Ok.
Sorry but this was an excerpt of the sip debug…I think my Putty had wrong setting and it delete the old lines.
The full sip debug: I call the Number, answer the call, set the call on hold, get the call back and hang up:
[code]— (7 headers 0 lines) —
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 487 Request Terminated
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK094c39be
Call-ID: 1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as34125ff0
To: sip:2000@CallerIP;ob;tag=eCxgGVvVAU2LgHAjSgMmi.iVRzwPyxI7
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Transmitting (NAT) to CallerIP:5060:
ACK sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK094c39be;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as34125ff0
To: sip:2000@CallerIP:5060;ob;tag=eCxgGVvVAU2LgHAjSgMmi.iVRzwPyxI7
Contact: sip:anonymous@ServerIP:5060
Call-ID: 1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
Scheduling destruction of SIP dialog ‘1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060’ in 32000 ms (Method: INVITE)
Really destroying SIP dialog ‘3ef8b2763aaf72746610a54554629d50@sipgate.de’ Method: BYE
[2012-08-06 11:42:57] NOTICE[27943]: chan_sip.c:13151 sip_reregister: – Re-registration for Nr@sipgate.de
REGISTER 11 headers, 0 lines
Reliably Transmitting (NAT) to sipgateIP:5060:
REGISTER sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK792dd0d5;rport
Max-Forwards: 70
From: sip:Nr@sipgate.de;tag=as6f1be703
To: sip:Nr@sipgate.de
Call-ID: 5b77d58a4585f32c52dd1d00657b433c@ServerIP
CSeq: 104 REGISTER
User-Agent: Asterisk PBX 1.8.15.0
Authorization: Digest username=“Nr”, realm=“sipgate.de”, algorithm=MD5, uri=“sip:sipgate.de”, nonce=“501f9390e91a7b775e34a39bdf76a205534dad00”, response="a03eedafba2f5e98ca579aa73bd1934e"
Expires: 120
Contact: sip:Nr@ServerIP:5060
Content-Length: 0
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK792dd0d5;rport=5060
From: sip:Nr@sipgate.de;tag=as6f1be703
To: sip:Nr@sipgate.de;tag=fbf1d80521ea9f98078b6998e7669f9b.46e9
Call-ID: 5b77d58a4585f32c52dd1d00657b433c@ServerIP
CSeq: 104 REGISTER
Contact: sip:Nr@ServerIP:5060;expires=120
Content-Length: 0
<------------->
— (8 headers 0 lines) —
Scheduling destruction of SIP dialog ‘5b77d58a4585f32c52dd1d00657b433c@ServerIP’ in 32000 ms (Method: REGISTER)
[2012-08-06 11:42:57] NOTICE[27943]: chan_sip.c:20905 handle_response_register: Outbound Registration: Expiry for sipgate.de is 120 sec (Scheduling reregistration in 105 s)
v2201205126208248CLI> clear
No such command ‘clear’ (type ‘core show help clear’ for other possible commands)
Really destroying SIP dialog ‘1c4358782f6f4ebc0f4a6e700f0200d5@ServerIP:5060’ Method: INVITE
v2201205126208248CLI>
Reliably Transmitting (NAT) to sipgateIP:5060:
OPTIONS sip:sipgate.de SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK34ea94ab;rport
Max-Forwards: 70
From: “asterisk” sip:Nr@ServerIP;tag=as206eec2c
To: sip:sipgate.de
Contact: sip:Nr@ServerIP:5060
Call-ID: 6645e4490327183f56037244615d04d1@ServerIP:5060
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 06 Aug 2012 09:43:12 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK34ea94ab;rport=5060
From: “asterisk” sip:Nr@ServerIP;tag=as206eec2c
To: sip:sipgate.de;tag=64c295986a77a1f756ad49f3e6513d0d.5ae7
Call-ID: 6645e4490327183f56037244615d04d1@ServerIP:5060
CSeq: 102 OPTIONS
Accept: /
Accept-Encoding:
Accept-Language: en
Supported:
Content-Length: 0
<------------->
— (11 headers 0 lines) —
Really destroying SIP dialog ‘6645e4490327183f56037244615d04d1@ServerIP:5060’ Method: OPTIONS
v2201205126208248*CLI>
Really destroying SIP dialog ‘5b77d58a4585f32c52dd1d00657b433c@ServerIP’ Method: REGISTER
<— SIP read from UDP:sipgateIP:5060 —>
INVITE sip:Nr@ServerIP:5060 SIP/2.0
Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8
Record-Route: sip:172.20.40.3;lr=on
Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8
Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.0
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.0
Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK3fc750e5
Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK3fc750e5;rport=5060
Max-Forwards: 67
From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
To: sip:00493069205575@sipgate.de
Contact: sip:anonymous@217.116.117.69
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 387
v=0
o=root 1143069018 1143069018 IN IP4 217.116.117.69
s=sipgate VoIP GW
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 8 0 3 18 112 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:112 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (18 headers 17 lines) —
Sending to sipgateIP:5060 (NAT)
