Jul 29, 2022, 12:24 PM • Last activity: Dec 8, 2022, 03:26 PM
0 votes
1 answers
479 views
asterisk debug extension bad quality audio - rtp out of sequence
FreePBX 16.0.10.34, or direct Asterisk 11/16/18.6.0 is the same, all of those with some SIP phones got weird audio issues, i.e. voice volume constantly change in one way, metallic voice in the other way ecc. Audio Codec: G711 or G729 Dial echo test > record pcap Wireshark > Voip > show call graph an...
FreePBX 16.0.10.34, or direct Asterisk 11/16/18.6.0 is the same, all of those with some SIP phones got weird audio issues, i.e. voice volume constantly change in one way, metallic voice in the other way ecc. Audio Codec: G711 or G729 Dial echo test > record pcap Wireshark > Voip > show call graph and get out of sequence notation ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT.... =========================================================================================================== 105-0000013e 00:00:25 g729 1258 589K 46884 0.000 1242 0 0 0.003 0.006 Phone makers just take packet log and disappear, how can I debug those issues? Tried to force codec G711 with ptime 20, added to echo test context the JITTERBUFFER : [app-echo-test] include => app-echo-test-custom exten => *43,1,Set(CONNECTEDLINE(name-charset,i)=utf8) exten => *43,n,Set(CONNECTEDLINE(name,i)=Test Eco) exten => *43,n,Set(CONNECTEDLINE(num,i)=*43) exten => *43,n,Answer exten => *43,n,Set(JITTERBUFFER(adaptive)=default) exten => *43,n,Macro(user-callerid,) exten => *43,n,Wait(1) exten => *43,n,Background(demo-echotest,,,app-echo-test-echo) exten => *43,n,Goto(app-echo-test-echo,1,1) ;--== end of [app-echo-test] ==--; It seems that it is used but got no results: sing SIP RTP Video TOS bits 136 in TCLASS field. == Using SIP RTP Video CoS mark 4 -- Executing [*43@from-internal:1] Set("PJSIP/101-00000143", "CONNECTEDLINE(name-charset,i)=utf8") in new stack -- Executing [*43@from-internal:2] Set("PJSIP/101-00000143", "CONNECTEDLINE(name,i)=Test Eco") in new stack -- Executing [*43@from-internal:3] Set("PJSIP/101-00000143", "CONNECTEDLINE(num,i)=*43") in new stack -- Executing [*43@from-internal:4] Answer("PJSIP/101-00000143", "") in new stack > 0x7fc0940e4a80 -- Strict RTP learning after remote address set to: 10.7.208.157:50248 > 0x7fc0941034e0 -- Strict RTP learning after remote address set to: 10.7.208.157:50246 > 0x7fc0940e4a80 -- Strict RTP qualifying stream type: audio > 0x7fc0940e4a80 -- Strict RTP switching source address to 10.168.5.201:39519 -- Executing [*43@from-internal:5] Set("PJSIP/101-00000143", "JITTERBUFFER(adaptive)=default") in new stack -- Executing [*43@from-internal:6] Macro("PJSIP/101-00000143", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("PJSIP/101-00000143", "TOUCH_MONITOR=1636707571.457") in new stack -- Executing [s@macro-user-callerid:2] Set("PJSIP/101-00000143", "AMPUSER=101") in new stack -- Executing [s@macro-user-callerid:3] Set("PJSIP/101-00000143", "HOTDESCKCHAN=101-00000143") in new stack -- Executing [s@macro-user-callerid:4] Set("PJSIP/101-00000143", "HOTDESKEXTEN=101") in new stack -- Executing [s@macro-user-callerid:5] Set("PJSIP/101-00000143", "HOTDESKCALL=0") in new stack -- Executing [s@macro-user-callerid:6] ExecIf("PJSIP/101-00000143", "0?Set(HOTDESKCALL=1)") in new stack -- Executing [s@macro-user-callerid:7] ExecIf("PJSIP/101-00000143", "0?Set(CALLERID(name)=)") in new stack -- Executing [s@macro-user-callerid:8] GotoI sing SIP RTP Video TOS bits 