Drop from conference after 6 seconds

bob_kuger · December 23, 2015, 9:23pm

Hi,
i’m new with asterisk and i try to mount an asterisk server with some conference services. For the moment i just want to test a simple conference in which users can talk to each others.

But i’ve got a problem. Every call i make are disconected 6 seconds after it has been accepted…

I have no zaptel cards so i’ve compiled the modules with ztdummy.

I’ve a Linux version 2.6.16-1.2069 Fedora Core 4.
lsmod give me that :

[root@Equitation ~]# lsmod |grep ztdummy
ztdummy                 3848  0 
zaptel                188768  8 ztdummy,wcusb,wctdm,wcfxo,wcte11xp,wct1xxp,wct4xxp,tor2

i use freepbx to control my asterisk server. the only thing which doesn’t work is the conference.
maybe my call way is not good. it’s a simple conference with no password to enter in, no leader, etc…

here is my meetme.conf

[code][root@Equitation ~]# cat /etc/asterisk/meetme.conf
[rooms]
#include meetme_additional.conf
[root@Equitation ~]# cat /etc/asterisk/meetme_additional.conf
conf => 218|

[/code]

and the extensions.conf

[ext-meetme]
include => ext-meetme-custom
exten => 218,1,Set(MEETME_ROOMNUM=218)
exten => 218,n,GotoIf($[${DIALSTATUS} = ANSWER]?USER)
exten => 218,n,Answer
exten => 218,n,Wait(1)
exten => 218,n(USER),Set(MEETME_OPTS=i)
exten => 218,n,Goto(STARTMEETME,1)
exten => STARTMEETME,1,MeetMe(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN})
exten => STARTMEETME,n,Hangup
exten => h,1,Hangup

; end of [ext-meetme]

if someone could answer quickly… i’ve got no ideas…

RichardHH · December 23, 2015, 9:23pm

[quote=“jovic”]For the moment i just want to test a simple conference in which users can talk to each others.
[/quote]

Extensions.conf:
Exten=>218,1,MeetMe(218)

meetme.conf:
[rooms]
conf => 218

Thats it.

bob_kuger · December 23, 2015, 9:23pm

it doesn’t works better…

here are my log with my conf :

[quote]Apr 25 10:01:59 DEBUG[3100] chan_sip.c: Setting NAT on RTP to 0
Apr 25 10:01:59 DEBUG[3100] chan_sip.c: Setting NAT on VRTP to 0
Apr 25 10:01:59 DEBUG[3100] chan_sip.c: Stopping retransmission on ‘629335D1-77CC-102C-9B1A-B11A9C64116F@155.208.211.204’ of Response 1: Match Found
Apr 25 10:01:59 DEBUG[3100] chan_sip.c: Setting NAT on RTP to 0
Apr 25 10:01:59 DEBUG[3100] chan_sip.c: Setting NAT on VRTP to 0
Apr 25 10:01:59 DEBUG[3100] chan_sip.c: Checking SIP call limits for device 201
Apr 25 10:01:59 DEBUG[3100] chan_sip.c: build_route: Contact hop: sip:201@155.208.211.204:5060
Apr 25 10:01:59 VERBOSE[3146] logger.c: – Executing Set(“SIP/201-7439”, “MEETME_ROOMNUM=218”) in new stack
Apr 25 10:01:59 WARNING[3146] ast_expr2.fl: ast_yyerror(): syntax error: syntax error, unexpected TOK_EQ, expecting TOK_MINUS or TOK_COMPL or TOK_LP or TOKEN; Input:
Apr 25 10:01:59 WARNING[3146] ast_expr2.fl: If you have questions, please refer to doc/README.variables in the asterisk source.
Apr 25 10:01:59 DEBUG[3146] pbx.c: Expression result is ‘0’
Apr 25 10:01:59 VERBOSE[3146] logger.c: – Executing GotoIf(“SIP/201-7439”, “0?USER”) in new stack
Apr 25 10:01:59 DEBUG[3146] pbx.c: Not taking any branch
Apr 25 10:01:59 VERBOSE[3146] logger.c: – Executing Answer(“SIP/201-7439”, “”) in new stack
Apr 25 10:01:59 DEBUG[3091] channel.c: Avoiding initial deadlock for ‘SIP/201-7439’
Apr 25 10:01:59 DEBUG[3091] channel.c: Avoiding initial deadlock for ‘SIP/201-7439’
Apr 25 10:01:59 VERBOSE[3146] logger.c: – Executing Wait(“SIP/201-7439”, “1”) in new stack
Apr 25 10:02:00 DEBUG[3100] chan_sip.c: Stopping retransmission on ‘629335D1-77CC-102C-9B1A-B11A9C64116F@155.208.211.204’ of Response 2: Match Found
Apr 25 10:02:00 VERBOSE[3146] logger.c: – Executing Festival(“SIP/201-7439”, “hi you are in the conference now| you can be fucked by each other”) in new stack
Apr 25 10:02:00 VERBOSE[3146] logger.c: == Parsing ‘/etc/asterisk/festival.conf’: Apr 25 10:02:00 VERBOSE[3146] logger.c: == Parsing ‘/etc/asterisk/festival.conf’: Found
Apr 25 10:02:00 DEBUG[3146] app_festival.c: Text passed to festival server : hi you are in the conference now
Apr 25 10:02:00 WARNING[3146] app_festival.c: festival_client: connect to server failed
Apr 25 10:02:00 VERBOSE[3146] logger.c: == Spawn extension (from-internal, 218, 5) exited non-zero on ‘SIP/201-7439’
Apr 25 10:02:00 VERBOSE[3146] logger.c: – Executing Macro(“SIP/201-7439”, “hangupcall”) in new stack
Apr 25 10:02:00 VERBOSE[3146] logger.c: – Executing ResetCDR(“SIP/201-7439”, “w”) in new stack
Apr 25 10:02:00 DEBUG[3146] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Apr 25 10:02:00 DEBUG[3146] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2006-04-25 10:01:59’,’“jot1” <201>’,‘201’,‘218’,‘from-internal’, ‘SIP/201-7439’,’’,‘ResetCDR’,‘w’,1,1,‘ANSWERED’,3,’’,‘1145952119.0’)
Apr 25 10:02:01 VERBOSE[3146] logger.c: – Executing NoCDR(“SIP/201-7439”, “”) in new stack
Apr 25 10:02:01 WARNING[3146] cdr.c: CDR on channel ‘SIP/201-7439’ not posted
Apr 25 10:02:01 WARNING[3146] cdr.c: CDR on channel ‘SIP/201-7439’ lacks end
Apr 25 10:02:01 VERBOSE[3146] logger.c: – Executing Wait(“SIP/201-7439”, “5”) in new stack
Apr 25 10:02:06 VERBOSE[3146] logger.c: – Executing Hangup(“SIP/201-7439”, “”) in new stack
Apr 25 10:02:06 VERBOSE[3146] logger.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/201-7439’ in macro 'hangupcall’
Apr 25 10:02:06 VERBOSE[3146] logger.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/201-7439’
Apr 25 10:02:06 DEBUG[3146] chan_sip.c: update_call_counter(201) - decrement call limit counter
Apr 25 10:02:06 DEBUG[3100] chan_sip.c: Stopping retransmission on ‘629335D1-77CC-102C-9B1A-B11A9C64116F@155.208.211.204’ of Request 102: Match Found[/quote]

