Asterisk ring twice when using Dial Application

ThanhLoiTL · November 1, 2023, 11:59am

Here is the application I used in the code:

$this->client->exec('Dial', [__transfer_number, __timeout, G()]);

Here is a snapshot of my asterisk:

`
– AGI Script Executing Application: (Dial) Options: (SIP/trunk_test/7573342035,30,G(from-answer,1698835192,1))
== Using SIP RTP CoS mark 5

– Called SIP/trunk_test/7573342035

– SIP/trunk_test-000003ab is ringing

– Local/1698835192@outreach-00000396;1 is ringing

– SIP/trunk_test-000003ab is ringing

– SIP/trunk_test-000003ab redirecting info has changed, passing it to Local/1698835192@outreach-00000396;2

– SIP/trunk_test-000003ab is busy

== Everyone is busy/congested at this time (1:1/0/0)`

Summary:

First, the asterisk dial SIP/trunk_test. Then I hang up/ don’t pick up. But instead of nothing happened after that. there’s another call to my phone number so I have to hang up/ don’t pick up again.

Goal:

After the first hangup/don’t pick up there will be no more calls to my number.

jcolp · November 1, 2023, 12:48pm

You would need to provide the complete SIP trace from “sip set debug on” to show what is actually happening at a SIP level.

ThanhLoiTL · November 2, 2023, 3:04am

Sure, here is the completed sip trace:

Really destroying SIP dialog '441f699a0e8b1c340c20747d5559a7fe@127.0.1.1' Method: REGISTER
Really destroying SIP dialog '2957a17e04b938004af90b915a0b900a@127.0.1.1' Method: REGISTER
Really destroying SIP dialog '0ec4a51b3d74cd245ad94fc91bfbe6e6@127.0.1.1' Method: REGISTER
[Nov  2 09:26:21] WARNING[951]: pbx_spool.c:253 parse_line: Invalid retrytime at line 4 of /var/spool/asterisk/outgoing/1698891981.call
    -- Attempting call on Local/1698891981@outreach for application Wait(3600) (Retry 1)
    -- Called 1698891981@outreach
    -- Executing [1698891981@outreach:1] AGI("Local/1698891981@outreach-000003b4;2", "callflow/outreach.php,1698891981")
    -- Launched AGI Script /var/lib/asterisk/agi-bin/callflow/outreach.php
 callflow/outreach.php,1698891981: Incoming callflow phone call.
 callflow/outreach.php,1698891981: Find phone numbers for outreach.
 callflow/outreach.php,1698891981: Finded call flow design.
[Nov  2 09:26:22] WARNING[78090][C-000003e5]: chan_sip.c:23340 func_headers_read2: This function can only be used on SIP channels.
[Nov  2 09:26:22] WARNING[78090][C-000003e5]: chan_sip.c:23275 func_header_read: This function requires a header name.
 callflow/outreach.php,1698891981: Start call flow design.
 callflow/outreach.php,1698891981: Call setting init to get and set asterisk and callflow varriable
 callflow/outreach.php,1698891981: Incomming setting callflow
 callflow/outreach.php,1698891981: Go to action: Check Outreach By
 callflow/outreach.php,1698891981: Go to action: Dial


  AGI Script Executing Application: (Dial) Options: (SIP/7573342035@trunk_test,30,G(from-answer,1698891981,1))
  == Using SIP RTP CoS mark 5
Audio is at 17924
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 64.2.142.26:5060:
INVITE sip:7573342035@xxxxx.xxxx:5060 SIP/2.0
Via: SIP/2.0/TCP 192.168.1.121:5060;branch=z9hG4bK0cc34020;rport
Max-Forwards: 70
From: <sip:2065621001@192.168.1.121>;tag=as2db0452d
To: <sip:7573342035@xxxxx.xxxx:5060>
Contact: <sip:2065621001@192.168.1.121:5060;transport=tcp>
Call-ID: 2d7086e628392405358d2c2b1422e84c@192.168.1.121:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.14.0
Date: Thu, 02 Nov 2023 02:26:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 240

v=0
o=root 569534293 569534293 IN IP4 192.168.1.121
s=Asterisk PBX 18.14.0
c=IN IP4 192.168.1.121
t=0 0
m=audio 17924 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

--- Called SIP/7573342035@trunk_test

<--- SIP read from TCP:64.2.142.26:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 192.168.1.121:5060;received=192.168.1.121;branch=z9hG4bK0cc34020;rport=63268
From: <sip:2065621001@192.168.1.121>;tag=as2db0452d
To: <sip:7573342035@xxxxx.xxxx:5060>
Call-ID: 2d7086e628392405358d2c2b1422e84c@192.168.1.121:5060
CSeq: 102 INVITE
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog '4f4b42ba2bc86c22-12046@127.0.0.1' in 32000 ms (Method: OPTIONS)

