Problems with attended transfer - ST2030 - SIP Error

Asterisk Asterisk Support
1

Hi,

I’ve an asterisk server with 2 Junghanns quadBri and Bristuff drivers.

My server works fine but attended transfer make with Thomson ST2030 doesn’t work.

Direct transfer works properly, but when I want to make an attended transfer this message appears on the CLI :

Got SIP response 400 “Bad Request” back from xxx.xxx.xxx.xxx

Here the complete sequence in my CLI :

– Called g1/06xxxxxxxx
– Zap/2-1 is proceeding passing it to SIP/9714-b6f1e668
– Zap/2-1 is ringing
– Zap/2-1 answered SIP/9714-b6f1e668
– Started music on hold, class ‘default’, on channel ‘Zap/2-1’
– Stopped music on hold on Zap/2-1
– Executing Macro(“SIP/9714-b6f0f748”, “local|9710”) in new stack
– Executing Answer(“SIP/9714-b6f0f748”, “”) in new stack
– Executing Dial(“SIP/9714-b6f0f748”, “SIP/9710|20|tT”) in new stack
Mar 31 17:24:23 NOTICE[17774]: chan_sip.c:2142 sip_call: called party number = 9710
– Called 9710
– SIP/9710-08af4f90 is ringing
– SIP/9710-08af4f90 answered SIP/9714-b6f0f748
– Started music on hold, class ‘default’, on channel ‘SIP/9710-08af4f90’
– Stopped music on hold on SIP/9710-08af4f90
== Spawn extension (macro-externe, s, 2) exited non-zero on ‘SIP/9714-b6f1e668’ in macro ‘externe’
== Spawn extension (macro-externe, s, 2) exited non-zero on ‘SIP/9714-b6f1e668’
– Got SIP response 400 “Bad Request” back from 192.168.2.13
== 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’
– Channel 0/2, span 1 received AOC-E charging 0 units

I’ve make a capture with tcpdum and i’ve find the last message send by asterisk to the receiver of the transfer.
Asterisk send to him a request INFO during an attended transfer and a INVITE during direct transfer.

May you help me ?

PS: Excuse my english, it’s not my native language.

2

If someone can help me ?

First is the caller who want to make a attended transfer.
Second is the callee who get the transfer.

[code]localhost*CLI> sip debug ip 192.168.2.34
SIP Debugging Enabled for IP: 192.168.2.34

localhost*CLI>

<-- SIP read from 192.168.2.34:5060:
INVITE sip:[TEL NUMBER]@192.168.2.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK253581925314869275-6484034
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 1 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: sip:9714@192.168.2.34:5060;user=phone
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270

v=0
o=9714 6484034 6484034 IN IP4 192.168.2.34
s=-
c=IN IP4 192.168.2.34
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv

— (15 headers 13 lines) —
Using INVITE request as basis request - 183e6e2-c0a80101-0-9@192.168.2.34
Sending to 192.168.2.34 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK253581925314869275-6484034;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as79dbe6ab
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="799bf269"
Content-Length: 0

Scheduling destruction of call ‘183e6e2-c0a80101-0-9@192.168.2.34’ in 15000 ms
Found user '9714’
localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
ACK sip:[TEL NUMBER]@192.168.2.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK253581925314869275-6484034
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as79dbe6ab
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 1 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (10 headers 0 lines) —

<-- SIP read from 192.168.2.34:5060:
INVITE sip:[TEL NUMBER]@192.168.2.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK7030818652687532643-6484045
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 2 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“799bf269”, uri=“sip:[TEL NUMBER]@192.168.2.254:5060;user=phone”, response=“61737cb1fcc40ed0f540138de5afd5e4”, algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: sip:9714@192.168.2.34:5060;user=phone
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270

v=0
o=9714 6484034 6484034 IN IP4 192.168.2.34
s=-
c=IN IP4 192.168.2.34
t=0 0
m=audio 41000 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv

