trunk switch probleme

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trunk switch probleme

Postby brett05 » Tue Jun 30, 2009 11:16 pm

hi
i have two trunk with two number phone here my config

register => user:password@keyyo.net:5060

[trunk_2]
type=friend
username=user
fromuser=user
fromdomain=keyyo.net
secret=password
host=83.136.161.72
qualify=300
dtmfmode=auto
canreinvite=no
nat=yes
disallow=all
allow=alaw

SIPtrunk2=SIP/trunk_2

exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},,To)
exten => _9XXXXXXXXXX,3,Hangup


the second trunk :

register => user2:password2@keyyo.net:5060

[trunk_3]
type=friend
username=user2
fromuser=user2
fromdomain=keyyo.net
secret=password2
host=83.136.161.72
qualify=300
dtmfmode=auto
canreinvite=no
nat=yes
disallow=all
allow=alaw

SIPtrunk3=SIP/trunk_3

exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk3}/${EXTEN:1},,To)
exten => _9XXXXXXXXXX,3,Hangup


the two is registred good also i can dial with them good
here :
vici*CLI> sip show peers
Name/username Host Dyn Nat ACL Port Status
101/101 192.168.1.3 D N 22722 Unmonitored
100/100 192.168.1.2 D N 54288 Unmonitored
trunk_3/33974761536 83.136.161.72 N 5060 OK (158 ms)
trunk_2/33974760973 83.136.161.72 N 5060 OK (158 ms)
5 sip peers [5 online , 0 offline]

also i have two phone connected good with 101 and 100
ok here what is my probleme :
when i use manual dial or auto dial with " ratio+1 or 2 or 3 as Dial Method" or if i use the "ADAPT_TAPERED with 1 dial methode "
then when i enter with 2 machine and two softphone in same time i see this:
-- Executing AGI("Local/90557692216@default-1926,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/90557692216@default-1926,2", "SIP/trunk_2/0557692216||To") in new stack
-- Called trunk_2/0557692216
-- SIP/trunk_2-0937dae0 is making progress passing it to Local/90557692216@default-1926,2
-- SIP/trunk_2-0937dae0 is ringing
-- parse_srv: SRV mapped to host prx2.sip.keyyo.net, port 5060
-- parse_srv: SRV mapped to host prx2.sip.keyyo.net, port 5060
== Spawn extension (default, 90557692216, 2) exited non-zero on 'Local/90557692216@default-1926,2'
-- Executing DeadAGI("Local/90557692216@default-1926,2", "agi://127.0.0.1:4577/call_log--HVcauses--PRI-----NODEBUG-----0-----CANCEL----------") in new stack
== Manager 'sendcron' logged off from 127.0.0.1
-- AGI Script agi://127.0.0.1:4577/call_log--HVcauses ... ---------- completed, returning 0
== Parsing '/etc/asterisk/manager.conf': Found
== Manager 'sendcron' logged on from 127.0.0.1
-- Executing AGI("Local/90557691375@default-7aa5,2", "agi://127.0.0.1:4577/call_log") in new stack
-- AGI Script agi://127.0.0.1:4577/call_log completed, returning 0
-- Executing Dial("Local/90557691375@default-7aa5,2", "SIP/trunk_2/0557691375||To") in new stack
-- Called trunk_2/0557691375
-- SIP/trunk_2-0937dae0 is making progress passing it to Local/90557691375@default-7aa5,2
-- SIP/trunk_2-0937dae0 is ringing

the softphoe "100" is ringing good with trunk_2 and the other softphone is stille waiting when softphone "100" finish his call to take the same trunk_2.
and my second sip trunk "trunk_3" is always avaible to make call too.
why my server use only one sip trunk to make the call for my two softphone in same time and he forget my second sip trunk to use it too.
thanks to answer me please
Last edited by brett05 on Wed Jul 01, 2009 1:05 am, edited 2 times in total.
brett05
 
Posts: 571
Joined: Sun May 24, 2009 5:48 pm
Location: tunisia

trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 12:48 am

please friend i need all your help because i have buy a second ligne sip and i want to use it with my two computer in same time with my two softphone.
and my provider say the probleme come from me ? it is true
when i try to active only sip trunk in admin --> carrier for exemple trunk_2 or trunk_3 they work and they dial correctly .
but when i active them the two only the one sip trunk try to make dial for my two softphone and i need to wait the first call finished to make the second one for the second softphone !!!!!
and the second trunk is still not used
it is strange
really with this forum i have learned many thing
thanks vicidial and big kiss
brett05
 
Posts: 571
Joined: Sun May 24, 2009 5:48 pm
Location: tunisia

trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 12:51 am

this is my file
extensions-vicidial.conf
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
SIPtrunk2=SIP/trunk_2
SIPtrunk3=SIP/trunk_3

[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... ----------)