Using INVITE request as basis request - 50b55e844c4028657506bebf100a5a1e@sipgate.de
Found peer ‘ext-sip-account’ for ‘anonymous’ from sipgateIP:5060
== Using SIP RTP CoS mark 5
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 18
Found RTP audio format 112
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format G729 for ID 18
Found audio description format G726-32 for ID 112
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x90e (gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.116.117.69:18490
Looking for Nr in von-voip-provider (domain ServerIP)
list_route: hop: sip:sipgateIP;lr;ftag=as0c7de2d8
list_route: hop: sip:172.20.40.3;lr=on
list_route: hop: sip:sipgateIP;lr;ftag=as0c7de2d8
<— Transmitting (NAT) to sipgateIP:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.0;received=sipgateIP;rport=5060
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.0
Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK3fc750e5
Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK3fc750e5;rport=5060
Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8
Record-Route: sip:172.20.40.3;lr=on
Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8
From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
To: sip:00493069205575@sipgate.de
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:Nr@ServerIP:5060
Content-Length: 0
<------------>
– Executing [Nr@von-voip-provider:1] Answer(“SIP/ext-sip-account-00000002”, “”) in new stack
Audio is at 11222
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.0;received=sipgateIP;rport=5060
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.0
Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK3fc750e5
Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK3fc750e5;rport=5060
Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8
Record-Route: sip:172.20.40.3;lr=on
Record-Route: sip:sipgateIP;lr;ftag=as0c7de2d8
From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
To: sip:00493069205575@sipgate.de;tag=as160de318
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:Nr@ServerIP:5060
Content-Type: application/sdp
Content-Length: 284
v=0
o=root 1630035855 1630035855 IN IP4 ServerIP
s=Asterisk PBX 1.8.15.0
c=IN IP4 ServerIP
t=0 0
m=audio 11222 RTP/AVP 3 0 8 101
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
<------------>
<— SIP read from UDP:sipgateIP:5060 —>
ACK sip:Nr@ServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKd479.81d35c97.2
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKd479.81d35c97.2
Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK781b2e7f
Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK781b2e7f;rport=5060
Max-Forwards: 67
From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
To: sip:00493069205575@sipgate.de;tag=as160de318
Contact: sip:anonymous@217.116.117.69
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 ACK
Content-Length: 0
X-hint: rr-enforced
<------------->
— (13 headers 0 lines) —
– Executing [Nr@von-voip-provider:2] Dial(“SIP/ext-sip-account-00000002”, “SIP/2000&SIP/2001,m”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 19112
Adding codec 0x8 (alaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x800000000000 (testlaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to CallerIP:5060:
INVITE sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK6e0a50d4;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP:5060;ob
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15.0
Date: Mon, 06 Aug 2012 09:43:31 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 282
v=0
o=root 270956748 270956748 IN IP4 ServerIP
s=Asterisk PBX 1.8.15.0
c=IN IP4 ServerIP
t=0 0
m=audio 19112 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
-- Called SIP/2000
Really destroying SIP dialog ‘6a1b82dd713d3cf9179df0fd5c7a7474@ServerIP:5060’ Method: INVITE
[2012-08-06 11:43:31] WARNING[518]: app_dial.c:2341 dial_exec_full: Unable to create channel of type ‘SIP’ (cause 20 - Unknown)
– Started music on hold, class ‘default’, on SIP/ext-sip-account-00000002
[2012-08-06 11:43:31] NOTICE[518]: channel.c:4176 __ast_read: Dropping incompatible voice frame on SIP/ext-sip-account-00000002 of format ulaw since our native format has changed to 0x8 (alaw)
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK6e0a50d4
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK6e0a50d4
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
CSeq: 102 INVITE
Contact: sip:2000@CallerIP:5060;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Content-Length: 0
<------------->
— (9 headers 0 lines) —
list_route: hop: sip:2000@CallerIP:5060;ob
