SIP Transfer (REFER)

Asterisk Asterisk Support
1

Hello,

I have some issues on attented SIP Transfers with Asterisk 11 (I also tried 1.4, 1.6 and 1.8) = >

Scenario :

A calls B
B Transfer A to C (B tried to transfer directly, not via asterisk)

sip.conf

[code][general]
defaultexpirey=1800
dtmfmode=auto
language=fr
disallow=all
allow=ulaw
nat=no
canreinvite=yes
directmedia=yes
allowguest=yes
directrtpsetup=yes
pedantic=no
autocreatepeer=yes
allowexternaldomains=yes
trustrpid=yes

[median]
type=peer
defaultuser=median
host=192.168.1.67
context=median
qualify=yes

[median2]
type=peer
defaultuser=median2
host=192.168.1.69
context=median
qualify=yes

[argos]
type=peer
defaultuser=argos
host=192.168.1.68
context=ireflet
qualify=yes

[192.168.20.11]
type=friend
call-limit=4
defaultuser=399
callerid=“399” <399>
insecure=port,invite
host=192.168.20.11
context=prod
qualify=yes

[399]
type=friend
call-limit=4
defaultuser=399
callerid=“399” <399>
insecure=port,invite
host=192.168.20.11
context=ireflet
qualify=yes[/code]

extensions.conf :

[code]; Globals settings
[general]
static=yes
writeprotect=no
autofallthrough=yes
clearglobalvars=no
priorityjumping=no

[median]
exten => _X.,1,Verbose(Appel Median 2000 - de ${CALLERID(number)} pour ${EXTEN})
same => n,Dial(SIP/argos/9813)
same => n,Hangup()[/code]

Traces :

[code][Jan 3 14:27:10] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.69:5060 —>
INVITE sip:9813@192.168.1.224;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKac162135657
Max-Forwards: 70
From: sip:170956464@192.168.1.69;tag=1c162127078
To: sip:9813@192.168.1.224;user=phone
Call-ID: 162125846312013142720@192.168.1.69
CSeq: 1 INVITE
Contact: sip:170956464@192.168.1.69:5060
Supported: em,100rel,timer,replaces,path,resource-priority,sdp-anat
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 2000/v.6.40A.019.008
Content-Type: application/sdp
Content-Length: 243

v=0
o=AudiocodesGW 162117073 162116726 IN IP4 192.168.1.69
s=Phone-Call
c=IN IP4 192.168.1.69
t=0 0
m=audio 7410 RTP/AVP 8 0 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:30
a=sendrecv

<------------->
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c: — (13 headers 12 lines) —
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Sending to 192.168.1.69:5060 (no NAT)
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Using INVITE request as basis request - 162125846312013142720@192.168.1.69
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found peer ‘median2’ for ‘170956464’ from 192.168.1.69:5060
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found RTP audio format 8
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found RTP audio format 0
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found RTP audio format 4
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found audio description format PCMA for ID 8
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found audio description format PCMU for ID 0
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found audio description format G723 for ID 4
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(g723|ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Peer audio RTP is at port 192.168.1.69:7410
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Looking for 9813 in median (domain 192.168.1.224)
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: list_route: hop: sip:170956464@192.168.1.69:5060
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.69:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKac162135657;received=192.168.1.69
From: sip:170956464@192.168.1.69;tag=1c162127078
To: sip:9813@192.168.1.224;user=phone
Call-ID: 162125846312013142720@192.168.1.69
CSeq: 1 INVITE
Server: Asterisk PBX 11.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:9813@192.168.1.224:5060
Content-Length: 0