<--- SIP read from TCP:64.2.142.26:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.1.121:5060;received=192.168.1.121;branch=z9hG4bK0cc34020;rport=63268
From: <sip:2065621001@192.168.1.121>;tag=as2db0452d
To: <sip:7573342035@xxxxx.xxxx:5060>;tag=as52fc6a00
Call-ID: 2d7086e628392405358d2c2b1422e84c@192.168.1.121:5060
CSeq: 102 INVITE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:7573342035@64.2.142.26:5060;transport=tcp>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:7573342035@64.2.142.26:5060;transport=tcp>
    -- SIP/trunk_test-000003c6 is ringing
    -- Local/1698891981@outreach-000003b4;1 is ringing
Really destroying SIP dialog '48a0e46d71b5278d34b900da01b893bc@192.168.1.121:5060' Method: OPTIONS
Retransmitting #3 (NAT) to 64.190.63.111:5060:
REGISTER sip:yourdomain.com SIP/2.0
Via: SIP/2.0/UDP 192.168.1.121:5060;branch=z9hG4bK5946980a;rport
Max-Forwards: 70
From: <sip:0123456789@yourdomain.com>;tag=as4a792ec6
To: <sip:0123456789@yourdomain.com>
Call-ID: 2957a17e04b938004af90b915a0b900a@127.0.1.1
CSeq: 2771 REGISTER
Supported: replaces, timer
User-Agent: Asterisk PBX 18.14.0
Expires: 120
Contact: <sip:trunk%2001@192.168.1.121:5060>
Content-Length: 0

---
Really destroying SIP dialog '4f4b42ba2bc86c1c-12046@127.0.0.1' Method: OPTIONS

<--- SIP read from TCP:64.2.142.26:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TCP 192.168.1.121:5060;received=192.168.1.121;branch=z9hG4bK0cc34020;rport=63268
From: <sip:2065621001@192.168.1.121>;tag=as2db0452d
To: <sip:7573342035@xxxxx.xxxx:5060>;tag=as52fc6a00
Call-ID: 2d7086e628392405358d2c2b1422e84c@192.168.1.121:5060
CSeq: 102 INVITE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Contact: <sip:7573342035@64.2.142.26:5060;transport=tcp>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:7573342035@64.2.142.26:5060;transport=tcp>
    -- SIP/trunk_test-000003c6 is ringing

<--- SIP read from TCP:64.2.142.26:5060 --->
SIP/2.0 480 Temporarily unavailable
Via: SIP/2.0/TCP 192.168.1.121:5060;received=192.168.1.121;branch=z9hG4bK0cc34020;rport=63268
From: <sip:2065621001@192.168.1.121>;tag=as2db0452d
To: <sip:7573342035@xxxxx.xxxx:5060>;tag=as52fc6a00
Call-ID: 2d7086e628392405358d2c2b1422e84c@192.168.1.121:5060
CSeq: 102 INVITE
Server: packetrino
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 64.2.142.26:5060:
ACK sip:7573342035@64.2.142.26:5060;transport=tcp SIP/2.0
Via: SIP/2.0/TCP 192.168.1.121:5060;branch=z9hG4bK0cc34020;rport
Max-Forwards: 70
From: <sip:2065621001@192.168.1.121>;tag=as2db0452d
To: <sip:7573342035@xxxxx.xxxx:5060>;tag=as52fc6a00
Contact: <sip:2065621001@192.168.1.121:5060;transport=tcp>
Call-ID: 2d7086e628392405358d2c2b1422e84c@192.168.1.121:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.14.0
Content-Length: 0


---SIP/trunk_test-000003c6 redirecting info has changed, passing it to Local/1698891981@outreach-000003b4;2
--- SIP/trunk_test-000003c6 is busy
Scheduling destruction of SIP dialog '2d7086e628392405358d2c2b1422e84c@192.168.1.121:5060' in 14848 ms (Method: INVITE)
  == Everyone is busy/congested at this time (1:1/0/0)
 callflow/outreach.php,1698891981: DIALSTATUS: BUSY
 callflow/outreach.php,1698891981: Go to action: Update  - Busy

In the above sip trace, I have censored the sip trunk address with xxxxx and changed the real phone number. Please let me know if you ever need more information. Thanks

jcolp · November 2, 2023, 8:30am

If the given SIP trace and log is all there is, the problem is not in Asterisk and doesn’t appear as though it can be resolved there. It sends a call out, gets a ringing indication, then ultimately gets a temporarily unavailable. It’s not the one behaving as you describe.

system · December 2, 2023, 8:30am

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Asterisk ring twice when using Dial Application

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