Remote host can't match request CANCEL to call

jeet · December 16, 2014, 12:35pm

Hi,

After updating our machine to Asterisk 11.14.1 we are a small problem. Our connected CISCO 7960 seems to be working fine, untill we call that CISCO extension from within itself, i.e call the same extension on the 7960 from that hard phone itself. Phone rings and as we answer the call gets dropped and following is displayed in CLI.

We didnt have an issue before this on Asterisk 11.12

Any help would be appreciated.

Thanks.

david55 · December 16, 2014, 2:27pm

That’s not the cause of the call dropping. It is the peer refusing to terminate the unanswered call at Asterisk’s request. Asterisk has already decided to drop the call.

You will need to provide the complete INVITE transaction for us to be able to work out what is really happening (sip set debug on).

jeet · December 17, 2014, 8:51am

Hi David,

Thanks for your help and time. Much appreciated!

Following is the complete debug log.

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK02111e41;rport
From: "Unknown" <sip:Unknown@192.168.1.150:5080>;tag=as06e7e920
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b00084276cc3a-0435affd
Call-ID: 4e3c9f8341aa000a416814405d95e3c7@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:44 GMT
CSeq: 102 OPTIONS
Server: Cisco-CP7960G/7.5
Content-Length: 0
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Allow-Events: kpml,dialog,refer
Accept: application/sdp,multipart/mixed,multipart/alternative
Accept-Encoding: identity
Accept-Language: en
Supported: replaces
Content-Length: 217
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27229 0 IN IP4 192.168.3.100
s=SIP Call
t=0 0
m=audio 0 RTP/AVP 0 8 18 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
--- (18 headers 10 lines) ---
Really destroying SIP dialog '4e3c9f8341aa000a416814405d95e3c7@192.168.1.150:5080' Method: OPTIONS

<--- SIP read from UDP:192.168.1.200:5080 --->
INVITE sip:5001@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK0855e56a
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
Max-Forwards: 70
Date: Wed, 17 Dec 2014 08:35:45 GMT
CSeq: 101 INVITE
User-Agent: Cisco-CP7960G/7.5
Contact: <sip:5001@192.168.3.100:5080>
Expires: 180
Accept: application/sdp
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces
Content-Length: 257
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27029 0 IN IP4 192.168.3.100
s=SIP Call
t=0 0
m=audio 18392 RTP/AVP 0 8 18 101
c=IN IP4 192.168.3.100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (17 headers 12 lines) ---
Sending to 192.168.1.200:5080 (NAT)
Sending to 192.168.1.200:5080 (NAT)
Using INVITE request as basis request - 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
Found peer '5001' for '5001' from 192.168.1.200:5080

<--- Reliably Transmitting (NAT) to 192.168.1.200:5080 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK0855e56a;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as1303db3d
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 101 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="283b6abe"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100' in 25408 ms (Method: INVITE)
Retransmitting #10 (NAT) to 192.168.1.200:5080:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK2911fa23;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b000641a2e1c7-0d58ec6c
To: <sip:5001@192.168.1.150>;tag=as12235e41
Call-ID: 000e833c-ad4b0004-3989d8ce-752e8b98@192.168.3.100
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.200:5080 --->
ACK sip:5001@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.100:5060;branch=z9hG4bK0855e56a
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as1303db3d
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
Date: Wed, 17 Dec 2014 08:35:45 GMT
CSeq: 101 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.200:5080 --->
INVITE sip:5001@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
Max-Forwards: 70
Date: Wed, 17 Dec 2014 08:35:45 GMT
CSeq: 102 INVITE
User-Agent: Cisco-CP7960G/7.5
Contact: <sip:5001@192.168.3.100:5080>
Authorization: Digest username="5001",realm="asterisk",uri="sip:5001@192.168.1.150",response="4c392454488ee0121b1c3fc8cbce8f33",nonce="283b6abe",algorithm=MD5
Expires: 180
Content-Length: 257
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 27029 0 IN IP4 192.168.3.100
s=SIP Call
t=0 0
m=audio 18392 RTP/AVP 0 8 18 101
c=IN IP4 192.168.3.100
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (15 headers 12 lines) ---
Sending to 192.168.1.200:5080 (NAT)
Using INVITE request as basis request - 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
Found peer '5001' for '5001' from 192.168.1.200:5080
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 18
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format G729 for ID 18
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw|alaw|g729)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.3.100:18392
Looking for 5001 in from-internal (domain 192.168.1.150)
list_route: hop: <sip:5001@192.168.3.100:5080>