136 in TCLASS field. == Using SIP RTP Video CoS mark 4 -- Executing [*43@from-internal:1] Set("PJSIP/101-00000143", "CONNECTEDLINE(name-charset,i)=utf8") in new stack -- Executing [*43@from-internal:2] Set("PJSIP/101-00000143", "CONNECTEDLINE(name,i)=Test Eco") in new stack -- Executing [*43@from-internal:3] Set("PJSIP/101-00000143", "CONNECTEDLINE(num,i)=*43") in new stack -- Executing [*43@from-internal:4] Answer("PJSIP/101-00000143", "") in new stack > 0x7fc0940e4a80 -- Strict RTP learning after remote address set to: 10.7.208.157:50248 > 0x7fc0941034e0 -- Strict RTP learning after remote address set to: 10.7.208.157:50246 > 0x7fc0940e4a80 -- Strict RTP qualifying stream type: audio > 0x7fc0940e4a80 -- Strict RTP switching source address to 10.168.5.201:39519 -- Executing [*43@from-internal:5] Set("PJSIP/101-00000143", "JITTERBUFFER(adaptive)=default") in new stack -- Executing [*43@from-internal:6] Macro("PJSIP/101-00000143", "user-callerid,") in new stack -- Executing [s@macro-user-callerid:1] Set("PJSIP/101-00000143", "TOUCH_MONITOR=1636707571.457") in new stack -- Executing [s@macro-user-callerid:2] Set("PJSIP/101-00000143", "AMPUSER=101") in new stack -- Executing [s@macro-user-callerid:3] Set("PJSIP/101-00000143", "HOTDESCKCHAN=101-00000143") in new stack -- Executing [s@macro-user-callerid:4] Set("PJSIP/101-00000143", "HOTDESKEXTEN=101") in new stack -- Executing [s@macro-user-callerid:5] Set("PJSIP/101-00000143", "HOTDESKCALL=0") in new stack Changed PBX Asterisk 16.13.0 (same phisical phone) but same beahvior: ...........Receive......... .........Transmit.......... BridgeId ChannelId ........ UpTime.. Codec. Count Lost Pct Jitter Count Lost Pct Jitter RTT.... =========================================================================================================== 101-00000144 00:00:34 ulaw 1744 917K 52608 0.000 1727 0 0 0.003 0.006
user2239318 (141 rep)
Nov 15, 2021, 07:39 AM • Last activity: Jan 19, 2022, 02:06 PM
2 votes
4 answers
3584 views
Compiling Asterisk on Debian: Cannot Find `ptlib-config`
My Asterisk `./configure` is returning: checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no check...
My Asterisk ./configure is returning: checking /root/pwlib/include/ptlib.h usability... no checking /root/pwlib/include/ptlib.h presence... no checking for /root/pwlib/include/ptlib.h... no checking /usr/local/include/ptlib.h usability... no checking /usr/local/include/ptlib.h presence... no checking for /usr/local/include/ptlib.h... no checking /usr/include/ptlib.h usability... yes checking /usr/include/ptlib.h presence... yes checking for /usr/include/ptlib.h... yes checking for ptlib-config... no ./configure: line 27253: --ldflags: command not found Cannot find ptlib-config - please install and try again However I clearly have ptlib-dev installed: # dpkg --list libpt* ii libpt-dbg 2.10.4~dfsg-1 ii libpt-dev 2.10.4~dfsg-1 ii libpt-doc 2.10.4~dfsg-1 ii libpt2.10.4 2.10.4~dfsg-1 ii libpth20 2.0.7-16 ii libpthread-stubs0:amd64 0.3-3 ii libpthread-stubs0-dev:amd64 0.3-3
Questionmark (4095 rep)
Jan 8, 2015, 10:04 PM • Last activity: Aug 1, 2021, 03:16 AM
0 votes
1 answers
687 views
reloading asterisk cronjob
I'm trying to create a crontab job that reloads the asterisk config (sip.conf). Running the command manually, works: `/usr/sbin/asterisk -rx "reload"` manually running the script with executable permissions, does not. my script is just 2 lines #!/bin/bash /usr/sbin/asterisk -rx "reload" should i jus...