; Local Server: 192.168.1.10
exten => _192*168*001*010*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: keyyo1 - keyyo
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Hangup

; VICIDIAL Carrier: keyyo2 - keyyo
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk3}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Hangup


exten => 100,1,Dial(SIP/100)
exten => 100,2,Voicemail,u100
exten => 101,1,Dial(SIP/101)
exten => 101,2,Voicemail,u101



and extensions.conf
[general]
static=yes
writeprotect=no

[globals]
CONSOLE=Console/dsp ; Console interface for demo
;TRUNK=Zap/g1 ; Trunk interface
;TRUNKX=Zap/g2 ; 2nd trunk interface
;TRUNKIAX=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
;TRUNKIAX1=IAX2/ASTtest1:test@10.10.10.16:4569 ; IAX trunk interface
;TRUNKBINFONE=IAX2/1112223333:PASSWORD@iax.binfone.com ; IAX trunk interface
;SIPtrunk=SIP/1234:PASSWORD@sip.provider.net ; SIP trunk
TRUNKloop = IAX2/ASTloop:test@127.0.0.1:40569 ; used for blind monitoring
TRUNKblind = IAX2/ASTblind:test@127.0.0.1:41569 ; used for testing


#include extensions-vicidial.conf

[trunkinbound]
; agent dial-in:
exten => 2345,1,Answer ; Answer the line
exten => 2345,2,AGI(agi-AGENT_dial_in.agi)
exten => 2345,3,Hangup

; DID call routing process
exten => _X.,1,AGI(agi-DID_route.agi)

; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})



[default]
include => vicidial-auto

; Local agent alert extensions
exten => _8600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
exten => _78600XXX*.,1,AGI(agi-VDADfixCXFER.agi)
; Local blind monitoring
exten => _08600XXX,1,Dial(${TRUNKblind}/6${EXTEN:1},55,To)


;;;;;;;;;; BEGIN Voicemail and Prompts Section ;;;;;;;;;;;;;;;;;;;;;;;
; Give voicemail at extension 8500
exten => 8500,1,VoicemailMain
exten => 8500,2,Goto(s,6)

; this is used to allow the GUI to send you directly into voicemail
; don't forget to set GUI variable $voicemail_exten to this extension
exten => 8501,1,VoicemailMain(s${CALLERIDNUM})
exten => 8501,2,Hangup

; this is used to allow the GUI to send live calls directly into voicemail
; don't forget to set GUI variable $voicemail_dump_exten to this extension
exten => _85026666666666.,1,Wait(1)
exten => _85026666666666.,2,Voicemail(${EXTEN:14}|u)
exten => _85026666666666.,3,Hangup

; prompts for recording AGI script, ID is 4321
; first variable is format (gsm/wav)
; second variable is timeout in milliseconds (default is 720000 [12 minutes])
exten => 8167,1,Answer
exten => 8167,2,AGI(agi-record_prompts.agi,wav-----720000)
exten => 8167,3,Hangup
exten => 8168,1,Answer
exten => 8168,2,AGI(agi-record_prompts.agi,gsm-----720000)
exten => 8168,3,Hangup

; playback of recorded prompts
exten => _851XXXXX,1,Answer
exten => _851XXXXX,2,Playback(${EXTEN})
exten => _851XXXXX,3,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
exten => _7851XXXXX,1,WaitForSilence(2000,2) ; AMD got machine. leave message after recording
exten => _7851XXXXX,2,Playback(${EXTEN:1})
exten => _7851XXXXX,3,AGI(VD_amd_post.agi,${EXTEN:1})
exten => _7851XXXXX,4,Hangup


;;;;;;;;;; END Voicemail and Prompts Section ;;;;;;;;;;;;;;;;;;;;;;;;;


; FastAGI for VICIDIAL/astGUIclient call logging
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})


; Example phone extensions

; Extension 2000 Sipura/Linksys ATA line 1
exten => 2000,1,Dial(sip/spa2000,30,to) ; Ring, 30 secs max
exten => 2000,2,Voicemail,u2000 ; Send to voicemail...
; Extension 2001 Sipura/Linksys ATA line 2
exten => 2001,1,Dial(sip/spa2001,30,to) ; Ring, 30 secs max
exten => 2001,2,Voicemail,u2001 ; Send to voicemail...
; Extension 2102 rings Grandstream phone
exten => 2102,1,Dial(sip/gs102,30,to) ; Ring, 30 secs max
exten => 2102,2,Voicemail,u2102 ; Send to voicemail...
; Extension 401 rings the firefly softphone
exten => 401,1,Dial((IAX2/firefly01@firefly01/s||t)
exten => 401,2,Hangup

; 100-350 phone extensions now auto-generated
; extensions for other SIP and IAX call center phones
; cc100-cc150 SIP Phones
;exten => _1[0-5]X,1,Dial(sip/cc${EXTEN},20,to)
; cc300-cc350 IAX Phones
;exten => _3[0-5]X,1,Dial(IAX2/cc${EXTEN},20,to)

; extensions if using a T1 channelbank
exten => _19XX,1,Dial(Zap/${EXTEN:2},30,o)
exten => _19XX,2,Hangup

; Extension 4001 rings Zap phone (this example for FXS on Zap port 1)
exten => 4001,1,Dial(Zap/1,30,o) ; ring Zap device 1
exten => 4001,2,Voicemail,u4001 ; Send to voicemail...