— (16 headers 13 lines) —
Using INVITE request as basis request - 183e6e2-c0a80101-0-9@192.168.2.34
Sending to 192.168.2.34 : 5060 (non-NAT)
Found user '9714’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Peer audio RTP is at port 192.168.2.34:41000
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for [TEL NUMBER] in thomsonint (domain 192.168.2.254)
list_route: hop: sip:9714@192.168.2.34:5060;user=phone
Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK7030818652687532643-6484045;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Content-Length: 0

We’re at 192.168.2.254 port 10668
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK7030818652687532643-6484045;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Content-Type: application/sdp
Content-Length: 213

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 10668 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -

Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK7030818652687532643-6484045;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Content-Length: 0

We’re at 192.168.2.254 port 10668
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK7030818652687532643-6484045;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 2453 2454 IN IP4 192.168.2.254
s=session
c=IN IP4 192.168.2.254
t=0 0
m=audio 10668 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
ACK sip:[TEL NUMBER]@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK3586474258647198208-6489267
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 2 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“799bf269”, uri=“sip:[TEL NUMBER]@192.168.2.254;user=phone”, response=“b52b60c94b9468e1e2531b7685554963”, algorithm=MD5
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (11 headers 0 lines) —

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
INVITE sip:[TEL NUMBER]@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK2585703147536081598-6490890
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 3 INVITE
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“799bf269”, uri=“sip:[TEL NUMBER]@192.168.2.254;user=phone”, response=“9ab61703c0e2ecec71559a557df53bde”, algorithm=MD5
Contact: sip:9714@192.168.2.34:5060;user=phone
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Type: application/sdp
Content-Length: 196

v=0
o=9714 6484034 6484035 IN IP4 192.168.2.34
s=-
c=IN IP4 192.168.2.34
t=0 0
m=audio 41000 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendonly

— (13 headers 10 lines) —
Using INVITE request as basis request - 183e6e2-c0a80101-0-9@192.168.2.34
Sending to 192.168.2.34 : 5060 (non-NAT)
Found RTP audio format 8
Found RTP audio format 97
Peer audio RTP is at port 192.168.2.34:41000
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)
– Started music on hold, class ‘default’, on channel 'Zap/1-1’
We’re at 192.168.2.254 port 10668
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK2585703147536081598-6490890;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 2453 2455 IN IP4 192.168.2.254
s=session
c=IN IP4 192.168.2.254
t=0 0
m=audio 10668 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
ACK sip:[TEL NUMBER]@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK5708036470869314720-6490899
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 3 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“799bf269”, uri=“sip:[TEL NUMBER]@192.168.2.254;user=phone”, response=“b52b60c94b9468e1e2531b7685554963”, algorithm=MD5
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (11 headers 0 lines) —
– Stopped music on hold on Zap/1-1
– Executing Macro(“SIP/9746-08affc58”, “local|9731”) in new stack
– Executing Answer(“SIP/9746-08affc58”, “”) in new stack
– Executing Dial(“SIP/9746-08affc58”, “SIP/9731|20|tT”) in new stack
Apr 11 17:38:47 NOTICE[22111]: chan_sip.c:2142 sip_call: called party number = 9731
– Called 9731
– SIP/9731-08b21348 is ringing
== Spawn extension (macro-local, s, 2) exited non-zero on ‘SIP/9738-08afcd90’ in macro ‘local’
== Spawn extension (macro-local, s, 2) exited non-zero on ‘SIP/9738-08afcd90’
– Zap/4-1 answered SIP/9713-08ad01f8
localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
INVITE sip:9710@192.168.2.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK8070259792092547043-6498581
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 1 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: sip:9714@192.168.2.34:5060;user=phone
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270

v=0
o=9714 6498581 6498581 IN IP4 192.168.2.34
s=-
c=IN IP4 192.168.2.34
t=0 0
m=audio 41002 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv

