---

Reliably Transmitting (no NAT) to 192.168.0.5:5060:

INVITE sip:1002@192.168.0.5;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK1def34d0

Max-Forwards: 70

From: "test" <sip:1004@192.168.0.76>;tag=as4ef56956

To: <sip:1002@192.168.0.5;transport=udp>

Contact: <sip:1004@192.168.0.76:5060>

Call-ID: 44835de82b8da0f208e4eade7f9cf6a5@192.168.0.76:5060

CSeq: 102 INVITE

User-Agent: Asterisk PBX 11.25.3

Date: Fri, 25 Sep 2026 12:22:52 GMT

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Content-Type: application/sdp

Content-Length: 508

 

v=0

o=root 1471945648 1471945648 IN IP4 192.168.0.76

s=Asterisk PBX 11.25.3

c=IN IP4 192.168.0.76

t=0 0

m=audio 15920 RTP/AVP 8 0 3 97 18 9 4 110 96 119 97

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:3 GSM/8000

a=rtpmap:97 iLBC/8000

a=rtpmap:18 G729/8000

a=fmtp:18 annexb=no

a=rtpmap:9 G722/8000

a=rtpmap:4 G723/8000

a=fmtp:4 annexa=no

a=rtpmap:110 speex/8000

a=rtpmap:96 speex/16000

a=rtpmap:119 speex/32000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-16

a=ptime:20

a=sendrecv

 

---

    -- Called SIP/1002

    -- Music class default requested but no musiconhold loaded.

Audio is at 14502

Adding codec 100004 (alaw) to SDP

Adding codec 100003 (ulaw) to SDP

Adding codec 100016 (speex16) to SDP

Adding non-codec 0x1 (telephone-event) to SDP

 

<--- Transmitting (no NAT) to 192.168.0.60:41790 --->

SIP/2.0 183 Session Progress

Via: SIP/2.0/UDP 192.168.0.60:41790;branch=z9hG4bK.NzNt4SWzk;received=192.168.0.60;rport=41790

From: "test" <sip:1004@192.168.0.76>;tag=wfJuoPfzn

To: sip:1002@192.168.0.76;tag=as78a05240

Call-ID: PxKWwjI46x

CSeq: 20 INVITE

Server: Asterisk PBX 11.25.3

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

Contact: <sip:1002@192.168.0.76:5060>

Content-Type: application/sdp

Content-Length: 283

 

v=0

o=root 935761684 935761684 IN IP4 192.168.0.76

s=Asterisk PBX 11.25.3

c=IN IP4 192.168.0.76

t=0 0

m=audio 14502 RTP/AVP 8 0 96 97

a=rtpmap:8 PCMA/8000

a=rtpmap:0 PCMU/8000

a=rtpmap:96 speex/16000

a=rtpmap:97 telephone-event/8000

a=fmtp:97 0-16

a=ptime:20

a=sendrecv

 

<------------>

 

<--- SIP read from UDP:192.168.0.5:5060 --->

SIP/2.0 100 Trying

Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK1def34d0

From: "test" <sip:1004@192.168.0.76>;tag=as4ef56956

To: <sip:1002@192.168.0.5;transport=udp>

Call-ID: 44835de82b8da0f208e4eade7f9cf6a5@192.168.0.76:5060

CSeq: 102 INVITE

 

<------------->

--- (6 headers 0 lines) ---

 

<--- SIP read from UDP:192.168.0.5:5060 --->

SIP/2.0 180 Ringing

Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK1def34d0

From: "test" <sip:1004@192.168.0.76>;tag=as4ef56956

To: <sip:1002@192.168.0.5;transport=udp>;tag=u6v6bf8

Call-ID: 44835de82b8da0f208e4eade7f9cf6a5@192.168.0.76:5060

CSeq: 102 INVITE

User-Agent: Linphone-Desktop/6.1.1 (DESKTOP-SYA2882) windows/10 Qt/6.9.1 LinphoneSDK/5.4.88

Supported: replaces, outbound, gruu, path, record-aware

 

<------------->

--- (8 headers 0 lines) ---

list_route: no route

    -- SIP/1002-00000005 is ringing

 

<--- SIP read from UDP:192.168.0.5:5060 --->

SIP/2.0 603 Decline

Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK1def34d0

From: "test" <sip:1004@192.168.0.76>;tag=as4ef56956

To: <sip:1002@192.168.0.5;transport=udp>;tag=u6v6bf8

Call-ID: 44835de82b8da0f208e4eade7f9cf6a5@192.168.0.76:5060

CSeq: 102 INVITE

User-Agent: Linphone-Desktop/6.1.1 (DESKTOP-SYA2882) windows/10 Qt/6.9.1 LinphoneSDK/5.4.88

Supported: replaces, outbound, gruu, path, record-aware

 

<------------->

--- (8 headers 0 lines) ---

    -- Got SIP response 603 "Decline" back from 192.168.0.5:5060

Transmitting (no NAT) to 192.168.0.5:5060:

ACK sip:1002@192.168.0.5;transport=udp SIP/2.0

Via: SIP/2.0/UDP 192.168.0.76:5060;branch=z9hG4bK1def34d0

Max-Forwards: 70

From: "test" <sip:1004@192.168.0.76>;tag=as4ef56956

To: <sip:1002@192.168.0.5;transport=udp>;tag=u6v6bf8

Contact: <sip:1004@192.168.0.76:5060>

Call-ID: 44835de82b8da0f208e4eade7f9cf6a5@192.168.0.76:5060

CSeq: 102 ACK

User-Agent: Asterisk PBX 11.25.3

Content-Length: 0

 

 

---

    -- SIP/1002-00000005 is busy

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

    -- Auto fallthrough, channel 'SIP/1004-00000004' status is 'BUSY'

 

<--- Reliably Transmitting (no NAT) to 192.168.0.60:41790 --->

SIP/2.0 486 Busy Here

Via: SIP/2.0/UDP 192.168.0.60:41790;branch=z9hG4bK.NzNt4SWzk;received=192.168.0.60;rport=41790

From: "test" <sip:1004@192.168.0.76>;tag=wfJuoPfzn

To: sip:1002@192.168.0.76;tag=as78a05240

Call-ID: PxKWwjI46x

CSeq: 20 INVITE

Server: Asterisk PBX 11.25.3

Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE

Supported: replaces, timer

X-Asterisk-HangupCause: Call Rejected

X-Asterisk-HangupCauseCode: 21

Content-Length: 0

 

 

<------------>

Really destroying SIP dialog '44835de82b8da0f208e4eade7f9cf6a5@192.168.0.76:5060' Method: INVITE

 

<--- SIP read from UDP:192.168.0.60:41790 --->

ACK sip:1002@192.168.0.76 SIP/2.0

Via: SIP/2.0/UDP 192.168.0.60:41790;branch=z9hG4bK.NzNt4SWzk;rport

Call-ID: PxKWwjI46x

From: "test" <sip:1004@192.168.0.76>;tag=wfJuoPfzn

To: <sip:1002@192.168.0.76>;tag=as78a05240

Contact: <sip:1004@192.168.0.60:41790;transport=udp>;expires=3600;+org.linphone.specs="lime"

Max-Forwards: 70

CSeq: 20 ACK