; # timeout invalid rules
exten => #,1,Playback(invalid) ; "Thanks for trying the demo"
exten => #,2,Hangup ; Hang them up.
exten => t,1,Goto(#,1) ; If they take too long, give up
exten => i,1,Playback(invalid) ; "That's not valid, try again"


; Inbound call from BINFONE
; exten => 1112223333,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => 1112223333,2,Dial(sip/gs102,55,o)
; exten => 1112223333,3,Hangup

; Extension 7275551212 - Inbound local number from PRI with 10 digit delivery
exten => 7275551212,1,Ringing
exten => 7275551212,2,Wait(1)
exten => 7275551212,3,AGI(agi://127.0.0.1:4577/call_log--fullCID--${EXTEN}-----${CALLERID}-----${CALLERIDNUM}-----${CALLERIDNAME})
exten => 7275551212,4,Answer
exten => 7275551212,5,Dial(sip/spa2000&sip/spa2001,30,To)
exten => 7275551212,6,Voicemail,u2000

; dial a long distance outbound number to the UK
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten => _901144XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _901144XXXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},55,To)
exten => _901144XXXXXXXXXX,3,Hangup

; dial a long distance outbound number to Australia
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls,
exten => _901161XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _901161XXXXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
exten => _901161XXXXXXXXX,3,Hangup

; Extensions for performance testing
exten => _91999NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91999NXXXXXX,2,Dial(${TRUNKloop}/${EXTEN:2},,tTo)
exten => _91999NXXXXXX,3,Hangup
exten => 999999999999,1,AGI(agi://127.0.0.1:4577/call_log)
exten => 999999999999,2,Dial(${TRUNKloop}/${EXTEN:1},,tTo)
exten => 999999999999,3,Hangup

; dial an 800 outbound number
exten => _91800NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91800NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91800NXXXXXX,3,Hangup
exten => _91888NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91888NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91888NXXXXXX,3,Hangup
exten => _91877NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91877NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91877NXXXXXX,3,Hangup
exten => _91866NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91866NXXXXXX,2,Dial(${TRUNK}/${EXTEN:1},,To)
exten => _91866NXXXXXX,3,Hangup

; dial a local outbound number (modified because of only LD T1)
exten => _9NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9NXXXXXX,2,Dial(${TRUNK}/1727${EXTEN:1},,To)
exten => _9NXXXXXX,3,Hangup

; dial a local 727 outbound number with area code
exten => _9727NXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9727NXXXXXX,2,Dial(${TRUNK}/1${EXTEN:1},,To)
exten => _9727NXXXXXX,3,Hangup

; dial a long distance outbound number
; This 'o' Dial flag is VERY important for VICIDIAL on outbound calls
exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _91NXXNXXXXXX,2,Dial(${TRUNKX}/${EXTEN:1},,To)
exten => _91NXXNXXXXXX,3,Hangup

; This is a loopback dialaround to allow for hearing of ringing for 3way calls
exten => _881NXXNXXXXXX,1,Answer
exten => _881NXXNXXXXXX,2,Dial(${TRUNKloop}/9${EXTEN:2},,To)
exten => _881NXXNXXXXXX,3,Hangup

; dial a long distance outbound number through BINFONE
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(${TRUNKIAX}/${EXTEN:1},55,To)
; exten => _91NXXNXXXXXX,3,Hangup
; dial a long distance outbound number through a SIP provider
; exten => _91NXXNXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91NXXNXXXXXX,2,Dial(sip/${EXTEN:1}@SIPtrunk,55,o)
; exten => _91NXXNXXXXXX,3,Hangup
; special extensions for North America to catch invalid phone numbers
; exten => _91XXX[0-1]XXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXX[0-1]XXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXX[0-1]XXXXXX,3,Hangup
; exten => _91[0-1]XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91[0-1]XXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91[0-1]XXXXXXXXX,3,Hangup
; exten => _91XXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXX,3,Hangup
; exten => _91XXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXX,3,Hangup
; exten => _91XXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXXXX,3,Hangup
; exten => _91XXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXXXXX,3,Hangup
; exten => _91XXXXXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
; exten => _91XXXXXXXXXXXXX,2,Dial(${TRUNKloop}/9990011112,,to)
; exten => _91XXXXXXXXXXXXX,3,Hangup

