My asterisk can move around the extension with "Goto()" but it exits immediately when meets the answer application. Could anyone help me to figure out the problem?
Executing [6138#from-sip-external:1] Goto("SIP/10.65.104.17-00000005", "kiem-tra-so-goi-vao,s,1") in new stack
-- Goto (kiem-tra-so-goi-vao,s,1)
-- Executing [s#kiem-tra-so-goi-vao:1] Answer("SIP/10.65.104.17-00000005", "") in new stack
== Spawn extension (kiem-tra-so-goi-vao, s, 1) exited non-zero on 'SIP/10.65.104.17-00000005'
Below is my dialplan:
[from-sip-external]
exten => 1234,1(dest-ext),Goto(kiem-tra-so-goi-vao,s,1)
[kiem-tra-so-goi-vao]
exten => s,1,Answer()
exten => s,n,Playback(hello-world)
exten => s,n,hangup()
Most likly you have NAT or codec issues
To get more info you have enable sip debug
asterisk -r
sip set debug on
If you see in rtp part local ips, while connection from outside, that mean you have issues with nat.
IF you see "not acceptable here", that is codec.
I guess, it should be Hangup() not hangup(). I hope that will solve the problem. If not, most likely, codec mismatch is the culprit.
Change your last line to this.
same =>n, Hangup()
Related
This is the code in my .conf file for asterisk freepbx I know for sure blocked is true
same => n,Noop(${Blocked})
same => n,GotoIf($["${Blocked}"=="TRUE"]?blah-ivr-v5-difficulties,s,1)
same => n,Noop("The code reaches here and doesnt hit the context at the top but its value is true")
This is the logs from asterisk saying Blocked is true
[2021-10-28 11:53:08] VERBOSE[11267][C-001e313b] pbx.c: Executing [s#blah-ivr-v5-pinreset-2:11] NoOp("SIP/blah-00245082", "true") in new stack
[2021-10-28 11:53:08] VERBOSE[11267][C-001e313b] pbx.c: Executing [s#blah-ivr-v5-pinreset-2:12] GotoIf("SIP/blah-00245082", "0?blah-ivr-v5-difficulties,s,1") in new stack
Any Idea why its not branching correctly?
Unfortanly asterisk know nothing about machine learning etc and does not think that
"true" == "TRUE"
In this case both are strings and strings with capitals not the same as lower strings(see UTF-8 table).
Even more, asterisk's boolean value is 1(integer)
https://unicode-table.com/en/
http://www.asteriskdocs.org/en/2nd_Edition/asterisk-book-html-chunk/asterisk-APP-B-89.html
https://wiki.asterisk.org/wiki/display/AST/Manipulating+Variables+Basics
Consider the following asterisk dialplan. After the last sound file is played it hangs up immediately and the waitexten timeout parameter seems to be ignored. The behaviour I am trying to achieve is to wait a given number of seconds after the last sound file is completed to get a response and then hang up. The last sound file says "Dial * to hear these options again". If I set the waitexten timeout to say 60, it hangs up before the sound files are played. A shorter time allows them all to be played but then it hangs up immediately. Any suggestions for handling this a better way are welcome
[mainmenu]
exten => s,1,Wait(0.25)
same => 2,Answer()
same => 3,Background(/srv/asterisk/sounds/optionslist)
same => n,Background(/srv/asterisk/sounds/dial2cs)
same => n,Background(/srv/asterisk/sounds/dial3ma)
same => n,Background(/srv/asterisk/sounds/dial4ac)
same => n,Background(/srv/asterisk/sounds/dial0)
same => n,Background(/srv/asterisk/sounds/dialstar)
same => n,WaitExten(20)
exten => 2,1,Goto(cs,2,1)
exten => *,1,Goto(s,3)
console output
== Using SIP RTP CoS mark 5
-- Executing [+12345#public:1] Goto("SIP/xxx.pstn.twilio.com-00000044", "mainmenu,s,1") in new stack
-- Goto (mainmenu,s,1)
-- Executing [s#mainmenu:1] Wait("SIP/xxx.pstn.twilio.com-00000044", "0.25") in new stack
-- Executing [s#mainmenu:4] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial2cs") in new stack
-- Executing [s#mainmenu:5] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial3ma") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial3ma.slin' (language 'en')
-- Executing [s#mainmenu:6] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial4ac") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial4ac.slin' (language 'en')
-- Executing [s#mainmenu:7] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dial0") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dial0.slin' (language 'en')
-- Executing [s#mainmenu:8] BackGround("SIP/xxx.pstn.twilio.com-00000044", "/srv/asterisk/sounds/dialstar") in new stack
-- <SIP/xxx.pstn.twilio.com-00000044> Playing '/srv/asterisk/sounds/dialstar.slin' (language 'en')
[
[Aug 4 05:03:28] WARNING[2225]: chan_sip.c:4175 retrans_pkt: Retransmission timeout reached on transmission xxx#0.0.0.0 for seqno 5305 (Critical Response) -- See https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions
Packet timed out after 32001ms with no response
[Aug 4 05:03:28] WARNING[2225]: chan_sip.c:4204 retrans_pkt: Hanging up call xxx#0.0.0.0 - no reply to our critical packet (see https://wiki.asterisk.org/wiki/display/AST/SIP+Retransmissions).