I'm trying to create a crontab job that reloads the asterisk config (sip.conf). Running the command manually, works: /usr/sbin/asterisk -rx "reload" manually running the script with executable permissions, does not. my script is just 2 lines #!/bin/bash /usr/sbin/asterisk -rx "reload" should i just try to run /usr/sbin/asterisk -rx "reload" straight from crontab, without actually having a .sh file somewhere?
Eric (1 rep)
Jul 9, 2021, 11:26 PM • Last activity: Jul 10, 2021, 11:09 AM
-1 votes
1 answers
277 views
How to run a PowerShell file during boot as root on Asterisk?
On Asterisk boot/reboot I want a PowerShell file to be executed. I have created a *.ps1* file on ```/var/spool/``` which is supposed to copy new files from an asterisk directory and transfer them to an Azure storage container. This file is supposed to get each last recording file and transfer it to...
On Asterisk boot/reboot I want a PowerShell file to be executed. I have created a *.ps1* file on
/var/spool/
which is supposed to copy new files from an asterisk directory and transfer them to an Azure storage container. This file is supposed to get each last recording file and transfer it to the Azure's container. When I run the command manually on root, it works. This is an output of a recording file successfully uploaded on Azure's container.
Name                 BlobType  Length          ContentType                    L
                                                                               a
                                                                               s
                                                                               t
                                                                               M
                                                                               o
                                                                               d
                                                                               i
                                                                               f
                                                                               i
                                                                               e
                                                                               d
----                 --------  ------          -----------                    -
out-067…9249.0.wav BlockBlob 44              application/octet-stream       2
uploaded!
To get every new recordings it has to be run endlessly ( check for new files every one minute). To do this I have used a cycle do/while($true), with a command
60
inside it. In case of any system reboot or power outage I want this file (ps1) to start running again after OS boot. To do this I tried adding the command
/var/spool/transferrecordings.ps1
on **/etc/rc.local** to make it work in case of a system reboot. I edited this directory with
/etc/rc.local
as below: This is the script I'm using in the directory
/etc/rc.local
.
#!/bin/bash

# THIS FILE IS ADDED FOR COMPATIBILITY PURPOSES
#
# It is highly advisable to create own systemd services or udev rules
# to run scripts during boot instead of using this file.
#
# In contrast to previous versions due to parallel execution during boot
# this script will NOT be run after all other services.
#
# Please note that you must run 'chmod +x /etc/rc.d/rc.local' to ensure
# that this script will be executed during boot.

pwsh /var/spool/transferrecordings.ps1
exit 0
But nothing seems to happen on server boot. I tried editing crontab with
-e
by adding command:
@reboot pwsh /var/spool/transferrecordings.ps1
. Nothing again. I'm using Sangoma Linux (CentOS 3.10.0). Any suggestions please?
admiri (101 rep)
Apr 26, 2021, 11:45 AM • Last activity: Apr 29, 2021, 02:18 PM
3 votes
1 answers
1911 views
Routing between two networks that have duplicate IP addresses
On a linux box we have three network interfaces, they look like below | CentOS 6 Server | ---------> eth0 (DHCP (192.168.1.x) Default Gateway, connects to a wired internet, |----------> eth1 (IP : (10.165.11.139) GW to be used : (10.165.11.137), connects to a network A |----------> eth2 (IP : (10.15...
On a linux box we have three network interfaces, they look like below | CentOS 6 Server | ---------> eth0 (DHCP (192.168.1.x) Default Gateway, connects to a wired internet, |----------> eth1 (IP : (10.165.11.139) GW to be used : (10.165.11.137), connects to a network A |----------> eth2 (IP : (10.150.114.190) GW to be used: (10.150.114.191), connects to a network B Problem here is that both network A and network B have nodes with same IP, example : 10.232.130.171 10.232.130.172 10.232.131.100 route-eth1 file looks like: 10.232.130.0/24 via 10.165.11.137 10.232.131.0/24 via 10.165.11.137 route-eth2 file looks like: 10.232.130.0/24 via 10.150.114.189 10.232.131.0/24 via 10.150.114.189 so pinging 10.232.130.171 will always route it thru eth1 and not eth2 tried with application which binds with interface (asterisk PBX), incoming connection from above IP work fine, but any response to it is sent via eth1, hence rejected. Any pointers how to resolve this?