– SIP/2000-00000003 is ringing
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK6e0a50d4
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
CSeq: 102 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Contact: sip:2000@CallerIP:5060;ob
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 3553235010 3553235011 IN IP4 192.168.178.23
s=pjmedia
c=IN IP4 192.168.178.23
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
a=rtcp:4007 IN IP4 192.168.178.23
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.23:4006
list_route: hop: sip:2000@CallerIP:5060;ob
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Transmitting (NAT) to CallerIP:5060:
ACK sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK7210ac68;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP:5060;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
-- SIP/2000-00000003 answered SIP/ext-sip-account-00000002
-- Stopped music on hold on SIP/ext-sip-account-00000002
-- Remotely bridging SIP/ext-sip-account-00000002 and SIP/2000-00000003
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Audio is at 11222
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to sipgateIP:5060:
INVITE sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK6394f884;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 240
v=0
o=root 1630035855 1630035856 IN IP4 192.168.178.23
s=Asterisk PBX 1.8.15.0
c=IN IP4 192.168.178.23
t=0 0
m=audio 4006 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
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Audio is at 19112
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to CallerIP:5060:
INVITE sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK5bc39143;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP:5060;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 239
v=0
o=root 270956748 270956749 IN IP4 217.116.117.69
s=Asterisk PBX 1.8.15.0
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 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
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK6394f884;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK6394f884;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:anonymous@217.116.117.69
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 1143069018 1143069019 IN IP4 217.116.117.69
s=sipgate VoIP GW
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.116.117.69:18490
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Transmitting (NAT) to sipgateIP:5060:
ACK sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK70251ad5;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 102 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK5bc39143
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
CSeq: 103 INVITE
Contact: sip:2000@CallerIP:5060;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 3553235010 3553235012 IN IP4 192.168.178.23
s=pjmedia
c=IN IP4 192.168.178.23
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
a=rtcp:4007 IN IP4 192.168.178.23
a=rtpmap:0 PCMU/8000
a=sendrecv
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.23:4006
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Transmitting (NAT) to CallerIP:5060:
ACK sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK4ba46e24;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP:5060;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
<— SIP read from UDP:CallerIP:5060 —>
INVITE sip:anonymous@ServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPjVSouxbDN2IS8ozVvere.MLNswTmawdo-
Max-Forwards: 70
From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
Contact: sip:2000@CallerIP:5060;ob
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 16255 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple r1108 / GT-I9100-15
Content-Type: application/sdp
Content-Length: 394
v=0
o=- 3553235010 3553235013 IN IP4 192.168.178.23
s=pjmedia
c=IN IP4 192.168.178.23
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 9 107 106 105 0 8 101
a=rtcp:4007 IN IP4 192.168.178.23
a=rtpmap:9 G722/8000
a=rtpmap:107 speex/32000
a=rtpmap:106 speex/16000
a=rtpmap:105 speex/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
<------------->
— (15 headers 17 lines) —
Sending to CallerIP:5060 (NAT)
Found RTP audio format 9
Found RTP audio format 107
Found RTP audio format 106
Found RTP audio format 105
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format G722 for ID 9
Found unknown media description format speex for ID 107
Found audio description format speex for ID 106
Found audio description format speex for ID 105