<--- Transmitting (NAT) to 192.168.1.200:5080 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@192.168.1.150:5080>
Content-Length: 0


<------------>
Audio is at 10392
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding codec 100003 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 192.168.1.200:5080:
INVITE sip:5001@192.168.3.100:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
Max-Forwards: 70
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>
Contact: <sip:5001@192.168.1.150:5080>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.14.1)
Date: Wed, 17 Dec 2014 08:35:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1389212455 1389212455 IN IP4 192.168.1.150
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.150
t=0 0
m=audio 10392 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- Transmitting (NAT) to 192.168.1.200:5080 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as5e4199ae
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@192.168.1.150:5080>
Content-Length: 0


<------------>
Retransmitting #1 (NAT) to 192.168.1.200:5080:
INVITE sip:5001@192.168.3.100:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
Max-Forwards: 70
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>
Contact: <sip:5001@192.168.1.150:5080>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
CSeq: 102 INVITE
User-Agent: FPBX-2.11.0(11.14.1)
Date: Wed, 17 Dec 2014 08:35:46 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 1389212455 1389212455 IN IP4 192.168.1.150
s=Asterisk PBX 11.14.1
c=IN IP4 192.168.1.150
t=0 0
m=audio 10392 RTP/AVP 8 3 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:46 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/7.5
Contact: <sip:5001@192.168.3.100:5080>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:46 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/7.5
Contact: <sip:5001@192.168.3.100:5080>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:5001@192.168.3.100:5080>

<--- Transmitting (NAT) to 192.168.1.200:5080 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as5e4199ae
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:5001@192.168.1.150:5080>
Content-Length: 0


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

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:46 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/7.5
Contact: <sip:5001@192.168.3.100:5080>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
list_route: hop: <sip:5001@192.168.3.100:5080>

<--- SIP read from UDP:192.168.1.200:5080 --->
CANCEL sip:5001@192.168.1.150 SIP/2.0
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
Max-Forwards: 70
Date: Wed, 17 Dec 2014 08:35:46 GMT
CSeq: 102 CANCEL
User-Agent: Cisco-CP7960G/7.5
Content-Length: 0
Authorization: Digest username="5001",realm="asterisk",uri="sip:5001@192.168.1.150",response="1fde85ae32f86773236ebc8daac73ce7",nonce="283b6abe",algorithm=MD5

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.1.200:5080 (NAT)

<--- Reliably Transmitting (NAT) to 192.168.1.200:5080 --->
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as5e4199ae
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


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

<--- Transmitting (NAT) to 192.168.1.200:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as5e4199ae
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 102 CANCEL
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080' in 25408 ms (Method: INVITE)
Reliably Transmitting (NAT) to 192.168.1.200:5080:
CANCEL sip:5001@192.168.3.100:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
Max-Forwards: 70
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
CSeq: 102 CANCEL
User-Agent: FPBX-2.11.0(11.14.1)
Content-Length: 0


---
Scheduling destruction of SIP dialog '5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080' in 25408 ms (Method: INVITE)

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:47 GMT
CSeq: 102 INVITE
Server: Cisco-CP7960G/7.5
Contact: <sip:5001@192.168.3.100:5080>
Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE
Supported: replaces
Content-Length: 207
Content-Type: application/sdp
Content-Disposition: session;handling=optional

v=0
o=Cisco-SIPUA 20437 0 IN IP4 192.168.3.100
s=SIP Call
t=0 0
m=audio 18396 RTP/AVP 0 101
c=IN IP4 192.168.3.100
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
<------------->
--- (14 headers 10 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (gsm|ulaw|alaw), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.3.100:18396
list_route: hop: <sip:5001@192.168.3.100:5080>
set_destination: Parsing <sip:5001@192.168.3.100:5080> for address/port to send to
set_destination: set destination to 192.168.3.100:5080
Transmitting (NAT) to 192.168.1.200:5080:
ACK sip:5001@192.168.3.100:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK2e923fc1;rport
Max-Forwards: 70
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Contact: <sip:5001@192.168.1.150:5080>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
CSeq: 102 ACK
User-Agent: FPBX-2.11.0(11.14.1)
Content-Length: 0