user1263746 (516 rep)
Apr 12, 2021, 10:46 AM • Last activity: Apr 12, 2021, 08:39 PM
0 votes
1 answers
156 views
Asterisk: a little confused about the "parameters" of extensions
Reading on [this wiki][1] i see this extension exten => 6001,1,Dial(PJSIP/demo-alice,20) As I understand exten is the extension 6001 the number to call 1 is the priority Dial the application to use PJSIP/demo-alice is defined in sip.conf 20 ??? I don't understand the 20, is a timeout? Or what? Thank...
Reading on this wiki i see this extension exten => 6001,1,Dial(PJSIP/demo-alice,20) As I understand exten is the extension 6001 the number to call 1 is the priority Dial the application to use PJSIP/demo-alice is defined in sip.conf 20 ??? I don't understand the 20, is a timeout? Or what? Thanks
elbarna (13690 rep)
Apr 5, 2021, 10:25 PM • Last activity: Apr 6, 2021, 08:16 PM
0 votes
1 answers
380 views
How to delete a speed dial in Asterisk/FreePBX
Using FreePBX 15. This is a test system for evaluating the possible use of FreePBX in our university environment. From a phone, I used the Asterisk star code *75 to enter a speed dial. Now I want to delete that. I cannot find any way to do it. Asterisk doesn't provide a cancellation code. I checked...
Using FreePBX 15. This is a test system for evaluating the possible use of FreePBX in our university environment. From a phone, I used the Asterisk star code *75 to enter a speed dial. Now I want to delete that. I cannot find any way to do it. Asterisk doesn't provide a cancellation code. I checked the Asterisk Phonebook, and my entry is not there, yet it is still active when I dial the speed code from a phone. Any help with this would be much appreciated. Thanks,
Phil D'Agostino (1 rep)
Jan 16, 2021, 09:40 PM • Last activity: Jan 18, 2021, 01:22 PM
1 votes
1 answers
1632 views
FreePBX No connection to Asterisk
I am new to asterisk and freePBX. I set up a server years ago but it was very basic and used for less than a month. I’m now launching a new company and found myself in need of a VOIP system so I’m back to asterisk. I installed asterisk on my debian VPS, then installed free PBX. At first i was having...
I am new to asterisk and freePBX. I set up a server years ago but it was very basic and used for less than a month. I’m now launching a new company and found myself in need of a VOIP system so I’m back to asterisk. I installed asterisk on my debian VPS, then installed free PBX. At first i was having issues with getting the freePBX installer to find my asterisk server. After creating an aserisk user on debian and changing the run user-group in /etc/default/asterisk the installer worked. After the FreePBX installation, localhost/ was redirecting to localhost/admin/config.php but was only showing a blank screen. After running fwconsole ma installall the page started working. Despite that, connection to asterisk cannot be established. Running fwconsole start works just fine, but when running fwconsole restart I get get UCP Node Server is not running. When running fwconsole restart again I get Core FastAGI Server is not running & UCP Node Server is not running. **After checking logs for freepbx I found this:**
[2020-Nov-23 14:42:32] [freepbx.INFO]: Connection attmempt to AMI failed [] []
**The UCP error log in asterisk yielded this:**
2020-11-23 13:40 +00:00: { [Error: Can’t connect to MySQL server on ‘::1’ (111 “Connection refused”)] code: 2003 }

2020-11-23 13:40 +00:00: There was an error with MySQL Connection
**ucp_out.log in asterisk displays this:**
2020-11-23 13:40 +00:00: Starting FreePBX…
2020-11-23 13:40 +00:00: { AMPDBUSER: ‘freepbxuser’,
2020-11-23 13:40 +00:00: AMPDBPASS: ‘****’,
2020-11-23 13:40 +00:00: AMPDBHOST: ‘localhost’,
2020-11-23 13:40 +00:00: AMPDBNAME: ‘asterisk’,
2020-11-23 13:40 +00:00: AMPDBENGINE: ‘mysql’,
2020-11-23 13:40 +00:00: datasource: ‘’ }
I am using mariadb, is there a way maybe this user or database does not exist? Thank you for your help! --- Forgot to mention that when accessing Advanced Setting in FreePBX I get a blank page. enter image description here
dnld (21 rep)
Nov 23, 2020, 01:52 PM • Last activity: Dec 6, 2020, 03:09 PM
0 votes
1 answers
1315 views
asterisk dial hangup direction - Is there a variable I can use to record the direction of hangup?
