Hello.
Please help with following problem - I'm trying to make a switch to another
phone via IVR, but RTP session establishing only in one way.
I have two phones
"Bob" with numer 3000, IP is x.x.x.87 and unnamed numer 2000, IP is x.x.x.239
and SEMS 1.2.1 at x.x.x.60:5080 and openSIPS 1.6.2 at same IP but port 5060.
I'm calling to "Alice" 1000 - it is redirect to IVR by openSIPS:
Jun 28 14:53:03 pbx sems[19408]: DEBUG: [b6f7cb90] run (udp_trsp.cpp:140):
recvd msg --++-- INVITE sip:[email protected]:5080 SIP/2.0
Record-Route: <sip:x.x.x.60;lr=on;ftag=3c3e4344b6c057a6o0>
Via: SIP/2.0/UDP x.x.x.60;branch=z9hG4bKcd56.0f35d6f5.0
Via: SIP/2.0/UDP x.x.x.87:5060;branch=z9hG4bK-e9d7f2a2
From: "Bob" <sip:[email protected]>;tag=3c3e4344b6c057a6o0
To: "Alice" <sip:[email protected]>
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 32
Contact: "Bob" <sip:[email protected]:5060>
Expires: 240
User-Agent: Linksys/SPA922-6.1.5(a)
Content-Length: 397
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
P-App-Name: ivr
v=0
o=- 536475 536475 IN IP4 x.x.x.87
s=-
c=IN IP4 x.x.x.87
t=0 0
m=audio 16470 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96
Jun 28 14:53:03 pbx sems[19408]: DEBUG: [b6d2fb90] send (udp_trsp.cpp:244):
send msg --++-- SIP/2.0 200 OK
Record-Route: <sip:x.x.x.60;lr=on;ftag=3c3e4344b6c057a6o0>
Via: SIP/2.0/UDP x.x.x.60;branch=z9hG4bKcd56.0f35d6f5.0
Via: SIP/2.0/UDP x.x.x.87:5060;branch=z9hG4bK-e9d7f2a2
From: "Bob" <sip:[email protected]>;tag=3c3e4344b6c057a6o0
To: "Alice" <sip:[email protected]>;tag=12461D48-4C287F0F000D74FC-B6F7CB90
Call-ID: [email protected]
CSeq: 102 INVITE
Server: Sip Express Media Server (1.2.1 (i386/linux))
Contact: <sip:[email protected]:5080>
Content-Type: application/sdp
Content-Length: 222
v=0
o=sems 1371675074 324093185 IN IP4 x.x.x.60
s=session
c=IN IP4 x.x.x.60
t=0 0
m=audio 10006 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
--++--
Jun 28 14:53:04 pbx sems[19408]: DEBUG: [b6f7cb90] run (udp_trsp.cpp:140):
recvd msg --++-- ACK sip:[email protected]:5080 SIP/2.0
Via: SIP/2.0/UDP x.x.x.60;branch=z9hG4bKcd56.0f35d6f5.2
Via: SIP/2.0/UDP x.x.x.87:5060;branch=z9hG4bK-63dc400f
From: "Bob" <sip:[email protected]>;tag=3c3e4344b6c057a6o0
To: "Alice" <sip:[email protected]>;tag=12461D48-4C287F0F000D74FC-B6F7CB90
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 32
Proxy-Authorization: Digest
username="3000",realm="domain.com",nonce="4c287f2d22c44ed5d7a50c2df46af460427b3b21",uri="sip:[email protected]",algorithm=MD5,response="b5a5bbd5ca98b376c553ad16804d718f"
Contact: "Bob" <sip:[email protected]:5060>
User-Agent: Linksys/SPA922-6.1.5(a)
Content-Length: 0
--++--
At this time executed follwing IVR script (t.py):
t.py
from log import *
from ivr import *
class IvrDialog(IvrDialogBase):
announcement = None
def onSessionStart(self, hdrs):
debug("IVR START")
self.announcement = IvrAudioFile()
self.announcement.open("/usr/local/lib/sems/ivr/ivr_ann.wav",
ivr.AUDIO_READ)
self.enqueue(self.announcement, None)
def onEmptyQueue(self):
B2BMode = True