== Spawn extension (mainmenu, s, 8) exited non-zero on 'SIP/xxx.pstn.twilio.com-00000044'
Asterisk 11.7.0~dfsg-1ubuntu1
I assume that you have a problem like in this topic Asterisk,SIP Retransmission timeout. Try to solve the problem with the NAT settings or firewall.
In this answer propose to set canreinvite=no in sip.conf.
I have installed Asterisk on Ubuntu
sip.conf
[10000001]
type=friend
host=dynamic
qualify=yes
secret=pw_random
context=demo
[10000002]
type=friend
host=dynamic
qualify=yes
secret=pw_random
context=demo
...
extensions.conf
[demo]
exten => _1XXXXXXX,1,Dial(SIP/${EXTEN})
exten => _1XXXXXXX,2,Set(CALLFILENAME=${EXTEN:1})
exten => _1XXXXXXX,3,Monitor(wav,${CALLFILENAME},m)
However, Asterisk runs Dial and gets stuck, the users can talk each other on call, but Asterisk doesn't record the audio
run asterisk -rvvv, I get
-- Executing [10000001#demo:1] Dial("SIP/10000002-00000045", "SIP/10000001") in new stack
== Using SIP RTP CoS mark 5
-- Called SIP/10000001
-- SIP/10000001-00000046 is ringing
-- SIP/10000001-00000046 answered SIP/10000002-00000045
-- Channel SIP/10000002-00000045 joined 'simple_bridge' basic-bridge <1b882cee-b0f0-473f-aafb-651169788159>
-- Channel SIP/10000001-00000046 joined 'simple_bridge' basic-bridge <1b882cee-b0f0-473f-aafb-651169788159>
Any idea? Thanks!!
Update:
If I modify extensions.conf to
exten => _1XXXXXXX,1,Set(CALLFILENAME=${EXTEN})
exten => _1XXXXXXX,2,Monitor(wav,${CALLFILENAME},m)
exten => _1XXXXXXX,3,Dial(SIP/${EXTEN})
log is
-- Executing [10000001#demo:1] Set("SIP/10000002-00000000", "CALLFILENAME=10000001") in new stack
[Apr 14 00:56:50] WARNING[8649][C-00000000]: pbx.c:4910 pbx_extension_helper: No application 'Monitor' for extension (demo, 10000001, 2)
== Spawn extension (demo, 10000001, 2) exited non-zero on 'SIP/10000002-00000000'
error pbx_extension_helper: No application 'Monitor' for extension is weird.
I fixed it.
It's due to pbx_extension_helper: No application 'Monitor' for extension
Application 'Monitor' is not found because asterisk doesn't load res_monitor module according to this link
So, I add load=res_monitor.so in /etc/asterisk/modules.conf
The file looks like
[modules]
autoload=yes
load=pbx_config.so
load=chan_sip.so
load=chan_iax2.so
load=res_rtp_asterisk.so
load=app_hangup.so
load=app_dial.so
load=app_stack.so
load=res_monitor.so
load=pbx_functions.so
load=codec_ulaw.so
load=codec_gsm.so
load=bridge_simple.so
Thank Shu Zhang!
Have you checked the directory /var/lib/asterisk/sounds/, /var/spool/asterisk/monitor, or you can find your ubuntu files by your file name. Since the verbose didn't show any warning I believe you are doing this right but the recorded file is hiding somewhere. You can fix the file name .
exten => _1XXXXXXX,3,Monitor(wav,fixedfilename)
And find this file name in linux.
Moreover you can take a look of this
[a link]http://www.voip-info.org/wiki/view/Asterisk+cmd+Record
this is an old but i had the same problem, the way you solved it its correct, but you are just loading ALL MODULES... instead of making an slim module load.
