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Topics - artur.baghdasaryan

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Other Yate server issues / Yate Redirect SIP 300 Multiple Choices
« on: September 15, 2017, 02:24:02 AM »
Dear Yate Team,

Please help to figure out why Yate cant redirect calls to destination which is in Contact header of SIP 300 Multiple Choices message.
I have enabled the pbxassist module and set diversion to yes.
Please could you advise where is the problem
Trace is below
20170915081051.612263 <sip:INFO> 'udp:0.0.0.0:5060' received 869 bytes SIP message from 1.1.1.1:5060 [0x1cfcef0]
------
INVITE sip:11111111@2.2.2.2;user=phone SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e
Max-Forwards: 70
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>
Contact: <sip:+2222222222@1.1.1.1:5060>
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
User-Agent: Test
Date: Fri, 15 Sep 2017 08:10:51 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 497095320 497095320 IN IP4 1.1.1.1
s=Test
c=IN IP4 1.1.1.1
t=0 0
m=audio 14266 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=sendrecv
------
20170915081051.621309 <sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x7f8ef0002b90 to 1.1.1.1:5060 [0x1cfcef0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e;received=1.1.1.1
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
Server: Test
Content-Length: 0

------
20170915081051.621410 <sip/23:ALL> YateSIPConnection::YateSIPConnection(0x7f8ef0010ad0,0x7f8ef402c1e0) [0x7f8ef0012020]
20170915081051.621550 <sip/23:ALL> NAT address is '(null)' [0x7f8ef0012020]
20170915081051.621793 <sip/23:ALL> Set media: audio=alaw,mulaw [0x7f8ef0012020]
20170915081051.622182 <INFO> Could not classify call from '+2222222222', wasted 214 usec
20170915081051.622255 <cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/23'
20170915081051.622328 <RegexRoute:ALL> Jumping to context 'from_internal' by rule #1 '${module}sip'
20170915081051.622664 <RegexRoute:ALL> Jumping to context 'aster_test' by rule #9 '${address}^1\.1\.1\.1:'
20170915081051.622723 <pbxassist:CALL> Created guest assistant for 'sip/23'
20170915081051.622855 <INFO> Routing call to '11111111' in context 'default' via 'sip/sip:2222222222@3.3.3.3' in 558 usec
20170915081051.622903 <sip/23:ALL> NAT address is '(null)' [0x7f8ef0012020]
20170915081051.623092 <sip/24:ALL> YateSIPConnection::YateSIPConnection(0x7f8ef00036d0,'sip:2222222222@3.3.3.3') [0x7f8edc00ee80]
20170915081051.623171 <sip:ALL> YateSIPEndPoint::buildParty(0x7f8edc014400,'(null)',0,(nil))
20170915081051.623239 <sip/24:ALL> NAT address is '(null)' [0x7f8edc00ee80]
20170915081051.623381 <sip/24:ALL> Set media: audio=alaw,mulaw,g729 [0x7f8edc00ee80]
20170915081051.627078 <sip:INFO> 'udp:0.0.0.0:5060' sending code 183 0x7f8edc010830 to 1.1.1.1:5060 [0x1cfcef0]
------
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e;received=1.1.1.1
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>;tag=1459504950
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
Server: Test
Contact: <sip:11111111@2.2.2.2:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
20170915081051.627200 <sip:INFO> 'udp:0.0.0.0:5060' sending 'INVITE sip:2222222222@3.3.3.3' 0x7f8edc014400 to 3.3.3.3:5060 [0x1cfcef0]
------
INVITE sip:2222222222@3.3.3.3 SIP/2.0
Max-Forwards: 69
Contact: <sip:2222222222@1.1.1.1:5060>
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 INVITE
User-Agent: Test
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 256

v=0
o=yate 1505463051 1505463051 IN IP4 2.2.2.2
s=SIP Call
c=IN IP4 2.2.2.2
t=0 0
m=audio 26938 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
------
20170915081051.628027 <pbxassist:CALL> Created assistant for 'sip/24'
20170915081051.629268 <sip:INFO> 'udp:0.0.0.0:5060' received 278 bytes SIP message from 3.3.3.3:5060 [0x1cfcef0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 INVITE
Server: Invoice SIP-server
Content-Length: 0