<------------>
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] pbx.c: – Executing [9813@median:1] Verbose(“SIP/median2-00000004”, “Appel Median 2000 - de 170956464 pour 9813”) in new stack
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] app_verbose.c: Appel Median 2000 - de 170956464 pour 9813
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] pbx.c: – Executing [9813@median:2] Dial(“SIP/median2-00000004”, “SIP/argos/9813”) in new stack
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] netsock2.c: == Using SIP RTP CoS mark 5
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c: Audio is at 53554
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c: Adding non-codec 0x1 (telephone-event) to SDP
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.68:5060:
INVITE sip:9813@192.168.1.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK6e678d9a
Max-Forwards: 70
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68
Contact: sip:170956464@192.168.1.224:5060
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.2.0-rc1
Date: Thu, 03 Jan 2013 13:27:10 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1130325334 1130325334 IN IP4 192.168.1.69
s=Asterisk PBX 11.2.0-rc1
c=IN IP4 192.168.1.69
t=0 0
m=audio 7410 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

[Jan 3 14:27:10] VERBOSE[31677][C-00000002] app_dial.c: – Called SIP/argos/9813
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK6e678d9a
From: sip:170956464@192.168.1.224;tag=as270090b3
To: “C3T SIP” sip:9813@192.168.1.68;tag=15550
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 102 INVITE
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 0

<------------->
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c: — (9 headers 0 lines) —
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK6e678d9a
From: sip:170956464@192.168.1.224;tag=as270090b3
To: “C3T SIP” sip:9813@192.168.1.68;tag=15550
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 102 INVITE
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 0

<------------->
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c: — (9 headers 0 lines) —
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: list_route: no route
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] app_dial.c: – SIP/argos-00000005 is ringing
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.69:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKac162135657;received=192.168.1.69
From: sip:170956464@192.168.1.69;tag=1c162127078
To: sip:9813@192.168.1.224;user=phone;tag=as3038dfd8
Call-ID: 162125846312013142720@192.168.1.69
CSeq: 1 INVITE
Server: Asterisk PBX 11.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:9813@192.168.1.224:5060
Content-Length: 0

<------------>
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK6e678d9a
From: sip:170956464@192.168.1.224;tag=as270090b3
To: “C3T SIP” sip:9813@192.168.1.68;tag=15550
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 102 INVITE
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 133
Contact: sip:192.168.1.68:5060;transport=UDP
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER

v=0
o=C3T-SIP 3566208441 3566208441 IN IP4 192.168.1.68
s=-
c=IN IP4 192.168.1.68
t=0 0
m=audio 9562 RTP/AVP 0 8
a=sendrecv

<------------->
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c: — (12 headers 8 lines) —
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found RTP audio format 0
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Found RTP audio format 8
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Peer audio RTP is at port 192.168.1.68:9562
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: list_route: hop: sip:192.168.1.68:5060;transport=UDP
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: set_destination: Parsing sip:192.168.1.68:5060;transport=UDP for address/port to send to
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: set_destination: set destination to 192.168.1.68:5060
[Jan 3 14:27:10] VERBOSE[31134][C-00000002] chan_sip.c: Transmitting (no NAT) to 192.168.1.68:5060:
ACK sip:192.168.1.68:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK0432e6e9
Max-Forwards: 70
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Contact: sip:170956464@192.168.1.224:5060
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.2.0-rc1
Content-Length: 0

[Jan 3 14:27:10] VERBOSE[31677][C-00000002] app_dial.c: – SIP/argos-00000005 answered SIP/median2-00000004
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c: Audio is at 52072
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] chan_sip.c:
<— Reliably Transmitting (no NAT) to 192.168.1.69:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKac162135657;received=192.168.1.69
From: sip:170956464@192.168.1.69;tag=1c162127078
To: sip:9813@192.168.1.224;user=phone;tag=as3038dfd8
Call-ID: 162125846312013142720@192.168.1.69
CSeq: 1 INVITE
Server: Asterisk PBX 11.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:9813@192.168.1.224:5060
Content-Type: application/sdp
Content-Length: 180

v=0
o=root 136326475 136326475 IN IP4 192.168.1.68
s=Asterisk PBX 11.2.0-rc1
c=IN IP4 192.168.1.68
t=0 0
m=audio 9562 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

