Hi all
Here is what I am doing.
My call file calls the Local Channel which in turn calls an agent and plays simple IVR press 1 to dial out and 2 for hangup.
But when agent presses 1 to call out its not making call .
Q.Is this possible to connect to local channel which calls a SIP channel (agent ) then conneting that SIP channel(agent who presses one to call out) to an outside number…?
Here is my dial plan
[call_out]
exten => 11,1,verbose(CALLID :${call_id} ----${agent})
exten => 11,n,verbose(-------callerid is : ${CALLERID(num)})
exten => 11,n,Dial(SIP/${agent}) ;dialling the agent
exten => 9090,1,Background(/opt/webcall)
exten => 9090,n,WaitExten(5)
exten => 2,1,verbose(–agent denied call–)
exten => 2,n,hangup()
exten => 1,1,verbose(–agent selected to call customer—${call_id})
exten => 1,n,Set(CALLERID(all)=${call_id})
exten => 1,n,Dial(SIP/flowroute/${phone_to_call})
My call file format :
Channel:Local/11@call_out
Context:call_out
Extension:9090
Priority:1
SetVar:call_id=< call id to ser>
SetVar:agent=8888
SetVar:phone_to_call=
This is my SIP debug output :
– SIP/8888-000000fc answered Local/11@call_out-a9ba;2
> Channel Local/11@call_out-a9ba;1 was answered.
– Executing [9090@call_out:1] Verbose(“Local/11@call_out-a9ba;1”, “------The number to call is: 12345678-----agent number selected is :8888”) in new stack
------The number to call is: 12345678-----agent number selected is :8888
– Executing [9090@call_out:2] BackGround(“Local/11@call_out-a9ba;1”, “/opt/webcall”) in new stack
– <Local/11@call_out-a9ba;1> Playing ‘/opt/webcall.gsm’ (language ‘en’)
== Spawn extension (call_out, 11, 3) exited non-zero on ‘Local/11@call_out-a9ba;2’
<— SIP read from UDP:SERVER-IP:17575 —>
SUBSCRIBE sip:504@192.168.0.107 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:9112;branch=z9hG4bK-d87543-34203777a0037b36-1–d87543-;rport
Max-Forwards: 70
Contact: sip:8888@SERIP:17575
To: sip:504@192.168.0.107
From: "nitesh"sip:8888@SERVER-IP;tag=61749242
Call-ID: 9012200fe132a720ODhkYTliYWEyMTQxYzg0NmMwZmQxMTk5MDNmZWZlMDk.
CSeq: 1 SUBSCRIBE
Expires: 3600
Accept: multipart/related, application/rlmi+xml, application/pidf+xml
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1002tx stamp 29712
Event: presence
Content-Length: 0
<------------->
— (14 headers 0 lines) —
Creating new subscription
Sending to SERIP:17575 (NAT)
list_route: hop: sip:8888@SERVER-IP:17575
Found peer ‘8888’ for ‘8888’ from SERVER-IP:17575
<— Transmitting (NAT) to SERVER-IP:17575 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.0.103:9112;branch=z9hG4bK-d87543-34203777a0037b36-1–d87543-;received=SERVER-IP;rport=17575
From: "nitesh"sip:8888@domain-name;tag=61749242
To: sip:504@192.168.0.107;tag=as2d8d13ac
Call-ID: 9012200fe132a720ODhkYTliYWEyMTQxYzg0NmMwZmQxMTk5MDNmZWZlMDk.
CSeq: 1 SUBSCRIBE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="4d149f43"
Content-Length: 0
<------------>
Scheduling destruction of SIP dialog ‘9012200fe132a720ODhkYTliYWEyMTQxYzg0NmMwZmQxMTk5MDNmZWZlMDk.’ in 27200 ms (Method: SUBSCRIBE)
<— SIP read from UDP:SERVER-IP:17575 —>
SUBSCRIBE sip:504@192.168.0.107 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.103:9112;branch=z9hG4bK-d87543-281924023362020b-1–d87543-;rport
Max-Forwards: 70
Contact: sip:8888@SERVER-IP:17575
To: sip:504@192.168.0.107
From: “nitesh"sip:8888@domain-name;tag=61749242
Call-ID: 9012200fe132a720ODhkYTliYWEyMTQxYzg0NmMwZmQxMTk5MDNmZWZlMDk.
CSeq: 2 SUBSCRIBE
Expires: 3600
Accept: multipart/related, application/rlmi+xml, application/pidf+xml
llow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO
User-Agent: X-Lite release 1002tx stamp 29712
Authorization: Digest username=“8888”,realm=“asterisk”,nonce=“4d149f43”,uri="sip:504@192.168.0.107”,response=“5b9b82692ef90e2dc4c8872cfdf44e8c”,algorithm=MD5
Event: presence
Content-Length: 0
<------------->
— (15 headers 0 lines) —
Creating new subscription
Sending to SERVER-IP:17575 (NAT)
Found peer ‘8888’ for ‘8888’ from SERVER-IP:17575
Looking for 504 in from-internal (domain 192.168.0.107)
<— Transmitting (NAT) to SERVER-IP:17575 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.0.103:9112;branch=z9hG4bK-d87543-281924023362020b-1–d87543-;received=SERVER-IP;rport=17575
From: "nitesh"sip:8888@domain-name;tag=61749242
To: sip:504@192.168.0.107;tag=as2d8d13ac
Call-ID: 9012200fe132a720ODhkYTliYWEyMTQxYzg0NmMwZmQxMTk5MDNmZWZlMDk.
CSeq: 2 SUBSCRIBE
Server: FPBX-2.8.1(1.8.11.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
Really destroying SIP dialog ‘9012200fe132a720ODhkYTliYWEyMTQxYzg0NmMwZmQxMTk5MDNmZWZlMDk.’ Method: SUBSCRIBE
– Executing [9090@call_out:3] WaitExten(“SIP/8888-000000fc”, “5”) in new stack
== CDR updated on SIP/8888-000000fc
– Executing [1@call_out:1] Verbose(“SIP/8888-000000fc”, “–agent selected to call customer—1234”) in new stack
–agent selected to call customer—1234
– Executing [1@call_out:2] Set(“SIP/8888-000000fc”, “CALLERID(all)=1234”) in new stack
– Executing [1@call_out:3] Dial(“SIP/8888-000000fc”, “SIP/flowroute/12345678”) in new stack
== Using SIP RTP TOS bits 184
== Using SIP RTP CoS mark 5
Really destroying SIP dialog ‘38ff09731846abc54b6160c14006d62d@provider:5060’ Method: INVITE
== Everyone is busy/congested at this time (1:0/0/1)
– Auto fallthrough, channel ‘SIP/8888-000000fc’ status is ‘CHANUNAVAIL’