; parameters for call_inbound.agi (7 fields separated by five dashes "-----"):
; 1. the extension of the phone to ring as defined in the asterisk.phones table
; 2. the phone number that was called, for the live_inbound/_log entry
; 3. a text description of the number that was called in
; 4-7. optional fields, they are also passed as fields in the GUI to web browser
; This is not part of VICIDIAL, it is for astGUIclient agent use only

; Extension 3429 - Inbound 800 number (1-800-555-3429) example of RBS T1
; with 10 digit ANI and 4 digit DNIS star separated
exten => _**3429,1,Ringing
exten => _**3429,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _**3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _**3429,4,Answer
exten => _**3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
exten => _**3429,6,Voicemail,u2000
; Extension 3429 - with ANI [callerID]
exten => _*NXXNXXXXXX*3429,1,Ringing
exten => _*NXXNXXXXXX*3429,2,AGI(agi://127.0.0.1:4577/call_log)
exten => _*NXXNXXXXXX*3429,3,AGI(call_inbound.agi,spa2000-----8005553429-----Inbound 800-----x-----y-----z-----w)
exten => _*NXXNXXXXXX*3429,4,Answer
exten => _*NXXNXXXXXX*3429,5,Dial(sip/spa2000&sip/spa2001,30,to)
exten => _*NXXNXXXXXX*3429,6,Voicemail,u2000


; parameters for agi-VDAD_ALL_inbound.agi (9 fields separated by five dashes "-----"):
; 1. the method of call handling for the script:
; - CID - CID received, add record with phone number
; - CIDLOOKUP - Lookup CID to find record in whole system
; - CIDLOOKUPRL - Restrict lookup to one list
; - CIDLOOKUPRC - Restrict lookup to one campaign's lists
; - CLOSER - Closer calls from VICIDIAL fronters
; - ANI - ANI received, add record with phone number
; - ANILOOKUP - Lookup ANI to find record in whole system
; - ANILOOKUPRL - Restrict lookup to one list
; - 3DIGITID - Enter 3 digit code to go to agent
; - 4DIGITID - Enter 4 digit code to go to agent
; - 5DIGITID - Enter 5 digit code to go to agent
; - 10DIGITID - Enter 10 digit code to go to agent
; 2. the method of searching for an available agent:
; - LO - Load Balance Overflow only (priority to home server)
; - LB - <default> Load Balance total system
; - SO - Home server only
; 3. the full name of the IN GROUP to be used in vicidial for the inbound call
; 4. the phone number that was called, for the log entry
; 5. the callerID or lead_id of the person that called(usually overridden)
; 6. the park extension audio file name if used
; 7. the status of the call initially(usually not used)
; 8. the list_id to insert the new lead under if it is new (and CID/ANI available)
; 9. the phone dialing code to insert with the new lead if new (and CID/ANI available)
; 10. the campaign_id to search within lists if CIDLOOKUPRC
; 11. the user to queue the call to for AGENTDIRECT in-group calls
; inbound VICIDIAL call with CID delivery through T1 PRI
exten => 1234,1,Answer ; Answer the line
exten => 1234,2,AGI(agi-VDAD_ALL_inbound.agi,CID-----LB-----CL_GALLERIA-----7274515134-----Closer-----park----------999-----1)
exten => 1234,3,Hangup

; inbound VICIDIAL transfer calls [can arrive through PRI T1 crossover, IAX or SIP channel]
exten => _90009.,1,Answer ; Answer the line
exten => _90009.,2,Dial(${TRUNKloop}/9${EXTEN},,to)
exten => _90009.,3,Hangup
exten => _990009.,1,Answer ; Answer the line
exten => _990009.,2,AGI(agi-VDAD_ALL_inbound.agi,CLOSER-----LB-----CL_TESTCAMP-----7275551212-----Closer-----park----------999-----1)
exten => _990009.,3,Hangup
; DID forwarded calls
exten => _99909*.,1,Answer
exten => _99909*.,2,AGI(agi-VDAD_ALL_inbound.agi)
exten => _99909*.,3,Hangup


; barge monitoring extension
exten => 8159,1,ZapBarge
exten => 8159,2,Hangup

; ZapBarge direct channel extensions
exten => _86120XX,1,ZapBarge(${EXTEN:5})


exten => _X48600XXX,1,MeetMeAdmin(${EXTEN:2},T,${EXTEN:0:1})
exten => _X48600XXX,2,Hangup

exten => _X38600XXX,1,MeetMeAdmin(${EXTEN:2},t,${EXTEN:0:1})
exten => _X38600XXX,2,Hangup

exten => _X28600XXX,1,MeetMeAdmin(${EXTEN:2},m,${EXTEN:0:1})
exten => _X28600XXX,2,Hangup

exten => _X18600XXX,1,MeetMeAdmin(${EXTEN:2},M,${EXTEN:0:1})
exten => _X18600XXX,2,Hangup

exten => _55558600XXX,1,MeetMeAdmin(${EXTEN:4},K)
exten => _55558600XXX,2,Hangup
exten => 8300,1,Hangup