<------------>
[Jan 3 14:27:10] VERBOSE[31677][C-00000002] rtp_engine.c: – Remotely bridging SIP/median2-00000004 and SIP/argos-00000005
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.69:5060 —>
ACK sip:9813@192.168.1.224:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKac163765601
Max-Forwards: 70
From: sip:170956464@192.168.1.69;tag=1c162127078
To: sip:9813@192.168.1.224;user=phone;tag=as3038dfd8
Call-ID: 162125846312013142720@192.168.1.69
CSeq: 1 ACK
Contact: sip:170956464@192.168.1.69:5060
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 2000/v.6.40A.019.008
Content-Length: 0

<------------->
[Jan 3 14:27:10] VERBOSE[31134] chan_sip.c: — (12 headers 0 lines) —
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.68:5060:
OPTIONS sip:192.168.1.68 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK6aaab33c
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.224;tag=as0449188a
To: sip:192.168.1.68
Contact: sip:asterisk@192.168.1.224:5060
Call-ID: 66c469a52e398a6d137c6b3e0652a906@192.168.1.224:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.0-rc1
Date: Thu, 03 Jan 2013 13:27:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK6aaab33c
From: “asterisk” sip:asterisk@192.168.1.224;tag=as0449188a
To: sip:192.168.1.68
Call-ID: 66c469a52e398a6d137c6b3e0652a906@192.168.1.224:5060
CSeq: 102 OPTIONS
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 0

<------------->
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: — (9 headers 0 lines) —
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: Really destroying SIP dialog ‘66c469a52e398a6d137c6b3e0652a906@192.168.1.224:5060’ Method: OPTIONS
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.69:5060:
OPTIONS sip:192.168.1.69 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK52cd14b8
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.224;tag=as55aa41d9
To: sip:192.168.1.69
Contact: sip:asterisk@192.168.1.224:5060
Call-ID: 1c5aa2c3676b2b951bf098f145f10ba5@192.168.1.224:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.0-rc1
Date: Thu, 03 Jan 2013 13:27:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.69:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK52cd14b8
From: “asterisk” sip:asterisk@192.168.1.224;tag=as55aa41d9
To: sip:192.168.1.69;tag=1c253662751
Call-ID: 1c5aa2c3676b2b951bf098f145f10ba5@192.168.1.224:5060
CSeq: 102 OPTIONS
Contact: sip:192.168.1.69:5060;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.6.40A.019.008
X-Resources: telchs=145/5;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 243

v=0
o=AudiocodesGW 253673288 253672905 IN IP4 192.168.1.69
s=Phone-Call
c=IN IP4 192.168.1.69
t=0 0
m=audio 6000 RTP/AVP 8 0 4
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=ptime:30
a=sendrecv

<------------->
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: — (14 headers 12 lines) —
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: Really destroying SIP dialog ‘1c5aa2c3676b2b951bf098f145f10ba5@192.168.1.224:5060’ Method: OPTIONS
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.67:5060:
OPTIONS sip:192.168.1.67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK10e20bca
Max-Forwards: 70
From: “asterisk” sip:asterisk@192.168.1.224;tag=as61a8471c
To: sip:192.168.1.67
Contact: sip:asterisk@192.168.1.224:5060
Call-ID: 4695f0f4301498b83cde2b8c54386b46@192.168.1.224:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 11.2.0-rc1
Date: Thu, 03 Jan 2013 13:27:16 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.67:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK10e20bca
From: “asterisk” sip:asterisk@192.168.1.224;tag=as61a8471c
To: sip:192.168.1.67;tag=1c2120727776
Call-ID: 4695f0f4301498b83cde2b8c54386b46@192.168.1.224:5060
CSeq: 102 OPTIONS
Contact: sip:192.168.1.67:5060;expires=0
Supported: 100rel
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
Server: Audiocodes-Sip-Gateway-Mediant 2000/v.6.00A.041.001
X-Resources: telchs=103/47;mediachs=0/0
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Type: application/sdp
Content-Length: 298

v=0
o=AudiocodesGW 2120737400 2120737030 IN IP4 192.168.1.67
s=Phone-Call
c=IN IP4 192.168.1.67
t=0 0
m=audio 6000 RTP/AVP 8 0 4 96
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=no
a=rtpmap:96 telephone-event/8000
a=fmtp:96 0-15
a=ptime:30
a=sendrecv