Is there a variable I can use to record the direction of hangup? Example: Calling ----> Called. Called party hangs up. Info I want --> Release from Called party Example 2: Calling ----> Called. Caller hangs up. Info I want --> Release from Calling party
Is there a variable I can use to record the direction of hangup? Example: Calling ----> Called. Called party hangs up. Info I want --> Release from Called party Example 2: Calling ----> Called. Caller hangs up. Info I want --> Release from Calling party
Bugfixer (101 rep)
Apr 22, 2015, 06:44 AM • Last activity: Nov 28, 2020, 01:03 PM
2 votes
2 answers
1453 views
Asterisk: how to change the CLI color prompt?
The [Asterisk wiki][1] says that the color of the CLI prompt can be changed with `%Cn[;n]`. ie: export ASTERISK_PROMPT="%C31[%H]: " but when I use the export above, I get broken prompt `[1;31m[voip]: [1;0m`, where `voip` is my hostname. My terminal is color capable. In fact, I am using colors in `zs...
The Asterisk wiki says that the color of the CLI prompt can be changed with %Cn[;n]. ie: export ASTERISK_PROMPT="%C31[%H]: " but when I use the export above, I get broken prompt [1;31m[voip]: [1;0m, where voip is my hostname. My terminal is color capable. In fact, I am using colors in zsh and bash. And echo $TERM gives me: xterm-256color Here is a screenshot of my terminal: enter image description here As can be seen, even the asterisk messages/logs are colored. I have asked on the asterisk mailing list and the syntax export ASTERISK_PROMPT="%C31[%H]: " is correct. Some people reported that color prompt works for them, others had same problem as I am reporting. This leads me to suspect, that the problem might not be in asterisk, but some complex interplay of OS environment, terminal, terminal emulator, etc. I have tried several different terminal emulators: terminator, xterm, gnome-terminal, eterm, konsole. The problem is the same. I use Debian 10 both on the asterisk server and my desktop from which I am connecting. **How can I get colored CLI prompt on Asterisk?**
Martin Vegter (586 rep)
Jun 4, 2020, 05:26 AM • Last activity: Jul 6, 2020, 12:41 AM
0 votes
0 answers
108 views
Trying to build a RPM for dahdi lead to boot fail on the destination machine
I am trying to build a rpm for Centos 7 to be able to install dahdi without the need of build tools on the destination machine. For that, I tried to create this specs file : ``` Name: dahdi-linux-complete Version: 3.1.0+3.1.0 Release: 1%{?dist} Summary: DAHDI Linux Asterisk kernel modules License: G...
I am trying to build a rpm for Centos 7 to be able to install dahdi without the need of build tools on the destination machine. For that, I tried to create this specs file :
Name:           dahdi-linux-complete
Version:        3.1.0+3.1.0
Release:        1%{?dist}
Summary:        DAHDI Linux Asterisk kernel modules

License:        GPL
URL:            http://downloads.digium.com/pub/telephony/dahdi-linux-complete/ 
Source0:        $RPM_SOURCE_DIR/dahdi-linux-complete-3.1.0+3.1.0.tar.gz

BuildRequires:  kernel-headers = 3.10.0-1127.8.2.el7, kernel-devel = 3.10.0-1127.8.2.el7 
Requires:       kernel = 3.10.0-1127.8.2.el7

%description
Kernel modules to use with the Asterisk telephony server.