self.connectCallee('<sip:[email protected]>', 'sip:[email protected]')
(I want to make a conversation between Bob 3000 and 2000)
Jun 28 14:53:19 pbx sems[19408]: DEBUG: [b6e55b90] send (udp_trsp.cpp:244):
send msg --++-- INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bK.9Q63ah2
From: "Bob" <sip:[email protected]>;tag=2C3D7CD9-4C287F1F000506D5-B6D2FB90
To: <sip:[email protected]>
CSeq: 10 INVITE
Call-ID: [email protected]
Contact: <sip:[email protected]:5080> // <-- it is SEMS account on openSIPS
Max-Forwards: 32
Expires: 240
User-Agent: Linksys/SPA922-6.1.5(a)
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
P-App-Name: ivr
Content-Type: application/sdp
Content-Length: 397
v=0
o=- 536475 536475 IN IP4 x.x.x.87
s=-
c=IN IP4 x.x.x.87
t=0 0
m=audio 16470 RTP/AVP 0 2 4 8 18 96 97 98 101
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:18 G729a/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-1
Jun 28 14:53:19 pbx sems[19408]: DEBUG: [b6f7cb90] run (udp_trsp.cpp:140):
recvd msg --++-- SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bK.9Q63ah2
From: "Bob" <sip:[email protected]>;tag=2C3D7CD9-4C287F1F000506D5-B6D2FB90
To: <sip:[email protected]>
CSeq: 10 INVITE
Call-ID: [email protected]
Server: OpenSIPS (1.6.2-notls (i386/linux))
Content-Length: 0
Warning: 392 x.x.x.60:5060 "Noisy feedback tells: pid=19308
req_src_ip=x.x.x.60 req_src_port=5080 in_uri=sip:[email protected]
out_uri=sip:[email protected]:5063 via_cnt==1"
--++--
Jun 28 14:53:19 pbx sems[19408]: DEBUG: [b6f7cb90] run (udp_trsp.cpp:140):
recvd msg --++-- SIP/2.0 180 Ringing
To: <sip:[email protected]>;tag=b12ac6158e8bdf9di3
From: "Bob" <sip:[email protected]>;tag=2C3D7CD9-4C287F1F000506D5-B6D2FB90
Call-ID: [email protected]
CSeq: 10 INVITE
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bK.9Q63ah2
Record-Route: <sip:x.x.x.60;lr=on;ftag=2C3D7CD9-4C287F1F000506D5-B6D2FB90>
Server: Linksys/SPA942-5.2.8
Content-Length: 0
--++--
Jun 28 14:53:20 pbx sems[19408]: DEBUG: [b6f7cb90] run (udp_trsp.cpp:140):
recvd msg --++-- SIP/2.0 200 OK
To: <sip:[email protected]>;tag=b12ac6158e8bdf9di3
From: "Bob" <sip:[email protected]>;tag=2C3D7CD9-4C287F1F000506D5-B6D2FB90
Call-ID: [email protected]
CSeq: 10 INVITE
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bK.9Q63ah2
Record-Route: <sip:x.x.x.60;lr=on;ftag=2C3D7CD9-4C287F1F000506D5-B6D2FB90>
Contact: "Hell" <sip:[email protected]:5063>
Server: Linksys/SPA942-5.2.8
Content-Length: 214
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
Content-Type: application/sdp
v=0
o=- 93275183 93275183 IN IP4 x.x.x.239
s=-
c=IN IP4 x.x.x.239
t=0 0
m=audio 16428 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--++--
Jun 28 14:53:20 pbx sems[19408]: DEBUG: [b6e55b90] send (udp_trsp.cpp:244):
send msg --++-- ACK sip:[email protected]:5063 SIP/2.0
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bKDxop9aO7
Route: <sip:x.x.x.60;lr=on;ftag=2C3D7CD9-4C287F1F000506D5-B6D2FB90>
From: "Bob" <sip:[email protected]>;tag=2C3D7CD9-4C287F1F000506D5-B6D2FB90
To: <sip:[email protected]>;tag=b12ac6158e8bdf9di3