; astGUIclient conferences
exten => _86000[0-4]X,1,Meetme,${EXTEN}|q
; VICIDIAL conferences
exten => _86000[5-9]X,1,Meetme,${EXTEN}|F
exten => _8600[1-2]XX,1,Meetme,${EXTEN}|F
; quiet entry and leaving conferences for VICIDIAL (inbound announce and SendDTMF)
exten => _78600XXX,1,Meetme,${EXTEN:1}|Fq
; quiet monitor-only extensions for meetme rooms (for room managers)
exten => _68600XXX,1,Meetme,${EXTEN:1}|Fmq
; quiet monitor-only entry and leaving conferences for VICIDIAL (recording)
exten => _58600XXX,1,Meetme,${EXTEN:1}|Fmq

; voicelab exten
exten => _86009XX,1,Meetme,${EXTEN}|Fmq
; voicelab exten moderator
exten => _986009XX,1,Meetme,${EXTEN:1}



; park channel for client GUI parking, hangup after 30 minutes
; create a GSM formatted audio file named "park.gsm" that is 30 minutes long
; and put it in /var/lib/asterisk/sounds
exten => 8301,1,Answer
exten => 8301,2,AGI(park_CID.agi)
exten => 8301,3,Playback(park)
exten => 8301,4,Hangup
exten => 8303,1,Answer
exten => 8303,2,AGI(park_CID.agi)
exten => 8303,3,Playback(conf)
exten => 8303,4,Hangup

; park channel for client GUI conferencing, hangup after 30 minutes
; create a GSM formatted audio file named "conf.gsm" that is 30 minutes long
; and put it in /var/lib/asterisk/sounds
exten => 8302,1,Answer
exten => 8302,2,Playback(conf)
exten => 8302,3,Hangup

exten => 8304,1,Answer
exten => 8304,2,Playback(ding)
exten => 8304,3,Hangup

; default audio for safe harbor 2-second-after-hello message then hangup
; create a GSM formatted audio file complies with safe harbor rules
; and put it in /var/lib/asterisk/sounds then change filename below
exten => 8307,1,Answer
exten => 8307,2,Playback(vm-goodbye)
exten => 8307,3,Hangup

; this is used for recording conference calls, the client app sends the filename
; value as a callerID recordings go to /var/spool/asterisk/monitor (WAV)
; Recording is limited to 1 hour, to make longer, just change the Wait,3600
exten => 8309,1,Answer
exten => 8309,2,Monitor(wav,${CALLERID(name)})
exten => 8309,3,Wait,3600
exten => 8309,4,Hangup

; this is used for recording conference calls, the client app sends the filename
; value as a callerID recordings go to /var/spool/asterisk/monitor (GSM)
; Recording is limited to 1 hour, to make longer, just change the Wait,3600
exten => 8310,1,Answer
exten => 8310,2,Monitor(gsm,${CALLERIDNAME})
exten => 8310,3,Wait,3600
exten => 8310,4,Hangup

; this is used for playing a message to an answering machine forwarded from AMD in VICIDIAL
; replace conf with the message file you want to leave
exten => 8320,1,WaitForSilence(2000,2) ; AMD got machine. leave message after recording
exten => 8320,2,Playback(conf)
exten => 8320,3,AGI(VD_amd_post.agi,${EXTEN})
exten => 8320,4,Hangup

; use for selective CallerID hangup by area code(hard-coded)
exten => 8352,1,AGI(agi-VDADselective_CID_hangup.agi,${EXTEN})
exten => 8352,2,Playback(safe_harbor)
exten => 8352,3,Hangup

; this is used for sending DTMF signals within conference calls, the client app
; sends the digits to be played in the callerID field
; sound files must be placed in /var/lib/asterisk/sounds
exten => 8500998,1,Answer
exten => 8500998,2,Playback(silence)
exten => 8500998,3,AGI(agi-dtmf.agi)
exten => 8500998,4,Hangup

; multi-remote-monitor entry extensions
exten => 8162,1,Dial(${TRUNKblind}/34567890123456789,55,to)

exten => 34567890123456789,1,Answer
exten => 34567890123456789,2,Goto(monitor,s,1)