%prep
%setup -q

%build
make %{?_smp_mflags}

%install
rm -rf $RPM_BUILD_ROOT
%make_install
make install-config

%files
%defattr(-,root,root,-)
%doc
/etc/bash_completion.d/dahdi
/etc/dahdi/assigned-spans.conf.sample
/etc/dahdi/modules.sample
/etc/dahdi/span-types.conf.sample
/etc/dahdi/system.conf.sample
/etc/udev/rules.d/dahdi.rules
/etc/udev/rules.d/xpp.rules
/lib/firmware/.dahdi-fw-a4a-a0017
/lib/firmware/.dahdi-fw-a4b-d001e
/lib/firmware/.dahdi-fw-a8a-1d0017
/lib/firmware/.dahdi-fw-a8b-1f001e
/lib/firmware/.dahdi-fw-hx8-2.06
/lib/firmware/.dahdi-fw-oct6114-032-1.05.01
/lib/firmware/.dahdi-fw-oct6114-064-1.05.01
/lib/firmware/.dahdi-fw-oct6114-128-1.05.01
/lib/firmware/.dahdi-fw-oct6114-256-1.05.01
/lib/firmware/.dahdi-fw-tc400m-MR6.12
/lib/firmware/.dahdi-fw-te133-7a001e
/lib/firmware/.dahdi-fw-te134-780017
/lib/firmware/.dahdi-fw-te435-13001e
/lib/firmware/.dahdi-fw-te436-10017
/lib/firmware/.dahdi-fw-te820-1.76
/lib/firmware/.dahdi-fw-vpmoct032-1.12.0
/lib/firmware/dahdi-fw-a4a.bin
/lib/firmware/dahdi-fw-a4b.bin
/lib/firmware/dahdi-fw-a8a.bin
/lib/firmware/dahdi-fw-a8b.bin
/lib/firmware/dahdi-fw-hx8.bin
/lib/firmware/dahdi-fw-oct6114-032.bin
/lib/firmware/dahdi-fw-oct6114-064.bin
/lib/firmware/dahdi-fw-oct6114-128.bin
/lib/firmware/dahdi-fw-oct6114-256.bin
/lib/firmware/dahdi-fw-tc400m.bin
/lib/firmware/dahdi-fw-te133.bin
/lib/firmware/dahdi-fw-te134.bin
/lib/firmware/dahdi-fw-te435.bin
/lib/firmware/dahdi-fw-te436.bin
/lib/firmware/dahdi-fw-te820.bin
/lib/firmware/dahdi-fw-vpmoct032.bin
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_dynamic.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_dynamic_eth.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_dynamic_ethmf.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_dynamic_loc.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_echocan_jpah.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_echocan_kb1.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_echocan_mg2.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_echocan_sec.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_echocan_sec2.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_transcode.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/dahdi_vpmadt032_loader.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/oct612x/oct612x.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/voicebus/dahdi_voicebus.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/wcaxx.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/wcb4xxp/wcb4xxp.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/wct4xxp/wct4xxp.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/wctc4xxp/wctc4xxp.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/wctdm24xxp/wctdm24xxp.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/wcte13xp.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/wcte43x.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/xpp/xpd_bri.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/xpp/xpd_echo.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/xpp/xpd_fxo.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/xpp/xpd_fxs.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/xpp/xpd_pri.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/xpp/xpp.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/dahdi/xpp/xpp_usb.ko
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.alias
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.alias.bin
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.builtin.bin
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.dep
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.dep.bin
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.devname