[modules]
autoload=yes <---- THIS
load=pbx_config.so <---- DISABLED THIS
load=chan_sip.so
load=chan_iax2.so
So in order to make it work in a slim module load configuracion you must load
[modules]
autoload=no <---- THIS to NO
...
load=func_periodic_hook.so <---- THIS IS NEEDED IN ORDER TO LOAD MONITOR
load=load=res_monitor.so.so
...
Obviouly you need codecs and formats, here is a good slim config.
https://www.voip-info.org/asterisk-slimming/
I have a continental calling card and I'm not sure how to make it possible to dial out with my asterisk server.
It is a VOIP prepaid card. I can call out on a softphone using their server address and my username and password.
I can't figure out my sip.conf or my dial plan.
Here is what I have.
sip.conf:
[continentalcard]
host=continental.com
defaultuser=username ;; user on continental's server
secret=password
register => username:password#continental.com
context=global
[frank]
type=friend
defaultuser=frank ;; user on my local asterisk server
secret=password
host=dynamic
context=internal
extensions.conf:
[global]
CARD=SIP/continentalcard
[internal]
exten => 100,1,Dial(SIP/frank)
same => n,Hangup()
include => continentalcard
[continentalcard] ;; outgoing
exten => _1NXXNXXXXXX,1,Dial(${CARD}/${EXTEN})
I get the following message on the CLI as I try to dial out 1-222-333-4444 (not the real number):
== Using SIP RTP CoS mark 5
-- Executing [12223334444#internal:1] Dial("SIP/frank-00000151", "SIP/continentalcard:12223334444") in new stack
== Using SIP RTP CoS mark 5
[Oct 3 04:02:57] ERROR[22923]: netsock2.c:269 ast_sockaddr_resolve: getaddrinfo("continentalcard", "12223334444", ...): Servname not supported for ai_socktype
[Oct 3 04:02:57] WARNING[22923]: chan_sip.c:5866 create_addr: No such host: continentalcard:12223334444
[Oct 3 04:02:57] WARNING[22923]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
== Everyone is busy/congested at this time (1:0/0/1)
-- Auto fallthrough, channel 'SIP/frank-00000151' status is 'CHANUNAVAIL'
Update: Filled sip.conf with the global context. Also just noticed your missing a / in extensions.conf. Please look below
You have your sip.conf formatted incorrectly.
[global]
register => username:password#continental.com
context=continentalcard
[continentalcard]
host=continental.com
defaultuser=username
secret=password
context=continentalcard
Registration should be placed under the [global] context in sip.conf.
Context should be continentalcard not global. When the softphone dials 1NXXNXXXXXX it should start using the continentalcard context from extensions and perform the Dial(${CARD}/${EXTEN})
I am trying to run a shell script using asterisk AGI. I have used the tutorial mentioned here
http://www.shiffman.net/p5/asterisk/
My extensions.conf is as follows
[default]
include => clicall
[clicall]
exten => _X,1,Goto(s,1);
exten => _X.,1,Goto(s,1);
exten => s,1,Answer();
exten => s,n,EAGI(runEAGI.sh);
The script I am trying to run (runEAGI.sh) is as follows
#!/bin/bash
java /home/sphata001/Downloads/EAGI/JEAGIClient $$
The permissions have been set as 755 and the script is placed in /var/lib/asterisk/agi-bin/. The java file(JEAGIClient) has been compiled beforehand as well . When executing the script manually it runs fine and connects to the server.
But when making a call from SIP client the script executes according to asterisk console but no results are to seen.
I get the following output in the console
== Using SIP RTP CoS mark 5
-- Executing [888#default:1] Goto("SIP/1001-00000027", "s,1") in new stack
-- Goto (default,s,1)
-- Executing [s#default:1] Answer("SIP/1001-00000027", "") in new stack
-- Executing [s#default:2] EAGI("SIP/1001-00000027", "runEAGI.sh") in new stack
-- Launched AGI Script /var/lib/asterisk/agi-bin/runEAGI.sh
-- <SIP/1001-00000027>AGI Script runEAGI.sh completed, returning 0
-- Auto fallthrough, channel 'SIP/1001-00000027' status is 'UNKNOWN'
Any solutions?
Thank you.
Most likly - you need specify full path to java.
Hint: For debugging asterisk AGI simple solution is stop asterisk and start it attached to console, that way you will see all script errors.
asterisk -rx "core stop now"
asterisk -vvvvgc
Also can be usfull enable AGI debugging in asterisk console:
agi set debug on
Check that the script and any resources it requires is owned by the Asterisk user, and that SELinux is not preventing the script from running correctly