— (15 headers 13 lines) —
Using INVITE request as basis request - 1e70ff6-c0a80101-0-a@192.168.2.34
Sending to 192.168.2.34 : 5060 (non-NAT)
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK8070259792092547043-6498581;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone;tag=as6e823519
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 1 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Proxy-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce="5d018380"
Content-Length: 0

Scheduling destruction of call ‘1e70ff6-c0a80101-0-a@192.168.2.34’ in 15000 ms
Found user '9714’
localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
ACK sip:9710@192.168.2.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK8070259792092547043-6498581
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone;tag=as6e823519
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 1 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (10 headers 0 lines) —

<-- SIP read from 192.168.2.34:5060:
INVITE sip:9710@192.168.2.254:5060;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK3585804247547092598-6498591
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 2 INVITE
Max-Forwards: 70
Supported: timer, replaces
Session-Expires: 1800
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“5d018380”, uri=“sip:9710@192.168.2.254:5060;user=phone”, response=“644c0ec524e3953a4a74086dc9d13d10”, algorithm=MD5
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: sip:9714@192.168.2.34:5060;user=phone
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Content-Type: application/sdp
Content-Length: 270

v=0
o=9714 6498581 6498581 IN IP4 192.168.2.34
s=-
c=IN IP4 192.168.2.34
t=0 0
m=audio 41002 RTP/AVP 8 0 18 4 97
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=rtpmap:4 G723/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendrecv

— (16 headers 13 lines) —
Using INVITE request as basis request - 1e70ff6-c0a80101-0-a@192.168.2.34
Sending to 192.168.2.34 : 5060 (non-NAT)
Found user '9714’
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 97
Peer audio RTP is at port 192.168.2.34:41002
Found description format PCMA
Found description format PCMU
Found description format G729
Found description format G723
Found description format telephone-event
Capabilities: us - 0x8 (alaw), peer - audio=0x10d (g723|ulaw|alaw|g729)/video=0x0 (nothing), combined - 0x8 (alaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Looking for 9710 in thomsonint (domain 192.168.2.254)
list_route: hop: sip:9714@192.168.2.34:5060;user=phone
Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK3585804247547092598-6498591;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:9710@192.168.2.254
Content-Length: 0

-- Executing Macro("SIP/9714-08b1ab08", "local|9710") in new stack
-- Executing Answer("SIP/9714-08b1ab08", "") in new stack

We’re at 192.168.2.254 port 19122
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK3585804247547092598-6498591;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone;tag=as066e52a7
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 2 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:9710@192.168.2.254
Content-Type: application/sdp
Content-Length: 213

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 19122 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -