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.softdep
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.symbols
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.symbols.bin
/usr/include/dahdi/dahdi_config.h
/usr/include/dahdi/fasthdlc.h
/usr/include/dahdi/kernel.h
/usr/include/dahdi/tonezone.h
/usr/include/dahdi/user.h
/usr/include/dahdi/wctdm_user.h
/usr/lib/dracut/dracut.conf.d/50-dahdi.conf
/usr/lib/libtonezone.a
/usr/lib/libtonezone.la
/usr/lib/libtonezone.so
/usr/lib/libtonezone.so.2
/usr/lib/libtonezone.so.2.0
/usr/lib/libtonezone.so.2.0.0
/usr/local/share/perl5/Dahdi.pm
/usr/local/share/perl5/Dahdi/Chans.pm
/usr/local/share/perl5/Dahdi/Config/Gen.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Assignedspans.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Chandahdi.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Freepbxdb.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Modules.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Spantypes.pm
/usr/local/share/perl5/Dahdi/Config/Gen/System.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Unicall.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Users.pm
/usr/local/share/perl5/Dahdi/Config/Gen/Xpporder.pm
/usr/local/share/perl5/Dahdi/Config/Params.pm
/usr/local/share/perl5/Dahdi/Hardware.pm
/usr/local/share/perl5/Dahdi/Hardware/PCI.pm
/usr/local/share/perl5/Dahdi/Hardware/USB.pm
/usr/local/share/perl5/Dahdi/Span.pm
/usr/local/share/perl5/Dahdi/Utils.pm
/usr/local/share/perl5/Dahdi/Xpp.pm
/usr/local/share/perl5/Dahdi/Xpp/Line.pm
/usr/local/share/perl5/Dahdi/Xpp/Mpp.pm
/usr/local/share/perl5/Dahdi/Xpp/Xbus.pm
/usr/local/share/perl5/Dahdi/Xpp/Xpd.pm
/usr/sbin/dahdi_cfg
/usr/sbin/dahdi_genconf
/usr/sbin/dahdi_hardware
/usr/sbin/dahdi_maint
/usr/sbin/dahdi_monitor
/usr/sbin/dahdi_registration
/usr/sbin/dahdi_scan
/usr/sbin/dahdi_span_assignments
/usr/sbin/dahdi_span_types
/usr/sbin/dahdi_speed
/usr/sbin/dahdi_test
/usr/sbin/dahdi_waitfor_span_assignments
/usr/sbin/fxotune
/usr/sbin/lsdahdi
/usr/sbin/sethdlc
/usr/sbin/twinstar
/usr/sbin/xpp_blink
/usr/sbin/xpp_sync
/usr/share/dahdi/FPGA_1141.hex
/usr/share/dahdi/FPGA_1151.hex
/usr/share/dahdi/FPGA_1161.201.hex
/usr/share/dahdi/FPGA_1161.202.hex
/usr/share/dahdi/FPGA_1161.203.hex
/usr/share/dahdi/FPGA_1161.hex
/usr/share/dahdi/FPGA_FXS.hex
/usr/share/dahdi/PIC_TYPE_1.hex
/usr/share/dahdi/PIC_TYPE_2.hex
/usr/share/dahdi/PIC_TYPE_3.hex
/usr/share/dahdi/PIC_TYPE_4.hex
/usr/share/dahdi/PIC_TYPE_6.hex
/usr/share/dahdi/USB_FW.201.hex
/usr/share/dahdi/USB_FW.202.hex
/usr/share/dahdi/USB_FW.203.hex
/usr/share/dahdi/USB_FW.hex
/usr/share/dahdi/USB_RECOV.hex
/usr/share/dahdi/XppConfig.pm
/usr/share/dahdi/astribank_hook
/usr/share/dahdi/dahdi_auto_assign_compat
/usr/share/dahdi/dahdi_handle_device
/usr/share/dahdi/dahdi_span_config
/usr/share/dahdi/handle_device.d/10-span-types
/usr/share/dahdi/handle_device.d/20-span-assignments
/usr/share/dahdi/init_card_1_30
/usr/share/dahdi/init_card_2_30
/usr/share/dahdi/init_card_3_30
/usr/share/dahdi/init_card_4_30
/usr/share/dahdi/init_card_5_30
/usr/share/dahdi/init_card_6_30
/usr/share/dahdi/span_config.d/10-dahdi-cfg
/usr/share/dahdi/span_config.d/20-fxotune
/usr/share/dahdi/span_config.d/50-asterisk
/usr/share/dahdi/waitfor_xpds
/usr/share/dahdi/xpp_fxloader
/usr/share/man/man8/dahdi_cfg.8.gz
/usr/share/man/man8/dahdi_genconf.8.gz
/usr/share/man/man8/dahdi_hardware.8.gz
/usr/share/man/man8/dahdi_maint.8.gz
/usr/share/man/man8/dahdi_monitor.8.gz
/usr/share/man/man8/dahdi_registration.8.gz
/usr/share/man/man8/dahdi_scan.8.gz
/usr/share/man/man8/dahdi_span_assignments.8.gz
/usr/share/man/man8/dahdi_span_types.8.gz
/usr/share/man/man8/dahdi_test.8.gz
