[asterisk-dev] send DTMF not Detected on SIP/XXXX using AMI +AGI+EXEC+SendDTMF

2015-04-22 Thread HKC323
This following setting has been configured. dtmfmode=rfc2833 has been set . set (Agi:async) sip client : jisti sip soft phone == issue : When Agent A1 and Agent A2 are on call DTMF must be Passed and Feature must be exeute . here DTMF detect but feaure not extecuted . in case of local

[asterisk-dev] send DTMF not Detected on SIP/XXXX using AMI +AGI+EXEC+SendDTMF

2015-04-22 Thread HKC323
This following setting has been configured. dtmfmode=rfc2833 has been set . set (Agi:async) sip client : jisti sip soft phone == issue : When Agent A1 and Agent A2 are on call DTMF must be Passed and Feature must be exeute . here DTMF detect but feaure not extecuted . in case of local

[asterisk-dev] Openais not supported in asterisk11.16

2015-03-12 Thread HKC323
Question : I want to get Device state from Different Asterisk Server Using Corosync and Openais . corosync-1.4.7 openais-1.1.4 nss nns-devel has been installed. i have refered [url]https://wiki.asterisk.org/wiki/display/AST/Distributed+Device+State +with+AIS[/url] i have configure Corosync ,

[asterisk-dev] confbridge struct (object) not release

2014-07-17 Thread hkc323
Asterisk 11.5.1 and Centos 6 module:app_confbridge.c Issue: How to delete confbridge struct in case of multiserver Confbridge using IAX2. NOTE:In single server there is no any such issue ? void conf_ended(struct conference_bridge *conference_bridge){ ao2_unlink(conference_bridges,

Re: [asterisk-dev] Bridge/0XXXXXX-in Bridge/0XXXXXX-

2014-07-09 Thread hkc323
hkc323 hkc323 at gmail.com writes: This problem has been solved now -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http

[asterisk-dev] Bridge/0XXXXXX-in Bridge/0XXXXXX-out channels did not hanguped after call hanguped .

2014-06-25 Thread hkc323
== Centos 6 Asterisk 11.5.1 app:Confbridge file:app_confbridge.c === issue : Bridge/0xb75254f channels did not hanguped after call hanguped . note1:After call hangup ast_bridge_change_state :2 which means Hangup note2:Patch also note

Re: [asterisk-dev] Asterisk-11.5.1 Confbridge Dailout using pbx_exce new user not placed into current conference

2014-04-29 Thread hkc323
hkc323 hkc323 at gmail.com writes: SOLVED NOW -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com

[asterisk-dev] Menufile did not played when user press * using Asterisk11.5.1 Confbridge

2014-04-22 Thread hkc323
Any Help ? ... Dialout user Pickuped/Answer call and merge into Confbridge but Admin getting Ringtone Asterisk-11.5.1 Confbridge . ? Expected : admin user (A 7002) ,of current Conference Dailout and Invite user (B 7001) to join Confernece. B Picked call and joined Confbridge. A and B

[asterisk-dev] Asterisk-11.5.1 Confbridge Dailout using pbx_exce new user not placed into current conference

2014-04-21 Thread hkc323
Task:Dialout from current running ConferenceBridge(eg.:1010101) and Add Sip user (eg:Sip/7001 on my newwork ) issue: We i Dailout with Dial app with help of func. pbx_exec User not merge current running ConferenceBridge(eg.:1010101) but create other new conferenceBridge. (eg :

Re: [asterisk-dev] Asterisk-11.5.1 Confbridge Dailout using pbx_exce new user not placed into current conference

2014-04-21 Thread hkc323
Matthew Jordan mjordan at digium.com writes: solved . strcat (dialstr,hg); strcat (dialstr,M); strcat (dialstr,(); strcat (dialstr,CONFDAILOUT); strcat