-- Executing Dial("SIP/9714-08b1ab08", "SIP/9710|20|tT") in new stack

Apr 11 17:38:59 NOTICE[22122]: chan_sip.c:2142 sip_call: called party number = 9710
– Called 9710
localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
ACK sip:9710@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK3586474258647198198-6498605
From: "[USER NAME]“sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone;tag=as066e52a7
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 2 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“5d018380”, uri="sip:9710@192.168.2.254;user=phone”, response=“6e7d4d802139bc2bf16a35621defe8aa”, algorithm=MD5
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (11 headers 0 lines) —
– SIP/9710-08aeffd8 is ringing
– Hungup ‘Zap/4-1’
== Spawn extension (macro-externe, s, 2) exited non-zero on ‘SIP/9713-08ad01f8’ in macro ‘externe’
== Spawn extension (macro-externe, s, 2) exited non-zero on ‘SIP/9713-08ad01f8’
– Channel 0/1, span 2 received AOC-E charging 0 units
– Nobody picked up in 20000 ms
– Executing Goto(“SIP/9746-08affc58”, “s-NOANSWER|1”) in new stack
– Goto (macro-local,s-NOANSWER,1)
– Executing Macro(“SIP/9746-08affc58”, “accueil”) in new stack
– Executing Answer(“SIP/9746-08affc58”, “”) in new stack
– Executing GotoIfTime(“SIP/9746-08affc58”, “|mon||?lundi|1") in new stack
– Executing GotoIfTime(“SIP/9746-08affc58”, "|tue||?mardi|1”) in new stack
– Executing GotoIfTime(“SIP/9746-08affc58”, “|wed||?mercredi|1") in new stack
– Executing GotoIfTime(“SIP/9746-08affc58”, "|thu||?jeudi|1”) in new stack
– Executing GotoIfTime(“SIP/9746-08affc58”, "|fri||?vendredi|1") in new stack
– Goto (macro-accueil,vendredi,1)
– Executing Dial(“SIP/9746-08affc58”, “SIP/9745&SIP/9743&SIP/9702|20|tT”) in new stack
Apr 11 17:39:08 NOTICE[22111]: chan_sip.c:2142 sip_call: called party number = 9745
– Called 9745
Apr 11 17:39:08 NOTICE[22111]: chan_sip.c:2142 sip_call: called party number = 9743
– Called 9743
Apr 11 17:39:08 NOTICE[22111]: chan_sip.c:2142 sip_call: called party number = 9702
– Called 9702
– SIP/9702-08b35868 is ringing
– SIP/9745-08ad01f8 is ringing
– SIP/9743-08ae25c8 is ringing
– SIP/9710-08aeffd8 answered SIP/9714-08b1ab08
– SIP/9745-08ad01f8 answered SIP/9746-08affc58
localhostCLI>
<-- SIP read from 192.168.2.34:5060:
INVITE sip:9710@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK5708696470469310320-6505767
From: "[USER NAME]“sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone;tag=as066e52a7
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 3 INVITE
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“5d018380”, uri="sip:9710@192.168.2.254;user=phone”, response=“aba700fd105db1f769975ffa75085ae6”, algorithm=MD5
Contact: sip:9714@192.168.2.34:5060;user=phone
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Type: application/sdp
Content-Length: 196

v=0
o=9714 6498581 6498582 IN IP4 192.168.2.34
s=-
c=IN IP4 192.168.2.34
t=0 0
m=audio 41002 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendonly

— (13 headers 10 lines) —
Using INVITE request as basis request - 1e70ff6-c0a80101-0-a@192.168.2.34
Sending to 192.168.2.34 : 5060 (non-NAT)
Found RTP audio format 8
Found RTP audio format 97
Peer audio RTP is at port 192.168.2.34:41002
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)
– Started music on hold, class ‘default’, on channel 'SIP/9710-08aeffd8’
We’re at 192.168.2.254 port 19122
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK5708696470469310320-6505767;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone;tag=as066e52a7
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 3 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: sip:9710@192.168.2.254
Content-Type: application/sdp
Content-Length: 213

v=0
o=root 2453 2454 IN IP4 192.168.2.254
s=session
c=IN IP4 192.168.2.254
t=0 0
m=audio 19122 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-16
a=silenceSupp:off - - - -

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
ACK sip:9710@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK1363682026325870376-6505779
From: "[USER NAME]“sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
To: sip:9710@192.168.2.254:5060;user=phone;tag=as066e52a7
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 3 ACK
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“5d018380”, uri="sip:9710@192.168.2.254;user=phone”, response=“6e7d4d802139bc2bf16a35621defe8aa”, algorithm=MD5
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (11 headers 0 lines) —
– Stopped music on hold on SIP/9710-08aeffd8
localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
REFER sip:[TEL NUMBER]@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK253581925314865875-6506270
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 4 REFER
Max-Forwards: 70
Referred-By: "[USER NAME]"sip:9714@192.168.2.254
Refer-To: sip:9710@192.168.2.254?Replaces=1e70ff6-c0a80101-0-a%40192.168.2.34%3Bto-tag%3Das066e52a7%3Bfrom-tag%3Dc0a80101-632914
Contact: sip:9714@192.168.2.34:5060;user=phone
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“799bf269”, uri=“sip:[TEL NUMBER]@192.168.2.254;user=phone”, response=“9f73092351a9047d09822b9a43b6cd8c”, algorithm=MD5
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (14 headers 0 lines) —
Transfer to 9710 in thomsonint
Transfer from 9714 in thomsonint
Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 202 Accepted
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK253581925314865875-6506270;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 4 REFER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing

set_destination: Parsing sip:9714@192.168.2.34:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.2.34, port 5060
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
NOTIFY sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 16