CSeq: 10 ACK
Call-ID: [email protected]
Contact: <sip:[email protected]:5080>
User-Agent: Sip Express Media Server (1.2.1 (i386/linux))
Max-Forwards: 70
Content-Length: 0
--++--
// re-INVITE
Jun 28 14:53:20 pbx sems[19408]: DEBUG: [b6d2fb90] send (udp_trsp.cpp:244):
send msg --++-- INVITE sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bKADd.qapI
Route: <sip:x.x.x.60;lr=on;ftag=3c3e4344b6c057a6o0>
From: "Alice" <sip:[email protected]>;tag=12461D48-4C287F0F000D74FC-B6F7CB90
To: "Bob" <sip:[email protected]>;tag=3c3e4344b6c057a6o0
CSeq: 10 INVITE
Call-ID: [email protected]
Contact: <sip:[email protected]:5080>
Content-Type: application/sdp
Content-Length: 214
v=0
o=- 93275183 93275183 IN IP4 x.x.x.239
s=-
c=IN IP4 x.x.x.239
t=0 0
m=audio 16428 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--++--
Jun 28 14:53:20 pbx sems[19408]: DEBUG: [b6f7cb90] run (udp_trsp.cpp:140):
recvd msg --++-- SIP/2.0 100 Giving a try
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bKADd.qapI
From: "Alice" <sip:[email protected]>;tag=12461D48-4C287F0F000D74FC-B6F7CB90
To: "Bob" <sip:[email protected]>;tag=3c3e4344b6c057a6o0
CSeq: 10 INVITE
Call-ID: [email protected]
Server: OpenSIPS (1.6.2-notls (i386/linux))
Content-Length: 0
Warning: 392 x.x.x.60:5060 "Noisy feedback tells: pid=19304
req_src_ip=x.x.x.60 req_src_port=5080 in_uri=sip:[email protected]:5060
out_uri=sip:[email protected]:5060 via_cnt==1"
--++--
Jun 28 14:53:20 pbx sems[19408]: DEBUG: [b6f7cb90] run (udp_trsp.cpp:140):
recvd msg --++-- SIP/2.0 200 OK
To: "Bob" <sip:[email protected]>;tag=3c3e4344b6c057a6o0
From: "Alice" <sip:[email protected]>;tag=12461D48-4C287F0F000D74FC-B6F7CB90
Call-ID: [email protected]
CSeq: 10 INVITE
Via: SIP/2.0/UDP domain.com.60:5080;branch=z9hG4bKADd.qapI
Record-Route: <sip:domain.com.60;lr=on;ftag=12461D48-4C287F0F000D74FC-B6F7CB90>
Contact: "Bob" <sip:[email protected]:5060>
Server: Linksys/SPA922-6.1.5(a)
Content-Length: 208
Content-Type: application/sdp
v=0
o=- 536475 536476 IN IP4 x.x.x.87
s=-
c=IN IP4 x.x.x.87
t=0 0
m=audio 16470 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv
--++--
Jun 28 14:53:20 pbx sems[19408]: DEBUG: [b6d2fb90] send (udp_trsp.cpp:244):
send msg --++-- ACK sip:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP x.x.x.60:5080;branch=z9hG4bK7QfqFaPw
Route: <sip:x.x.x.60;lr=on;ftag=3c3e4344b6c057a6o0>
From: "Alice" <sip:[email protected]>;tag=12461D48-4C287F0F000D74FC-B6F7CB90
To: "Bob" <sip:[email protected]>;tag=3c3e4344b6c057a6o0
CSeq: 10 ACK
Call-ID: [email protected]
Contact: <sip:[email protected]:5080>
User-Agent: Sip Express Media Server (1.2.1 (i386/linux))
Max-Forwards: 70
Content-Length: 0
--++--
And after this i hear Bob 3000 in 2000, but don't hear 2000 in Bob 3000.
I've looked on x.x.x.60 traffic via Wireshark and noticed that after when 2000
answered the call there is a RTP traffic FROM x.x.x.60(sems) TO x.x.x.87( Bob
3000 ) - is this normal?
Thank you!_______________________________________________
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