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x20000120c (ulaw|alaw|speex|speex16|g722)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0xc (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.23:4006
<— Transmitting (NAT) to CallerIP:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP CallerIP:5060;branch=z9hG4bKPjVSouxbDN2IS8ozVvere.MLNswTmawdo-;received=CallerIP;rport=5060
From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 16255 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:anonymous@ServerIP:5060
Content-Length: 0
<------------>
Audio is at 19112
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (NAT) to CallerIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP CallerIP:5060;branch=z9hG4bKPjVSouxbDN2IS8ozVvere.MLNswTmawdo-;received=CallerIP;rport=5060
From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 16255 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: sip:anonymous@ServerIP:5060
Content-Type: application/sdp
Content-Length: 263
v=0
o=root 270956748 270956750 IN IP4 217.116.117.69
s=Asterisk PBX 1.8.15.0
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly
<------------>
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Audio is at 11222
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to sipgateIP:5060:
INVITE sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK500526ab;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 103 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 237
v=0
o=root 1630035855 1630035857 IN IP4 ServerIP
s=Asterisk PBX 1.8.15.0
c=IN IP4 ServerIP
t=0 0
m=audio 11222 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
-- Started music on hold, class 'default', on SIP/ext-sip-account-00000002
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK500526ab;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 103 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK500526ab;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:anonymous@217.116.117.69
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 1143069018 1143069020 IN IP4 217.116.117.69
s=sipgate VoIP GW
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.116.117.69:18490
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Transmitting (NAT) to sipgateIP:5060:
ACK sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK57ef9aa3;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 103 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Audio is at 11222
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to sipgateIP:5060:
INVITE sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK69d13ef5;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 237
v=0
o=root 1630035855 1630035858 IN IP4 ServerIP
s=Asterisk PBX 1.8.15.0
c=IN IP4 ServerIP
t=0 0
m=audio 11222 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
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK69d13ef5;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 104 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK69d13ef5;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 104 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:anonymous@217.116.117.69
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 1143069018 1143069021 IN IP4 217.116.117.69
s=sipgate VoIP GW
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.116.117.69:18490
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Transmitting (NAT) to sipgateIP:5060:
ACK sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK6a4ca94e;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
<— SIP read from UDP:CallerIP:5060 —>
ACK sip:anonymous@ServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPju.FKdbREse-IBM5e98lXY0IRPZb2WfrF
Max-Forwards: 70
From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 16255 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Audio is at 19112
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to CallerIP:5060:
INVITE sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK351d5160;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 104 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 263
v=0
o=root 270956748 270956751 IN IP4 217.116.117.69
s=Asterisk PBX 1.8.15.0
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly
<— SIP read from UDP:CallerIP:5060 —>
INVITE sip:anonymous@ServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPj6rhAyTStDrBvwpuX1g2M-bucKe457oyL
Max-Forwards: 70
From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
Contact: sip:2000@CallerIP:5060;ob
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 16256 INVITE
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 3553235010 3553235014 IN IP4 192.168.178.23
s=pjmedia