SIP/2.0 200 OK

set_destination: Parsing sip:9714@192.168.2.34:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.2.34, port 5060
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
BYE sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK033ed1ab;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
Content-Length: 0

Scheduling destruction of call ‘1e70ff6-c0a80101-0-a@192.168.2.34’ in 32000 ms
set_destination: Parsing sip:9714@192.168.2.34:5060;user=phone for address/port to send to
set_destination: set destination to 192.168.2.34, port 5060
Reliably Transmitting (no NAT) to 192.168.2.34:5060:
BYE sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK62a0ee69;rport
From: sip:9710@192.168.2.254:5060;user=phone;tag=as066e52a7
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 102 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0

== Spawn extension (macro-externe, s, 2) exited non-zero on ‘SIP/9714-08adb7e0’ in macro ‘externe’
== Spawn extension (macro-externe, s, 2) exited non-zero on ‘SIP/9714-08adb7e0’
– Got SIP response 400 “Bad Request” back from 192.168.2.13
== Spawn extension (macro-local, s, 2) exited non-zero on ‘Zap/1-1’ in macro ‘local’
== Spawn extension (macro-local, s, 2) exited non-zero on ‘Zap/1-1’
– Hungup 'Zap/1-1’
localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: “[TEL NUMBER]”<sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
Content-Length: 0

— (7 headers 0 lines) —
localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
BYE sip:[TEL NUMBER]@192.168.2.254;user=phone SIP/2.0
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK8031369703192647053-6506283
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 5 BYE
Max-Forwards: 70
Allow-Events: refer,dialog,message-summary,check-sync,talk,hold
Proxy-Authorization: Digest username=“9714”, realm=“asterisk”, nonce=“799bf269”, uri=“sip:[TEL NUMBER]@192.168.2.254;user=phone”, response=“e43a167926a34824aa2967b58ad86e8c”, algorithm=MD5
User-Agent: THOMSON ST2030 hw5 fw1.59 00-14-7F-E1-3E-26
Content-Length: 0

— (11 headers 0 lines) —
Sending to 192.168.2.34 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.34:5060;branch=z9hG4bK8031369703192647053-6506283;received=192.168.2.34
From: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
To: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 5 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Content-Length: 0

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK033ed1ab;rport
From: “[TEL NUMBER]”<sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 103 BYE
Content-Length: 0

— (7 headers 0 lines) —

<-- SIP read from 192.168.2.34:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK62a0ee69;rport
From: "9710"sip:9710@192.168.2.254:5060;user=phone;tag=as066e52a7
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-632914
Call-ID: 1e70ff6-c0a80101-0-a@192.168.2.34
CSeq: 102 BYE
Content-Length: 0

— (7 headers 0 lines) —
Destroying call '1e70ff6-c0a80101-0-a@192.168.2.34’
Retransmitting #1 (no NAT) to 192.168.2.34:5060:
NOTIFY sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 16

SIP/2.0 200 OK

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
Content-Length: 0

— (7 headers 0 lines) —
Retransmitting #2 (no NAT) to 192.168.2.34:5060:
NOTIFY sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 16

SIP/2.0 200 OK

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
Content-Length: 0

— (7 headers 0 lines) —
– Channel 0/1, span 1 received AOC-E charging 0 units
Retransmitting #3 (no NAT) to 192.168.2.34:5060:
NOTIFY sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 16

SIP/2.0 200 OK

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
Content-Length: 0

— (7 headers 0 lines) —
Retransmitting #4 (no NAT) to 192.168.2.34:5060:
NOTIFY sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 16