;#### VDAD STANDARD TRANSFER ENTRIES ####
; Below are the parameters needed for the agi-VDAD_ALL_outbound.agi script to be run properly
; 1. the method of call handling for the script:
; - NORMAL - <default> Standard outbound routing to agent
; - TEST - For performance testing only
; - BROADCAST - For no-agent broadcast dialing
; - SURVEY - For survery question then on to agent
; - REMINDER - Reminder campaign
; - REMINDX - Reminder with transfer to agent
; 2. the method of searching for an available agent:
; - LB - <default> Load Balance total system
; - LO - Load Balance Overflow only (priority to home server)
; - SO - Home server only
; 3. the sound file to play when doing a SURVEY, REMINDER, REMINDX campaign
; 4. the acceptible dtmf digits for a SURVEY
; 5. the out-opt digit for a SURVEY (must be in the digit map)
; 6. the sound file to play for a SURVEY when transfering to an agent
; 7. the sound file to play for a SURVEY when DNCing the call
; 8. OPTIN or OPTOUT: if OPTIN call is only sent to agent with button press
; if OPTOUT call is sent to agent if no button press at all
; 9. the status that is use for a SURVEY when someone opts out
; if the status is DNC it will also add them to the internal dnc table

; VICIDIAL_auto_dialer transfer script for no-agent campaigns:
exten => 8364,1,Playback(sip-silence)
exten => 8364,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8364,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8364,5,Hangup

; VICIDIAL_auto_dialer transfer script:
exten => 8365,1,Playback(sip-silence)
exten => 8365,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8365,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----SO)
exten => 8365,5,Hangup

; VICIDIAL_auto_dialer transfer script SURVEY at beginning:
exten => 8366,1,Playback(sip-silence)
exten => 8366,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8366,3,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,4,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8366,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balance Overflow:
exten => 8367,1,Playback(sip-silence)
exten => 8367,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8367,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LO)
exten => 8367,5,Hangup

; VICIDIAL_auto_dialer transfer script Load Balanced:
exten => 8368,1,Playback(sip-silence)
exten => 8368,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8368,3,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,4,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8368,5,Hangup

; VICIDIAL_auto_dialer transfer script AMD with Load Balanced:
exten => 8369,1,Playback(sip-silence)
exten => 8369,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8369,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8369,4,AGI(VD_amd.agi,${EXTEN})
exten => 8369,5,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,6,AGI(agi-VDAD_ALL_outbound.agi,NORMAL-----LB)
exten => 8369,7,Hangup

; VICIDIAL auto-dial reminder script
exten => 8372,1,Playback(sip-silence)
exten => 8372,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8372,3,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,4,AGI(agi-VDADautoREMINDER.agi,${EXTEN})
exten => 8372,5,Hangup

; VICIDIAL SURVEY transfer script AMD with Load Balanced:
exten => 8373,1,Playback(sip-silence)
exten => 8373,2,AGI(agi://127.0.0.1:4577/call_log)
exten => 8373,3,AMD(2000|2000|1000|5000|120|50|4|256)
exten => 8373,4,AGI(VD_amd.agi,${EXTEN})
exten => 8373,5,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,6,AGI(agi-VDAD_ALL_outbound.agi,SURVEYCAMP-----LB)
exten => 8373,7,Hangup




; PERFORMANCE TESTING
exten => _999XXXXXX1,1,Answer
exten => _999XXXXXX1,2,Wait(2)
exten => _999XXXXXX1,3,Playback(vicidial-welcome)
exten => _999XXXXXX1,4,Hangup

exten => _999XX11112,1,Wait(2)
exten => _999XX11112,2,Answer
exten => _999XX11112,3,Playback(ss-noservice)
exten => _999XX11112,4,Playback(vm-goodbye)
exten => _999XX11112,5,Hangup

exten => _999XX18112,1,Wait(2)
exten => _999XX18112,2,Answer
exten => _999XX18112,3,Playback(vtiger-fax)
exten => _999XX18112,4,Playback(vtiger-fax)
exten => _999XX18112,5,Hangup

exten => _999XX19112,1,Wait(2)
exten => _999XX19112,2,Answer
exten => _999XX19112,3,Playback(vtiger-email)
exten => _999XX19112,4,Playback(vtiger-email)
exten => _999XX19112,5,Hangup

exten => _999XXXX112,1,Wait(5)
exten => _999XXXX112,2,Answer
exten => _999XXXX112,3,Playback(demo-instruct)
exten => _999XXXX112,4,Playback(demo-instruct)
exten => _999XXXX112,5,Hangup

exten => _999XXXXXX2,1,Wait(8)
exten => _999XXXXXX2,2,Answer
exten => _999XXXXXX2,3,Playback(demo-instruct)
exten => _999XXXXXX2,4,Hangup

exten => _999XXXXXX3,1,SetVar(PRI_CAUSE=1)
exten => _999XXXXXX3,2,Hangup

exten => _999XXXXXX4,1,SetVar(PRI_CAUSE=27)
exten => _999XXXXXX4,2,Hangup

exten => _999XXXXXX5,1,Wait(60)
exten => _999XXXXXX5,2,Hangup

exten => _999XXXXXX6,1,Wait(10)
exten => _999XXXXXX6,2,Answer
exten => _999XXXXXX6,3,Playback(demo-instruct)
exten => _999XXXXXX6,4,Hangup