here are the log with your config:

[quote]Apr 25 10:07:14 DEBUG[3316] chan_sip.c: Setting NAT on RTP to 0
Apr 25 10:07:14 DEBUG[3316] chan_sip.c: Setting NAT on VRTP to 0
Apr 25 10:07:14 DEBUG[3316] chan_sip.c: Stopping retransmission on ‘9324A387-A74A-8550-BC87-0298F0766890@155.208.211.204’ of Response 1: Match Found
Apr 25 10:07:14 DEBUG[3316] chan_sip.c: Setting NAT on RTP to 0
Apr 25 10:07:14 DEBUG[3316] chan_sip.c: Setting NAT on VRTP to 0
Apr 25 10:07:14 DEBUG[3316] chan_sip.c: Checking SIP call limits for device 201
Apr 25 10:07:14 DEBUG[3316] chan_sip.c: build_route: Contact hop: sip:201@155.208.211.204:5060
Apr 25 10:07:14 WARNING[3384] pbx.c: No application ‘MeetMe’ for extension (from-internal, 218, 1)
Apr 25 10:07:14 VERBOSE[3384] logger.c: == Spawn extension (from-internal, 218, 1) exited non-zero on ‘SIP/201-3033’
Apr 25 10:07:14 VERBOSE[3384] logger.c: – Executing Macro(“SIP/201-3033”, “hangupcall”) in new stack
Apr 25 10:07:14 VERBOSE[3384] logger.c: – Executing ResetCDR(“SIP/201-3033”, “w”) in new stack
Apr 25 10:07:14 DEBUG[3384] cdr_addon_mysql.c: cdr_mysql: inserting a CDR record.
Apr 25 10:07:14 DEBUG[3384] cdr_addon_mysql.c: cdr_mysql: SQL command as follows: INSERT INTO cdr (calldate,clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,duration,billsec,disposition,amaflags,accountcode,uniqueid) VALUES (‘2006-04-25 10:07:14’,’“jot1” <201>’,‘201’,‘218’,‘from-internal’, ‘SIP/201-3033’,’’,‘ResetCDR’,‘w’,0,0,‘NO ANSWER’,3,’’,‘1145952434.1’)
Apr 25 10:07:14 VERBOSE[3384] logger.c: – Executing NoCDR(“SIP/201-3033”, “”) in new stack
Apr 25 10:07:14 WARNING[3384] cdr.c: CDR on channel ‘SIP/201-3033’ not posted
Apr 25 10:07:14 WARNING[3384] cdr.c: CDR on channel ‘SIP/201-3033’ lacks end
Apr 25 10:07:14 VERBOSE[3384] logger.c: – Executing Wait(“SIP/201-3033”, “5”) in new stack
Apr 25 10:07:19 VERBOSE[3384] logger.c: – Executing Hangup(“SIP/201-3033”, “”) in new stack
Apr 25 10:07:19 VERBOSE[3384] logger.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on ‘SIP/201-3033’ in macro 'hangupcall’
Apr 25 10:07:19 VERBOSE[3384] logger.c: == Spawn extension (macro-hangupcall, s, 4) exited non-zero on 'SIP/201-3033’
Apr 25 10:07:19 DEBUG[3384] chan_sip.c: update_call_counter(201) - decrement call limit counter
Apr 25 10:07:19 DEBUG[3384] chan_sip.c: AST hangup cause 16 (no match found in SIP)
Apr 25 10:07:19 DEBUG[3316] chan_sip.c: Stopping retransmission on ‘9324A387-A74A-8550-BC87-0298F0766890@155.208.211.204’ of Response 2: Match Found
[/quote]