---
set_destination: Parsing <sip:5001@192.168.3.100:5080> for address/port to send to
set_destination: set destination to 192.168.3.100:5080
Reliably Transmitting (NAT) to 192.168.1.200:5080:
BYE sip:5001@192.168.3.100:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK34d9b002;rport
Max-Forwards: 70
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.14.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
Scheduling destruction of SIP dialog '5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080' in 25408 ms (Method: INVITE)
Retransmitting #1 (NAT) to 192.168.1.200:5080:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as5e4199ae
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


---
Retransmitting #1 (NAT) to 192.168.1.200:5080:
CANCEL sip:5001@192.168.3.100:5080;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
Max-Forwards: 70
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
CSeq: 102 CANCEL
User-Agent: FPBX-2.11.0(11.14.1)
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:47 GMT
CSeq: 102 CANCEL
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Retransmitting #1 (NAT) to 192.168.1.200:5080:
BYE sip:5001@192.168.3.100:5080 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK34d9b002;rport
Max-Forwards: 70
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
CSeq: 103 BYE
User-Agent: FPBX-2.11.0(11.14.1)
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK34d9b002;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:47 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/7.5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 481 Call Leg/Transaction Does Not Exist
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK25d8f02d;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:47 GMT
CSeq: 102 CANCEL
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
[2014-12-17 08:35:48] WARNING[3284][C-00000041]: chan_sip.c:23972 handle_response: Remote host can't match request CANCEL to call '5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080'. Giving up.
Really destroying SIP dialog '5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080' Method: INVITE

<--- SIP read from UDP:192.168.1.200:5080 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.150:5080;branch=z9hG4bK34d9b002;rport
From: "User Name" <sip:5001@192.168.1.150:5080>;tag=as3d91ec44
To: <sip:5001@192.168.3.100:5080;user=phone>;tag=000e833cad4b000a73d0d46d-5be09cab
Call-ID: 5c415997226db5dd3f7173cf19eee7a9@192.168.1.150:5080
Date: Wed, 17 Dec 2014 08:35:47 GMT
CSeq: 103 BYE
Server: Cisco-CP7960G/7.5
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Retransmitting #2 (NAT) to 192.168.1.200:5080:
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP 192.168.3.100:5080;branch=z9hG4bK745d7cfe;received=192.168.1.200;rport=5080
From: "5001" <sip:5001@192.168.1.150>;tag=000e833cad4b00090a975a07-2466c342
To: <sip:5001@192.168.1.150>;tag=as5e4199ae
Call-ID: 000e833c-ad4b0005-37a901b3-13d0f0e6@192.168.3.100
CSeq: 102 INVITE
Server: FPBX-2.11.0(11.14.1)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

david55 · December 17, 2014, 10:23am

The caller abandoned the call because it was taking too long to answer. However, just as Asterisk was about to cancel the call, it was finally answered. There seems to have been a collision between the cancel and the OK.

Is it possible that only one call could be supported at a time, so the peer didn’t answer until the outgoing call cleared?

jeet · December 18, 2014, 10:10am

The caller abandoned the call because it was taking too long to answer. However, just as Asterisk was about to cancel the call, it was finally answered. There seems to have been a collision between the cancel and the OK.

[quote=“david55”]
Is it possible that only one call could be supported at a time, so the peer didn’t answer until the outgoing call cleared?[/quote]

Hi David55,

Not sure, but I think no. How may I check? Can you suggest something that comes to mind Captain.

Thanks.

david55 · December 18, 2014, 10:25am

Watch the SIP trace carefully as you try and abandon the call at different times. If the OK always happens when you abandon the call, it will tend to confirm the theory.