exten => _999XXXXXX7,1,Wait(12)
exten => _999XXXXXX7,2,Answer
exten => _999XXXXXX7,3,Playback(demo-enterkeywords)
exten => _999XXXXXX7,4,Hangup

exten => _999XXXXXX8,1,SetVar(PRI_CAUSE=17)
exten => _999XXXXXX8,2,Hangup

exten => _999XXXXXX9,1,Wait(6)
exten => _999XXXXXX9,2,Answer
exten => _999XXXXXX9,3,Playback(demo-abouttotry)
exten => _999XXXXXX9,4,Hangup

exten => _999XXXXXX0,1,Wait(5)
exten => _999XXXXXX0,2,Answer
exten => _999XXXXXX0,3,Playback(vm-goodbye)
exten => _999XXXXXX0,4,Hangup

exten => 99999999999,1,Answer
exten => 99999999999,2,Playback(conf)
exten => 99999999999,3,Playback(conf)
exten => 99999999999,4,Playback(conf)
exten => 99999999999,5,Playback(conf)
exten => 99999999999,6,Playback(conf)
exten => 99999999999,7,Playback(conf)
exten => 99999999999,8,Playback(conf)
exten => 99999999999,9,Playback(conf)
exten => 99999999999,10,Playback(conf)
exten => 99999999999,11,Playback(conf)
exten => 99999999999,12,Playback(conf)
exten => 99999999999,13,Playback(conf)
exten => 99999999999,14,Hangup


[monitor]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => s,1,SetVar(TIMEOUT(digit)=10)
exten => s,n,SetVar(TIMEOUT(response)=10)
exten => s,n,SetVar(MEETME_EXIT_CONTEXT=monitor_exit)
exten => s,n,Background(vm-extension) ; need audio prompt.
exten => s,n,WaitExten(10)

exten => i,1,Goto(monitor_exit,s,1)
exten => #,1,Goto(monitor_exit,s,1)
exten => t,1,Goto(monitor_exit,s,1)

exten => _8[0-2]XX,1,Meetme(8600${EXTEN:1},FmqX) ; Listen
exten => _99[0-2]XX,1,Meetme(8600${EXTEN:2},FX) ; Barge

[monitor_exit]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... EBUG-----${HANGUPCAUSE}-----${DIALSTATUS}-----${DIALEDTIME}-----${ANSWEREDTIME})

exten => _X,1,Goto(monitor,s,1)

exten => i,1,Goto(monitor,s,1)
exten => #,1,Goto(monitor,s,1)
exten => t,1,Goto(monitor,s,1)
brett05
 
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trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 4:02 pm

over 26 view and no one can answer me ?
really i need it please [/u]
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Location: tunisia

Postby mflorell » Wed Jul 01, 2009 6:06 pm

Two hits against you:
- your postings are hard to understand
- you posted your entire dialplan without being asked for it

what kind of SIP accounts do you have?
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Location: Florida

Re: trunk switch probleme

Postby Geil21 » Wed Jul 01, 2009 6:11 pm

You have a conflict on your dial plan

; VICIDIAL Carrier: keyyo1 - keyyo
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Hangup

; VICIDIAL Carrier: keyyo2 - keyyo
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk3}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Hangup


These entries have the same extension and priorities so one is going to override the other.

If you want both carriers to work within a single dial rule this might just get it done

Just combine the dial plan entries in the first trunk and leave the second blank

Try this change

; VICIDIAL Carrier: keyyo1 - keyyo
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Dial(${SIPtrunk3}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,4,Hangup

; VICIDIAL Carrier: keyyo2 - keyyo


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trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 6:36 pm

so i need to go to admin-->carriier and put

exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Dial(${SIPtrunk3}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,4,Hangup

in each carrier or
i need to go to
etc/asterisk/extensions-vicidial.conf and put this line
???
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Location: tunisia

Postby brett05 » Wed Jul 01, 2009 6:38 pm

mflorell wrote:Two hits against you:
- your postings are hard to understand
- you posted your entire dialplan without being asked for it

what kind of SIP accounts do you have?



i have a sip account from keyyo
he is a french provider
and i have buy two account with two line and two number .
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Location: tunisia

trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 6:43 pm

and to explain good my probleme
i have two sip account and two computer with 2 x lite softphone .
i use manual dial and some times i use also auto dial.
so when i open two xlite from my two computer in same time .
for exemple agent "100" he use my carrier sip number one and he make dial progresse but my computer number two with agent "101" he still waiting when agent "100" finish his call with the carrier sip number one to make his dial progresse too.
and here i have two sip account in my server .
why agent "101" don't try to use my carrier sip number two when agent "100" have the carrier number one.
so here it is a solution for swithing sip carrier to the agent call.
this is my probleme i think i have explain good my probleme
thanks to all
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Location: tunisia

Postby Geil21 » Wed Jul 01, 2009 6:55 pm

The reason that carrier 2 can't be reached is because it does not actually exist in your dial plan. Your dialplan entry for carrier 1 is the same as your dialplan entry for carrier 2

because of this, asterisk only makes the first entry available to dial on

combine the entries as I described earlier


Go to vicidial carrier 1 and make the change then leave carrier 2 empty and see if that helps.