<------------->
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: — (14 headers 14 lines) —
[Jan 3 14:27:16] VERBOSE[31134] chan_sip.c: Really destroying SIP dialog ‘4695f0f4301498b83cde2b8c54386b46@192.168.1.224:5060’ Method: OPTIONS
[Jan 3 14:27:17] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.67:5060 —>
OPTIONS sip:gateway@192.168.1.67 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.67;branch=z9hG4bKac2135226421
Max-Forwards: 70
From: sip:gateway@192.168.1.67:5060;tag=1c2135216541
To: sip:gateway@192.168.1.67
Call-ID: 2135214809312013142727@192.168.1.67
CSeq: 1 OPTIONS
Contact: sip:gateway@192.168.1.67:5060
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 2000/v.6.00A.041.001
Accept: application/sdp, application/simple-message-summary, message/sipfrag
Content-Length: 0

<------------->
[Jan 3 14:27:17] VERBOSE[31134] chan_sip.c: — (12 headers 0 lines) —
[Jan 3 14:27:17] VERBOSE[31134] chan_sip.c: Looking for gateway in default (domain 192.168.1.67)
[Jan 3 14:27:17] VERBOSE[31134] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.67:5060 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.1.67;branch=z9hG4bKac2135226421;received=192.168.1.67
From: sip:gateway@192.168.1.67:5060;tag=1c2135216541
To: sip:gateway@192.168.1.67;tag=as4a2dc2d6
Call-ID: 2135214809312013142727@192.168.1.67
CSeq: 1 OPTIONS
Server: Asterisk PBX 11.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0

<------------>
[Jan 3 14:27:17] VERBOSE[31134] chan_sip.c: Scheduling destruction of SIP dialog ‘2135214809312013142727@192.168.1.67’ in 32000 ms (Method: OPTIONS)
[Jan 3 14:27:28] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
REFER sip:170956464@192.168.1.224:5060 SIP/2.0
To: sip:170956464@192.168.1.224;tag=as270090b3
From: “C3T SIP” sip:9813@192.168.1.68;tag=15550
CSeq: 1 REFER
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK78f09f34c7279458fa2deb594d61374a
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
Contact: sip:999@192.168.1.68:5060;transport=UDP
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 0
Referred-By: sip:9813@192.168.1.68
Refer-To: sip:399@192.168.20.11?Replaces=2d57a27d9e52ec56ff692931aa15c45e%40192.168.1.68%3Bto-tag%3D87541%3Bfrom-tag%3D67413

<------------->
[Jan 3 14:27:28] VERBOSE[31134] chan_sip.c: — (12 headers 0 lines) —
[Jan 3 14:27:28] VERBOSE[31134][C-00000002] chan_sip.c: Call 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060 got a SIP call transfer from callee: (REFER)!
[Jan 3 14:27:28] VERBOSE[31134][C-00000002] chan_sip.c: SIP transfer to extension 399@ireflet by 9813@192.168.1.68
[Jan 3 14:27:28] VERBOSE[31134][C-00000002] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.68:5060 —>
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.1.68:5060;branch=z9hG4bK78f09f34c7279458fa2deb594d61374a;received=192.168.1.68
From: “C3T SIP” sip:9813@192.168.1.68;tag=15550
To: sip:170956464@192.168.1.224;tag=as270090b3
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 1 REFER
Server: Asterisk PBX 11.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:170956464@192.168.1.224:5060
Content-Length: 0

<------------>
[Jan 3 14:27:28] VERBOSE[31134][C-00000002] chan_sip.c: set_destination: Parsing sip:192.168.1.68:5060;transport=UDP for address/port to send to
[Jan 3 14:27:28] VERBOSE[31134][C-00000002] chan_sip.c: set_destination: set destination to 192.168.1.68:5060
[Jan 3 14:27:28] VERBOSE[31134][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.68:5060:
NOTIFY sip:192.168.1.68:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK3ff97677
Max-Forwards: 70
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Contact: sip:170956464@192.168.1.224:5060
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 103 NOTIFY
User-Agent: Asterisk PBX 11.2.0-rc1
Event: refer;id=1
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 49

