Transferring call to remote agent
Posted: Thu Jun 05, 2014 7:15 am
ViciBox Redux v.5.0.2-130821| VERSION: 2.8-432a BUILD: 140404-1007 | asterisk-1.8.3 | Single Server |4 port allo gsm card | No Extra Software After Installation | Intel(R) Core(TM) i5-3450 CPU @ 3.10GHz
hello every one this is my first post so let me know if any other information is required..
i want to transfer my call to mobile.(i have two gsm sim,gsm1 and gsm2)
what i do is, i make did 1234, set did rout to CALLMENU and set callmenu value to my menu (gjr_1) wit different option like
1 for product
2 for sales
3 to talk with bob.
In my call menu(gjr_1) when we press 3, it goes in ingroup(salesline) to our loged agent 9999, till here it work fine..
now for transferring the call to my mobile i make a remote agent 9999 and set External Extension:09278510651 (according to my dial plan)
when i make incoming call to gsm2 and press 3 to talk with bob, the call come to agent screen and got disconnect in next second without transferring it to my mobile. i check my cli> it try to transfer the call through gsm2 which already busy due to incoming call on it.
i dont understand what to do next..
please help me to solve this issue, i dont understand how to transfer the call to my mobile or how to use gsm1 to do this.
Thank you
cli>
[Jun 5 15:34:48] -- Executing [s@trunkinbound:1] Answer("GSM/2", "") in new stack
[Jun 5 15:34:48] event=OK
[Jun 5 15:34:49] -- Executing [s@trunkinbound:2] Goto("GSM/2", "trunkinbound,1234,1") in new stack
[Jun 5 15:34:49] -- Goto (trunkinbound,1234,1)
[Jun 5 15:34:49] -- Executing [1234@trunkinbound:1] AGI("GSM/2", "agi-DID_route.agi") in new stack
[Jun 5 15:34:49] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jun 5 15:34:49] -- <GSM/2>AGI Script agi-DID_route.agi completed, returning 0
[Jun 5 15:34:49] -- Executing [s@gjr_1:1] Answer("GSM/2", "") in new stack
[Jun 5 15:34:49] -- Executing [s@gjr_1:2] AGI("GSM/2", "agi-VDAD_inbound_calltime_check.agi,SALESLINE-----YES-----gjr_1-------------------------NO") in new stack
[Jun 5 15:34:49] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
[Jun 5 15:34:49] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:34:49] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:34:49] -- <GSM/2>AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
[Jun 5 15:34:49] -- Executing [s@gjr_1:3] Set("GSM/2", "INVCOUNT=0") in new stack
[Jun 5 15:34:49] -- Executing [s@gjr_1:4] BackGround("GSM/2", "welcome-main-forIV") in new stack
[Jun 5 15:34:49] -- <GSM/2> Playing 'welcome-main-forIV.gsm' (language 'en')
[Jun 5 15:35:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 5 15:35:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 5 15:35:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 5 15:35:03] DTMF[2116]: chan_gsm.c:3433 gsm_read_dtmf: DTMF Received is 3
[Jun 5 15:35:03] DTMF[8554]: channel.c:4066 __ast_read: DTMF end '3' received on GSM/2, duration 110 ms
[Jun 5 15:35:03] DTMF[8554]: channel.c:4135 __ast_read: DTMF end passthrough '3' on GSM/2
[Jun 5 15:35:03] -- Executing [3@gjr_1:1] AGI("GSM/2", "agi-VDAD_ALL_inbound.agi,CID-----LB-----SALESLINE-----gjr_1--------------------998----------TEST_IN------------------------------") in new stack
[Jun 5 15:35:03] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Jun 5 15:35:03] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:03] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:03] -- Playing 'call_transfer_agent' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 5 15:35:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 5 15:35:07] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:07] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- <GSM/2>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Jun 5 15:35:08] -- Executing [192*168*001*015*09278510651@default:1] Goto("GSM/2", "default,09278510651,1") in new stack
[Jun 5 15:35:08] -- Goto (default,09278510651,1)
[Jun 5 15:35:08] -- Executing [09278510651@default:1] Answer("GSM/2", "") in new stack
[Jun 5 15:35:08] -- Executing [09278510651@default:2] Dial("GSM/2", "GSM/2/09278510651,,To") in new stack
[Jun 5 15:35:08] DEBUG[8554]: chan_gsm.c:3878 gsm_request: [ALLO_GSM] gsm_data '2/09278510651'
[Jun 5 15:35:08] portstr:'2' ext:'09278510651' gsm_data '2/09278510651'
[Jun 5 15:35:08] WARNING[8554]: chan_gsm.c:3839 check_and_fetch_port: There is already a call on port :2
[Jun 5 15:35:08] ERROR[8554]: chan_gsm.c:3920 gsm_request: [ALLO_GSM] port may be invalid !!!!!!!!!!!
[Jun 5 15:35:08] WARNING[8554]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'GSM' (cause 0 - Unknown)
[Jun 5 15:35:08] == Everyone is busy/congested at this time (1:0/0/1)
[Jun 5 15:35:08] -- Executing [09278510651@default:3] Hangup("GSM/2", "") in new stack
[Jun 5 15:35:08] == Spawn extension (default, 09278510651, 3) exited non-zero on 'GSM/2'
[Jun 5 15:35:08] -- Executing [h@default:1] AGI("GSM/2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Jun 5 15:35:08] -- <GSM/2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jun 5 15:35:11] == Manager 'sendcron' logged off from 127.0.0.1
hello every one this is my first post so let me know if any other information is required..
i want to transfer my call to mobile.(i have two gsm sim,gsm1 and gsm2)
what i do is, i make did 1234, set did rout to CALLMENU and set callmenu value to my menu (gjr_1) wit different option like
1 for product
2 for sales
3 to talk with bob.