Of course this is proposed solution based on my interpretation of your post.

It's hard to understand, but I noticed the dialplan error right away! so if that doesn't work then we can go from there. I'll do my best for you because I know how hard it can be sometimes wth the language barrier.


MAKE SURE YOU ONLY REMOVE THE DIALPLAN SECTION FROM THE SECOND CARRIER AND NOT THE WHOLE THING
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trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 7:08 pm

ok sir i will try
and thanks
finaly i have found same one who understand my probleme
because my provider have say this is impossible with asterisk :D
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trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 7:25 pm

no same thing
; WARNING- THIS FILE IS AUTO-GENERATED BY VICIDIAL, ANY EDITS YOU MAKE WILL BE LOST
SIPtrunk2=SIP/trunk_2
SIPtrunk3=SIP/trunk_3

[vicidial-auto]
exten => h,1,DeadAGI(agi://127.0.0.1:4577/call_log--HVcauses ... ----------)

; Local Server: 192.168.1.10
exten => _192*168*001*010*.,1,Goto(default,${EXTEN:16},1)
; VICIDIAL Carrier: keyyo1 - keyyo
exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Dial(${SIPtrunk3}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,4,Hangup
; VICIDIAL Carrier: keyyo2 - keyyo

exten => 100,1,Dial(SIP/100)
exten => 100,2,Voicemail,u100
exten => 101,1,Dial(SIP/101)
exten => 101,2,Voicemail,u101

and same thing only trunk_2 work asterisk refuse the use of trunk_3 in same time for my agent "101" he still waiting the trunk_2 and when he will finish
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Postby Geil21 » Wed Jul 01, 2009 7:30 pm

Did you reload asterisk from the CLI to apply the new config?


From the CLI type "reload"
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Postby brett05 » Wed Jul 01, 2009 7:39 pm

Geil21 wrote:Did you reload asterisk from the CLI to apply the new config?


From the CLI type "reload"


ok i will do it now
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Postby Geil21 » Wed Jul 01, 2009 7:48 pm

Just a side note

Think of it like this

exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Dial(${SIPtrunk3}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,4,Hangup

What you are telling asterisk is the following...

When someone dials 9XXXXXXXXXX do the following

Step 1 Log the call
Step 2 Dial over the first carrier ( If it fails then go to Step 3 )
Step 3 Dial over the second carrier ( If it fails then go to Step 4 )
Step 4 Hang Up

That's the simplest explanation for what we are trying to accomplish
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trunk switch probleme

Postby brett05 » Wed Jul 01, 2009 8:09 pm

it is true to write as this line ?


exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}@{SIPtrunk3}/${EXTEN:1},55,o)
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Postby brett05 » Wed Jul 01, 2009 8:12 pm

Geil21 wrote:Just a side note

Think of it like this

exten => _9XXXXXXXXXX,1,AGI(agi://127.0.0.1:4577/call_log)
exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,3,Dial(${SIPtrunk3}/${EXTEN:1},55,o)
exten => _9XXXXXXXXXX,4,Hangup

What you are telling asterisk is the following...

When someone dials 9XXXXXXXXXX do the following

Step 1 Log the call
Step 2 Dial over the first carrier ( If it fails then go to Step 3 )
Step 3 Dial over the second carrier ( If it fails then go to Step 4 )
Step 4 Hang Up

That's the simplest explanation for what we are trying to accomplish


great explain thank teacher
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Location: tunisia

Re: trunk switch probleme

Postby Geil21 » Wed Jul 01, 2009 8:16 pm

brett05 wrote:it is true to write as this line ?


exten => _9XXXXXXXXXX,2,Dial(${SIPtrunk2}@{SIPtrunk3}/${EXTEN:1},55,o)


No, I don't believe that would work, never seen it done that way.

did the first solution not work after reloading cli?
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re

Postby brett05 » Wed Jul 01, 2009 11:01 pm

not work sir same tthe first agent "100" log in he use the trunk_2 and the agent "101" i can not see it dialing only if the agent "100" finish his dial??
this is strange from my server ?
i'am sure that it can from the dial plan but using of asterisk with different dial this is not easy
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re

Postby brett05 » Wed Jul 01, 2009 11:02 pm

if some one here have used a server with 2 agent log in and 2 provider please share this thing ?
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