/usr/share/man/man8/dahdi_tool.8.gz
/usr/share/man/man8/dahdi_waitfor_span_assignments.8.gz
/usr/share/man/man8/fxotune.8.gz
/usr/share/man/man8/lsdahdi.8.gz
/usr/share/man/man8/twinstar.8.gz
/usr/share/man/man8/xpp_blink.8.gz
/usr/share/man/man8/xpp_sync.8.gz


%changelog
So far it seems to works as expected, but the machine where I am installing the generated rpm will failed at boot, the systemd fails at mounting the efi partition, because it can't find the vfat driver. Maybe I need to perform a dracut action as a post install hook. But I am not really sure this is what I need to do, and I don't know how to do it. Update : it seems that providing one of these files :
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.alias
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.alias.bin
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.builtin.bin
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.dep
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.dep.bin
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.devname
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.softdep
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.symbols
/lib/modules/3.10.0-1127.8.2.el7.x86_64/modules.symbols.bin
Make the kernel fail to load. For a test I have saved these files, and restored them after the installation of my rpm, the kernel load correctly this time, but dahdi drivers are not loaded. So it seems that instead of overwriting these files, I have to "patch" them in a post hook script of the rpm ?
iXô (327 rep)
Jun 1, 2020, 11:40 AM • Last activity: Jun 1, 2020, 12:30 PM
3 votes
1 answers
854 views
editrc: changing keybindings in /etc/editrc
`Asterisk` uses the `editline` library, and the keybindings can be configured in `/etc/editrc`. I have defined some of my own keybindins, some other are left to default values. How can I print the current keybindings in Asterisk? I am looking for something similar to what `bindkey` does in `zsh`. Al...
Asterisk uses the editline library, and the keybindings can be configured in /etc/editrc. I have defined some of my own keybindins, some other are left to default values. How can I print the current keybindings in Asterisk? I am looking for something similar to what bindkey does in zsh. Also, how can I "unbind" a key, such as Ctrl+C ? And How would I create new keybinding that would bind Ctrl+D to exit/quit ? here is my current /etc/editrc: bind "^W" ed-delete-prev-word bind "\e[1;5D" vi-prev-word bind "\e[1;5C" vi-next-word bind ^[[5~ ed-search-next-history bind ^[[6~ ed-search-prev-history
Martin Vegter (586 rep)
Oct 25, 2019, 10:54 AM • Last activity: May 31, 2020, 05:11 PM
0 votes
2 answers
841 views
Error: you do not appear to have the sources for the 2.6.32-042stab102.9 kernel installed
I've just bought a new VPS with Cent OS 6.6 installed. I'm attempting to install Asterisk 11 on this VPS via command line remotely. I've used the directions [here](http://www.voip-info.org/wiki/view/Asterisk+11+Installation+on+CentOS+6) however I get this error: you do not appear to have the sources...
I've just bought a new VPS with Cent OS 6.6 installed. I'm attempting to install Asterisk 11 on this VPS via command line remotely. I've used the directions [here](http://www.voip-info.org/wiki/view/Asterisk+11+Installation+on+CentOS+6) however I get this error: you do not appear to have the sources for the 2.6.32-042stab102.9 kernel installed when running: cd /usr/src/dahdi-linux-complete* make && make install && make config How can I install this kernel and continue my install?
Totally Tech IT (1 rep)
Mar 29, 2015, 04:56 PM • Last activity: May 25, 2020, 09:04 PM
Showing page 1 of 20 total questions