Hi,
I've a probleme since few weeks that I don't be able to solve.
I use Thomson ST2030 phone and I've an error when I want to do an
attended transfer with the soft key.
The receiver of the transfer return an : Got SIP response 400 "Bad
Request" back from 192.168.2.13
The direct transfer with soft key works fine and attended transfer with
*2 (features.conf) works too.
Can you help me ?
This is what I've during a sip debug of the receiver of the transfer.
localhost*CLI> sip debug ip 192.168.2.13
SIP Debugging Enabled for IP: 192.168.2.13
-- Executing Macro("SIP/9714-08ade6a8", "externe|[TEL
NUMBER]|[NAME]
<[TEL NUMBER]>") in new stack
-- Executing Set("SIP/9714-08ade6a8", "CALLERID(all)=[NAME]
<[TEL
NUMBER]>") in new stack
-- Executing Dial("SIP/9714-08ade6a8", "Zap/g1/[TEL
NUMBER]||tT") in
new stack
-- Requested transfer capability: 0x00 - SPEECH
-- Called g1/[TEL NUMBER]
-- Zap/2-1 is proceeding passing it to SIP/9714-08ade6a8
-- Zap/2-1 is ringing
-- Zap/2-1 answered SIP/9714-08ade6a8
-- Started music on hold, class 'default', on channel
'Zap/2-1'
-- Stopped music on hold on Zap/2-1
-- Executing Macro("SIP/9714-08affc58", "local|9710") in
new stack
-- Executing Answer("SIP/9714-08affc58", "") in new
stack
-- Executing Dial("SIP/9714-08affc58", "SIP/9710|20|tT")
in new stack
Apr 11 17:43:59 NOTICE[22239]: chan_sip.c:2142 sip_call: called party
number = 9710
We're at 192.168.2.254 port 10330
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.2.13:5060:
INVITE sip:9710 at 192.168.2.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]" <sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 11 Apr 2008 15:43:59 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216
v=0
o=root 2453 2453 IN IP4 192.168.2.254
s=session
c=IN IP4 192.168.2.254
t=0 0
m=audio 10330 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
---
-- Called 9710
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]"<sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 102 INVITE
Content-Length: 0
--- (7 headers 0 lines) ---
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]"<sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 102 INVITE
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:9710 at 192.168.2.13:5060;user=phone>
Content-Length: 0
--- (9 headers 0 lines) ---
-- SIP/9710-08aec0b8 is ringing
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK655cd59a;rport
From: "[USER NAME]"<sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 102 INVITE
Allow:
INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: <sip:9710 at 192.168.2.13:5060;user=phone>
Content-Type: application/sdp
Content-Length: 199
v=0
o=9710 6696599 6696599 IN IP4 192.168.2.13
s=-
c=IN IP4 192.168.2.13
t=0 0
m=audio 41000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
--- (10 headers 10 lines) ---
Found RTP audio format 8
Found RTP audio format 101
Peer audio RTP is at port 192.168.2.13:41000
Found description format PCMA
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0
(nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1
(telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:9710 at 192.168.2.13:5060;user=phone>
set_destination: Parsing <sip:9710 at 192.168.2.13:5060;user=phone> for
address/port to send to
set_destination: set destination to 192.168.2.13, port 5060
Transmitting (no NAT) to 192.168.2.13:5060:
ACK sip:9710 at 192.168.2.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK7a52f4a4;rport
From: "[USER NAME]" <sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
-- SIP/9710-08aec0b8 answered SIP/9714-08affc58
-- Started music on hold, class 'default', on channel
'SIP/9710-08aec0b8'
-- Stopped music on hold on SIP/9710-08aec0b8
set_destination: Parsing <sip:9710 at 192.168.2.13:5060;user=phone> for
address/port to send to
set_destination: set destination to 192.168.2.13, port 5060
Transmitting (no NAT) to 192.168.2.13:5060:
INFO sip:9710 at 192.168.2.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK7a6adfe4;rport
From: "[USER NAME]" <sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 103 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Type: message/sipfrag
Content-Length: 32
From: "9710"
To: "[TEL NUMBER]"
---
== Spawn extension (macro-externe, s, 2) exited non-zero on
'SIP/9714-08ade6a8' in macro 'externe'
== Spawn extension (macro-externe, s, 2) exited non-zero on
'SIP/9714-08ade6a8'
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 400 Bad Request
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK7a6adfe4;rport
From: "[USER NAME]"<sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 103 INFO
Content-Length: 0
--- (7 headers 0 lines) ---
-- Got SIP response 400 "Bad Request" back from 192.168.2.13
Scheduling destruction of call
'7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254' in 32000 ms
set_destination: Parsing <sip:9710 at 192.168.2.13:5060;user=phone> for
address/port to send to
set_destination: set destination to 192.168.2.13, port 5060
Reliably Transmitting (no NAT) to 192.168.2.13:5060:
BYE sip:9710 at 192.168.2.13:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK0f8cf4b2;rport
From: "[USER NAME]" <sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 104 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0
---
== Spawn extension (macro-local, s, 2) exited non-zero on 'Zap/2-1' in
macro 'local'
== Spawn extension (macro-local, s, 2) exited non-zero on 'Zap/2-1'
-- Hungup 'Zap/2-1'
localhost*CLI>
<-- SIP read from 192.168.2.13:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK0f8cf4b2;rport
From: "[USER NAME]"<sip:[TEL
NUMBER]@192.168.2.254>;tag=as548073e8
To: <sip:9710 at 192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254
CSeq: 104 BYE
Content-Length: 0
--- (7 headers 0 lines) ---
Destroying call '7c0d17bd14ef93d945d447433cff22d8 at 192.168.2.254'
-- Channel 0/2, span 1 received AOC-E charging 0 units
localhost*CLI>
--
Cordialement,
Mathieu