RichardHH · December 23, 2015, 9:23pm

Apr 25 10:07:14 WARNING[3384] pbx.c: No application ‘MeetMe’ for extension (from-internal, 218, 1)

meetme isnt initialized ? You might wanna check why.

Im not sure, but try the uper/lowercase like this:
Meetme

Not sure if asterisk is case sensitive there, but at least asterisk complains about not knowing an application called “MeetMe”.

bob_kuger · December 23, 2015, 9:23pm

always the same warning

[quote]Apr 25 11:24:26 WARNING[32012] pbx.c: No application ‘Meetme’ for extension (from-internal, 218, 1)
[/quote]

in my extensions.conf i’ve this :

[code][from-internal]
;allow phones to use applications
include => app-userlogonoff
include => app-directory
include => app-dnd
include => app-callforward
include => app-callwaiting
include => app-messagecenter
include => app-calltrace
include => parkedcalls
include => from-internal-custom
;allow phones to dial other extensions
include => ext-fax
include => ext-zapbarge
include => ext-record
include => ext-test
;allow phones to access generated contexts
include => from-internal-additional
exten => s,1,Macro(hangupcall)
exten => h,1,Macro(hangupcall)

[from-internal-additional]
include => from-internal-additional-custom
include => ext-meetme
include => ext-queues
include => ext-group
include => ext-local
include => outbound-allroutes
exten => h,1,Hangup

[ext-meetme]
;include => ext-meetme-custom
;exten => 218,1,Set(MEETME_ROOMNUM=218)
;exten => 218,n,GotoIf($[${DIALSTATUS} = ANSWER]?USER)
;exten => 218,n,Answer
;exten => 218,n,Wait(1)
;exten => 218,n(USER),Set(MEETME_OPTS=i)
;exten => 218,n,Goto(STARTMEETME,1)
;exten => STARTMEETME,1,Meetme(${MEETME_ROOMNUM},${MEETME_OPTS},${PIN})
;exten => STARTMEETME,n,Hangup
;exten => h,1,Hangup
exten => 218,1,Meetme(218)
[/code]

as we say in french “j’en perds mon latin…”

i let the url where i put my .conf files : http://terrijovic.free.fr/sip

whoiswes · December 23, 2015, 9:23pm

do a ‘show applications’ from the CLI - if you don’t see Meetme listed, then the module isn’t being loaded. you might also check your modules.conf, see if there is a noload => app_meetme.so or something similar…

bob_kuger · December 23, 2015, 9:23pm

yeah i’ve done it and i’ve seen the app_meetme.so didn’t exist.

I don’t know why asterisk has compiled whithout the app_meetme.so, but after an other compilation the meetme.so appeared.

Thank you for your help !!

just a final question, is there something to do to have a videoconference, or with these params are enough ?

and one more time : sorry my english ^^

RichardHH · December 23, 2015, 9:23pm

i let the url where i put my .conf files

hehehe… the all important
modules.conf

is missing

Make sure the lines

[modules]
autoload=yes

are there…

RichardHH · December 23, 2015, 9:23pm

just a final question, is there something to do to have a
videoconference, or with these params are enough ?

Dunno, im not into cybersex… Joking…no seriously, i dont use video so no experience with that.

validius · December 23, 2015, 9:50pm

[quote=“bob_kuger”]yeah i’ve done it and i’ve seen the app_meetme.so didn’t exist.

I don’t know why asterisk has compiled whithout the app_meetme.so, but after an other compilation the meetme.so appeared.

Thank you for your help !!

just a final question, is there something to do to have a videoconference, or with these params are enough ?

and one more time : sorry my english ^^[/quote]

There is a directive in the makefile that tests to see if Zaptel is isntalled. If it isnt then meetme (and a few other apps) wownt be built. That is that it worked on the second build and not the first.