SIP/2.0 481 Call leg/transaction does not exist

[Jan 3 14:27:28] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK3ff97677
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 103 NOTIFY
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 0

<------------->
[Jan 3 14:27:28] VERBOSE[31134] chan_sip.c: — (9 headers 0 lines) —
[Jan 3 14:27:28] NOTICE[31134][C-00000002] chan_sip.c: Got OK on REFER Notify message
[Jan 3 14:27:32] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.69:5060 —>
BYE sip:9813@192.168.1.224:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKac517262979
Max-Forwards: 70
From: sip:170956464@192.168.1.69;tag=1c162127078
To: sip:9813@192.168.1.224;user=phone;tag=as3038dfd8
Call-ID: 162125846312013142720@192.168.1.69
CSeq: 2 BYE
Supported: em,timer,replaces,path,resource-priority
Allow: REGISTER,OPTIONS,INVITE,ACK,CANCEL,BYE,NOTIFY,PRACK,REFER,INFO,SUBSCRIBE,UPDATE
User-Agent: Audiocodes-Sip-Gateway-Mediant 2000/v.6.40A.019.008
Reason: Q.850 ;cause=16
Content-Length: 0

<------------->
[Jan 3 14:27:32] VERBOSE[31134] chan_sip.c: — (12 headers 0 lines) —
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Sending to 192.168.1.69:5060 (no NAT)
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog ‘162125846312013142720@192.168.1.69’ in 6400 ms (Method: BYE)
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c:
<— Transmitting (no NAT) to 192.168.1.69:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.69;branch=z9hG4bKac517262979;received=192.168.1.69
From: sip:170956464@192.168.1.69;tag=1c162127078
To: sip:9813@192.168.1.224;user=phone;tag=as3038dfd8
Call-ID: 162125846312013142720@192.168.1.69
CSeq: 2 BYE
Server: Asterisk PBX 11.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------>
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: set_destination: Parsing sip:192.168.1.68:5060;transport=UDP for address/port to send to
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: set_destination: set destination to 192.168.1.68:5060
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: Audio is at 53554
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: Adding codec 100003 (ulaw) to SDP
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.68:5060:
INVITE sip:192.168.1.68:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK7d0f4f3f
Max-Forwards: 70
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Contact: sip:170956464@192.168.1.224:5060
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 11.2.0-rc1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 185

v=0
o=root 1130325334 1130325335 IN IP4 192.168.1.224
s=Asterisk PBX 11.2.0-rc1
c=IN IP4 192.168.1.224
t=0 0
m=audio 53554 RTP/AVP 0
a=rtpmap:0 PCMU/8000
a=ptime:20
a=sendrecv

[Jan 3 14:27:32] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK7d0f4f3f
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 104 INVITE
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 133
Contact: sip:192.168.1.68:5060;transport=UDP
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER

v=0
o=C3T-SIP 3566208441 3566208442 IN IP4 192.168.1.68
s=-
c=IN IP4 192.168.1.68
t=0 0
m=audio 9562 RTP/AVP 0 8
a=sendrecv

<------------->
[Jan 3 14:27:32] VERBOSE[31134] chan_sip.c: — (12 headers 8 lines) —
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Found RTP audio format 0
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Found RTP audio format 8
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Capabilities: us - (ulaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw)
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x0 (nothing), combined - 0x0 (nothing)
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Peer audio RTP is at port 192.168.1.68:9562
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: set_destination: Parsing sip:192.168.1.68:5060;transport=UDP for address/port to send to
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: set_destination: set destination to 192.168.1.68:5060
[Jan 3 14:27:32] VERBOSE[31134][C-00000002] chan_sip.c: Transmitting (no NAT) to 192.168.1.68:5060:
ACK sip:192.168.1.68:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK17b001e3
Max-Forwards: 70
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Contact: sip:170956464@192.168.1.224:5060
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 11.2.0-rc1
Content-Length: 0