c=IN IP4 192.168.178.23
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 8 101
a=rtcp:4007 IN IP4 192.168.178.23
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
<------------->
— (14 headers 12 lines) —
<— Reliably Transmitting (NAT) to CallerIP:5060 —>
SIP/2.0 491 Request Pending
Via: SIP/2.0/UDP CallerIP:5060;branch=z9hG4bKPj6rhAyTStDrBvwpuX1g2M-bucKe457oyL;received=CallerIP;rport=5060
From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 16256 INVITE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
<------------>
Sending to CallerIP:5060 (NAT)
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 491 Another INVITE transaction in progress
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK351d5160
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
CSeq: 104 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Transmitting (NAT) to CallerIP:5060:
ACK sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK351d5160;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 104 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
[2012-08-06 11:43:37] WARNING[27943]: chan_sip.c:20526 handle_response_invite: just did sched_add waitid(51) for sip_reinvite_retry for dialog 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060 in handle_response_invite
<— SIP read from UDP:CallerIP:5060 —>
ACK sip:anonymous@ServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP CallerIP:5060;rport;branch=z9hG4bKPj6rhAyTStDrBvwpuX1g2M-bucKe457oyL
Max-Forwards: 70
From: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
To: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 16256 ACK
Content-Length: 0
<------------->
— (8 headers 0 lines) —
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Audio is at 19112
Adding codec 0x8 (alaw) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to CallerIP:5060:
INVITE sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK1a367da5;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 263
v=0
o=root 270956748 270956752 IN IP4 217.116.117.69
s=Asterisk PBX 1.8.15.0
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK1a367da5
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
CSeq: 105 INVITE
Contact: sip:2000@CallerIP:5060;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 3553235010 3553235014 IN IP4 192.168.178.23
s=pjmedia
c=IN IP4 192.168.178.23
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
a=rtcp:4007 IN IP4 192.168.178.23
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.23:4006
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Transmitting (NAT) to CallerIP:5060:
ACK sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK216abeba;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
-- Stopped music on hold on SIP/ext-sip-account-00000002
-- Started music on hold, class 'default', on SIP/ext-sip-account-00000002
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Audio is at 11222
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to sipgateIP:5060:
INVITE sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK374031f1;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 105 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 237
v=0
o=root 1630035855 1630035859 IN IP4 ServerIP
s=Asterisk PBX 1.8.15.0
c=IN IP4 ServerIP
t=0 0
m=audio 11222 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
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK374031f1;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 105 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
<— SIP read from UDP:sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;received=ServerIP;branch=z9hG4bK374031f1;rport=5060
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 105 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
Contact: sip:anonymous@217.116.117.69
Content-Type: application/sdp
Content-Length: 262
v=0
o=root 1143069018 1143069022 IN IP4 217.116.117.69
s=sipgate VoIP GW
c=IN IP4 217.116.117.69
t=0 0
m=audio 18490 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 217.116.117.69:18490
set_destination: Parsing sip:sipgateIP;lr;ftag=as0c7de2d8 for address/port to send to
set_destination: set destination to sipgateIP:5060
Transmitting (NAT) to sipgateIP:5060:
ACK sip:anonymous@217.116.117.69 SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK56ba5450;rport
Route: sip:sipgateIP;lr;ftag=as0c7de2d8,sip:172.20.40.3;lr=on,sip:sipgateIP;lr;ftag=as0c7de2d8
Max-Forwards: 70
From: sip:00493069205575@sipgate.de;tag=as160de318
To: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
Contact: sip:Nr@ServerIP:5060
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 105 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