SIP/2.0 200 OK

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
Content-Length: 0

— (7 headers 0 lines) —
Retransmitting #5 (no NAT) to 192.168.2.34:5060:
NOTIFY sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 16

SIP/2.0 200 OK

localhost*CLI> sip debug ip 192.168.2.34
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
Content-Length: 0

— (7 headers 0 lines) —
localhost*CLI> sip no debug ip 192.168.2.34
Usage: sip no debug
Disables dumping of SIP packets for debugging purposes
Retransmitting #6 (no NAT) to 192.168.2.34:5060:
NOTIFY sip:9714@192.168.2.34:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: "[USER NAME]"sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
User-Agent: Asterisk PBX
Max-Forwards: 70
Event: refer;id=4
Subscription-state: terminated;reason=noresource
Content-Type: message/sipfrag;version=2.0
Content-Length: 16

SIP/2.0 200 OK

localhost*CLI>
<-- SIP read from 192.168.2.34:5060:
SIP/2.0 500 Internal Server Error
Via: SIP/2.0/UDP 192.168.2.254:5060;branch=z9hG4bK23a53af4;rport
From: <sip:[TEL NUMBER]@192.168.2.254:5060;user=phone>;tag=as4c775d9e
To: sip:9714@192.168.2.254:5060;user=phone;tag=c0a80101-62f041
Call-ID: 183e6e2-c0a80101-0-9@192.168.2.34
CSeq: 102 NOTIFY
Content-Length: 0

— (7 headers 0 lines) —
Apr 11 17:39:34 WARNING[18334]: chan_sip.c:1256 retrans_pkt: Maximum retries exceeded on transmission 183e6e2-c0a80101-0-9@192.168.2.34 for seqno 102 (Non-critical Request)
Destroying call ‘183e6e2-c0a80101-0-9@192.168.2.34’

localhostCLI> sip no debug
SIP Debugging Disabled
localhostCLI>[/code]

[code]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@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@192.168.2.13:5060;user=phone
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@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@192.168.2.13:5060;user=phone
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@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@192.168.2.13:5060;user=phone;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@192.168.2.254
CSeq: 102 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: sip:9710@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@192.168.2.13:5060;user=phone;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@192.168.2.254
CSeq: 102 INVITE
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,SUBSCRIBE,NOTIFY,UPDATE,REFER,REGISTER,INFO
Contact: sip:9710@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@192.168.2.13:5060;user=phone
set_destination: Parsing sip:9710@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@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@192.168.2.13:5060;user=phone;tag=c0a80101-6628fd
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@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@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@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@192.168.2.13:5060;user=phone;tag=c0a80101-6628fd
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@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@192.168.2.13:5060;user=phone;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@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@192.168.2.254’ in 32000 ms
set_destination: Parsing sip:9710@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@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@192.168.2.13:5060;user=phone;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@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@192.168.2.13:5060;user=phone;tag=c0a80101-6628fd
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@192.168.2.254
CSeq: 104 BYE
Content-Length: 0

— (7 headers 0 lines) —
Destroying call ‘7c0d17bd14ef93d945d447433cff22d8@192.168.2.254’
– Channel 0/2, span 1 received AOC-E charging 0 units
localhost*CLI>
[/code]

3

I’m facing the same problem. Asterisk version is 1.2.26-BRIstuffed-0.3.0-PRE-1y-q. On Asterisk 1.2.24 it works w/o problems.

Udo

4 
  -- 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@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@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@192.168.2.13:5060;user=phone>;tag=c0a80101-6628fd
Contact: <sip:[TEL NUMBER]@192.168.2.254>
Call-ID: 7c0d17bd14ef93d945d447433cff22d8@192.168.2.254
CSeq: 103 INFO
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Type: message/sipfrag
Content-Length: 32

From: "9710"             <---------- the problem
To: "[TEL NUMBER]"   <----------

the bristuff patch change de attended transfer process.
With version without bristuff, works. (and old bristuff patchs)