[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: Scheduling destruction of SIP dialog ‘6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060’ in 6400 ms (Method: REFER)
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: set_destination: Parsing sip:192.168.1.68:5060;transport=UDP for address/port to send to
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: set_destination: set destination to 192.168.1.68:5060
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] chan_sip.c: Reliably Transmitting (no NAT) to 192.168.1.68:5060:
BYE sip:192.168.1.68:5060;transport=UDP SIP/2.0
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK232b44cb
Max-Forwards: 70
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 105 BYE
User-Agent: Asterisk PBX 11.2.0-rc1
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

[Jan 3 14:27:32] VERBOSE[31134] chan_sip.c:
<— SIP read from UDP:192.168.1.68:38860 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.224:5060;branch=z9hG4bK232b44cb
From: sip:170956464@192.168.1.224;tag=as270090b3
To: sip:9813@192.168.1.68;tag=15550
Call-ID: 6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060
CSeq: 105 BYE
User-Agent: C3T SIP Stack 1.2
Max-Forwards: 70
Content-Length: 0

<------------->
[Jan 3 14:27:32] VERBOSE[31134] chan_sip.c: — (9 headers 0 lines) —
[Jan 3 14:27:32] VERBOSE[31134] chan_sip.c: Really destroying SIP dialog ‘6535801523e988a9179e5dbf3de2925a@192.168.1.224:5060’ Method: REFER
[Jan 3 14:27:32] VERBOSE[31677][C-00000002] pbx.c: == Spawn extension (median, 9813, 2) exited non-zero on ‘SIP/median2-00000004’
[/code]

Thanks a lot for help !

Antoine

2

I would really try to avoid non-B2BUA type operations.

However, I believe the basic principle is that you you have to configure Asterisk with a list of domain parts it should consider local. I believe, if you do that, if it finds a refer replacing a non-existent call, and the domain part is not in the list, it will generate an INVITE/replaces.

Finding the details is left as an exercise to the reader.

3

First off all if you wanna use SIP REFER, you have to have it enabled and supported
by your provider on your trunk…

4

Hi,

Thanks for replies.

I’m sure it can because it works perfecty with a GW audiocode median 2000 without asterisk :

A (median 2000) ==> calls B (Sip Provider)
B answer the call and send a SIP Refer to Transfer to C (SIP Softphone)
After that we have : A in communication with C

What I try with Asterisk : (asterisk is between A and B)

A (median 2000) ==> calls asterisk
asterisk calls B (Sip Provider) : bridging channels OK
Then B send a SIP Refer to Transfer to C, but the Transfer fails

Thanks,

Antoine

5

How you do it from dialplan? You use Transfer application? Something like:
same => n,Transfer(SIP/${EXTEN})
where ${EXTEN} is number to be called (C number in your example)

6

Hi,

B handle the transfer , an attended transfer to C, but B is not contacting C through asterisk, B send invites directly to C, and when C answer the call , B send to asterisk the Refer with Replace method.

asterisk calls B with a simple Dial, Dial(SIP/argos/9813) in my dialplan.

It could be OK if instated of a ‘Dial’ I use the asterisk’s Transfer application, BUT I want asterisk to handle SIP and media. The AudioCodes median 2000 will no longer exist in the future.

Antoine

7

lun: This is an incoming REFER. You are talking about outgoing REFERs.

antioine: The options you need to investigate are domain and autodomain. They must exclude the destination domain of the transfer.

8

Hi,

Setting ‘autodomain=yes’ with right extensions and contexts in the dialplan fix the pb !
Thank you man, you’re a genius.

Later,

Antoine

9

Is this something related to SIP UUI where the REFER method is used?

So If I use autodomain and TRANSFER application can I adhere SIP UUI?

Please advice.

Thanks.

10

No.