<— SIP read from UDP:sipgateIP:5060 —>
BYE sip:Nr@ServerIP:5060 SIP/2.0
Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKe479.a7d3cb64.0
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKe479.a7d3cb64.0
Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK0cc06cb6
Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK0cc06cb6;rport=5060
Max-Forwards: 67
From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
To: sip:00493069205575@sipgate.de;tag=as160de318
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 103 BYE
Content-Length: 0
X-hint: rr-enforced
<------------->
— (12 headers 0 lines) —
Sending to sipgateIP:5060 (NAT)
Scheduling destruction of SIP dialog ‘50b55e844c4028657506bebf100a5a1e@sipgate.de’ in 6400 ms (Method: BYE)
<— Transmitting (NAT) to sipgateIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP sipgateIP:5060;branch=z9hG4bKe479.a7d3cb64.0;received=sipgateIP;rport=5060
Via: SIP/2.0/UDP 172.20.40.3;branch=z9hG4bKe479.a7d3cb64.0
Via: SIP/2.0/UDP sipgateIP:5060;received=217.10.68.222;branch=z9hG4bK0cc06cb6
Via: SIP/2.0/UDP 217.116.117.69:5060;received=217.116.117.69;branch=z9hG4bK0cc06cb6;rport=5060
From: “anonymous” sip:anonymous@sipgate.de;tag=as0c7de2d8
To: sip:00493069205575@sipgate.de;tag=as160de318
Call-ID: 50b55e844c4028657506bebf100a5a1e@sipgate.de
CSeq: 103 BYE
Server: Asterisk PBX 1.8.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0
<------------>
– Stopped music on hold on SIP/ext-sip-account-00000002
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Audio is at 19112
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to CallerIP:5060:
INVITE sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK5c5f7955;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 106 INVITE
User-Agent: Asterisk PBX 1.8.15.0
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: 235
v=0
o=root 270956748 270956753 IN IP4 ServerIP
s=Asterisk PBX 1.8.15.0
c=IN IP4 ServerIP
t=0 0
m=audio 19112 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=recvonly
Scheduling destruction of SIP dialog ‘222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060’ in 32000 ms (Method: ACK)
== Spawn extension (von-voip-provider, Nr, 2) exited non-zero on ‘SIP/ext-sip-account-00000002’
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK5c5f7955
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
CSeq: 106 INVITE
Contact: sip:2000@CallerIP:5060;ob
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Content-Type: application/sdp
Content-Length: 257
v=0
o=- 3553235010 3553235015 IN IP4 192.168.178.23
s=pjmedia
c=IN IP4 192.168.178.23
t=0 0
a=X-nat:0
m=audio 4006 RTP/AVP 0 101
a=rtcp:4007 IN IP4 192.168.178.23
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendonly
<------------->
— (11 headers 12 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 - 0x80000008000e (gsm|ulaw|alaw|h263|testlaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing)/text=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.178.23:4006
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Transmitting (NAT) to CallerIP:5060:
ACK sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK060ea50b;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Contact: sip:anonymous@ServerIP:5060
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 106 ACK
User-Agent: Asterisk PBX 1.8.15.0
Content-Length: 0
set_destination: Parsing sip:2000@CallerIP:5060;ob for address/port to send to
set_destination: set destination to CallerIP:5060
Reliably Transmitting (NAT) to CallerIP:5060:
BYE sip:2000@CallerIP:5060;ob SIP/2.0
Via: SIP/2.0/UDP ServerIP:5060;branch=z9hG4bK0bcd7924;rport
Max-Forwards: 70
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
CSeq: 107 BYE
User-Agent: Asterisk PBX 1.8.15.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
Scheduling destruction of SIP dialog ‘222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060’ in 32000 ms (Method: ACK)
<— SIP read from UDP:CallerIP:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP ServerIP:5060;rport=5060;received=ServerIP;branch=z9hG4bK0bcd7924
Call-ID: 222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060
From: “anonymous” sip:anonymous@ServerIP;tag=as6d379f75
To: sip:2000@CallerIP;ob;tag=59gwgFTPkFilTtVKproVybJ4vOBwWOic
CSeq: 107 BYE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
Really destroying SIP dialog ‘222bee2e793d4ffd02fdf3592e24069e@ServerIP:5060’ Method: ACK
Really destroying SIP dialog ‘50b55e844c4028657506bebf100a5a1e@sipgate.de’ Method: BYE
<— SIP read from UDP:CallerIP:5060 —>
<------------->
[/code]