------
20170915081051.639711 <sip:INFO> 'udp:0.0.0.0:5060' received 344 bytes SIP message from 3.3.3.3:5060 [0x1cfcef0]
------
SIP/2.0 300 Multiple choices
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 INVITE
Contact: <sip:2222222222@188.115.195.177:5060>
                                               TG:  0
Server: Invoice SIP-server
Content-Length: 0

------
20170915081051.642651 <sip/24:ALL> YateSIPConnection::hangup() state=1 trans=0x7f8edc015c40 error='(null)' code=300 reason='Multiple choices' [0x7f8edc00ee80]
20170915081051.642724 <ALL> Cleaning up RTP 0x7f8edc00a950 [0x7f8edc00a7a0]
20170915081051.642781 <sip/23:ALL> YateSIPConnection::disconnected() 'Multiple choices' [0x7f8ef0012020]
20170915081051.642817 <sip/24:ALL> YateSIPConnection::~YateSIPConnection() [0x7f8edc00ee80]
20170915081051.642884 <sip:INFO> 'udp:0.0.0.0:5060' sending 'ACK sip:2222222222@3.3.3.3' 0x7f8ef4028160 to 3.3.3.3:5060 [0x1cfcef0]
------
ACK sip:2222222222@3.3.3.3 SIP/2.0
Via: SIP/2.0/UDP 2.2.2.2:5060;rport;branch=z9hG4bK962542606
From: <sip:2222222222@2.2.2.2>;tag=306964556
To: <sip:2222222222@3.3.3.3>
Call-ID: 564312445@2.2.2.2
CSeq: 44 ACK
Max-Forwards: 69
Contact: <sip:2222222222@1.1.1.1:5060>
User-Agent: Test
Content-Length: 0

------
20170915081051.643742 <INFO> Could not classify call from '2222222222', wasted 104 usec
20170915081051.643830 <INFO> Could not route call to '11111111' in context 'default', wasted 22 usec
20170915081051.643857 <pbxassist:MILD> Chan 'sip/23' operation failed: no route
20170915081051.643884 <sip/23:ALL> YateSIPConnection::hangup() state=0 trans=0x7f8ef402c1e0 error='noanswer' code=487 reason='Multiple choices' [0x7f8ef0012020]
20170915081051.643966 <sip/23:ALL> YateSIPConnection::~YateSIPConnection() [0x7f8ef0012020]
20170915081051.648052 <sip:INFO> 'udp:0.0.0.0:5060' sending code 487 0x7f8eec0175a0 to 1.1.1.1:5060 [0x1cfcef0]
------
SIP/2.0 487 Multiple choices
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e;received=1.1.1.1
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>;tag=1459504950
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 INVITE
Server: Test
Contact: <sip:11111111@2.2.2.2:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
20170915081051.648564 <sip:INFO> 'udp:0.0.0.0:5060' received 418 bytes SIP message from 1.1.1.1:5060 [0x1cfcef0]
------
ACK sip:11111111@2.2.2.2:5060 SIP/2.0
Via: SIP/2.0/UDP 1.1.1.1:5060;branch=z9hG4bK6a19fc4e
Max-Forwards: 70
From: "+2222222222" <sip:+2222222222@1.1.1.1>;tag=as6e0e860a
To: <sip:11111111@2.2.2.2;user=phone>;tag=1459504950
Contact: <sip:+2222222222@1.1.1.1:5060>
Call-ID: 40d6a8f200621e456cbf8c1431df234a@1.1.1.1:5060
CSeq: 102 ACK
User-Agent: Test
Content-Length: 0

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