In my call menu(gjr_1) when we press 3, it goes in ingroup(salesline) to our loged agent 9999, till here it work fine..
now for transferring the call to my mobile i make a remote agent 9999 and set External Extension:09278510651 (according to my dial plan)
when i make incoming call to gsm2 and press 3 to talk with bob, the call come to agent screen and got disconnect in next second without transferring it to my mobile. i check my cli> it try to transfer the call through gsm2 which already busy due to incoming call on it.
i dont understand what to do next..
please help me to solve this issue, i dont understand how to transfer the call to my mobile or how to use gsm1 to do this.
Thank you
cli>
[Jun 5 15:34:48] -- Executing [s@trunkinbound:1] Answer("GSM/2", "") in new stack
[Jun 5 15:34:48] event=OK
[Jun 5 15:34:49] -- Executing [s@trunkinbound:2] Goto("GSM/2", "trunkinbound,1234,1") in new stack
[Jun 5 15:34:49] -- Goto (trunkinbound,1234,1)
[Jun 5 15:34:49] -- Executing [1234@trunkinbound:1] AGI("GSM/2", "agi-DID_route.agi") in new stack
[Jun 5 15:34:49] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-DID_route.agi
[Jun 5 15:34:49] -- <GSM/2>AGI Script agi-DID_route.agi completed, returning 0
[Jun 5 15:34:49] -- Executing [s@gjr_1:1] Answer("GSM/2", "") in new stack
[Jun 5 15:34:49] -- Executing [s@gjr_1:2] AGI("GSM/2", "agi-VDAD_inbound_calltime_check.agi,SALESLINE-----YES-----gjr_1-------------------------NO") in new stack
[Jun 5 15:34:49] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_inbound_calltime_check.agi
[Jun 5 15:34:49] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:34:49] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:34:49] -- <GSM/2>AGI Script agi-VDAD_inbound_calltime_check.agi completed, returning 0
[Jun 5 15:34:49] -- Executing [s@gjr_1:3] Set("GSM/2", "INVCOUNT=0") in new stack
[Jun 5 15:34:49] -- Executing [s@gjr_1:4] BackGround("GSM/2", "welcome-main-forIV") in new stack
[Jun 5 15:34:49] -- <GSM/2> Playing 'welcome-main-forIV.gsm' (language 'en')
[Jun 5 15:35:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 5 15:35:01] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 5 15:35:01] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 5 15:35:03] DTMF[2116]: chan_gsm.c:3433 gsm_read_dtmf: DTMF Received is 3
[Jun 5 15:35:03] DTMF[8554]: channel.c:4066 __ast_read: DTMF end '3' received on GSM/2, duration 110 ms
[Jun 5 15:35:03] DTMF[8554]: channel.c:4135 __ast_read: DTMF end passthrough '3' on GSM/2
[Jun 5 15:35:03] -- Executing [3@gjr_1:1] AGI("GSM/2", "agi-VDAD_ALL_inbound.agi,CID-----LB-----SALESLINE-----gjr_1--------------------998----------TEST_IN------------------------------") in new stack
[Jun 5 15:35:03] -- Launched AGI Script /usr/share/asterisk/agi-bin/agi-VDAD_ALL_inbound.agi
[Jun 5 15:35:03] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:03] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:03] -- Playing 'call_transfer_agent' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:06] == Manager 'sendcron' logged on from 127.0.0.1
[Jun 5 15:35:06] == Manager 'sendcron' logged off from 127.0.0.1
[Jun 5 15:35:07] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:07] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- Playing 'sip-silence' (escape_digits=) (sample_offset 0)
[Jun 5 15:35:08] -- <GSM/2>AGI Script agi-VDAD_ALL_inbound.agi completed, returning 0
[Jun 5 15:35:08] -- Executing [192*168*001*015*09278510651@default:1] Goto("GSM/2", "default,09278510651,1") in new stack
[Jun 5 15:35:08] -- Goto (default,09278510651,1)
[Jun 5 15:35:08] -- Executing [09278510651@default:1] Answer("GSM/2", "") in new stack
[Jun 5 15:35:08] -- Executing [09278510651@default:2] Dial("GSM/2", "GSM/2/09278510651,,To") in new stack
[Jun 5 15:35:08] DEBUG[8554]: chan_gsm.c:3878 gsm_request: [ALLO_GSM] gsm_data '2/09278510651'
[Jun 5 15:35:08] portstr:'2' ext:'09278510651' gsm_data '2/09278510651'
[Jun 5 15:35:08] WARNING[8554]: chan_gsm.c:3839 check_and_fetch_port: There is already a call on port :2
[Jun 5 15:35:08] ERROR[8554]: chan_gsm.c:3920 gsm_request: [ALLO_GSM] port may be invalid !!!!!!!!!!!
[Jun 5 15:35:08] WARNING[8554]: app_dial.c:2345 dial_exec_full: Unable to create channel of type 'GSM' (cause 0 - Unknown)
[Jun 5 15:35:08] == Everyone is busy/congested at this time (1:0/0/1)
[Jun 5 15:35:08] -- Executing [09278510651@default:3] Hangup("GSM/2", "") in new stack
[Jun 5 15:35:08] == Spawn extension (default, 09278510651, 3) exited non-zero on 'GSM/2'
[Jun 5 15:35:08] -- Executing [h@default:1] AGI("GSM/2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----16-----CHANUNAVAIL----------") in new stack
[Jun 5 15:35:08] -- <GSM/2>AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
[Jun 5 15:35:11] == Manager 'sendcron' logged off from 127.0.0.1