[ previous ] [ next ] [ threads ]
 To :  Xiaokui Qin <xkqin@y...>
 From :  Bill Simon <bill@b...>
 Subject :  Re: [yate] no ring when calling out using googlevoice
 Date :  Wed, 28 Dec 2011 22:39:58 -0500 (EST)
I think you need this solution from last month's discussion. 180 Ringing without early media: http://yate.null.ro/archive/?action=show_msg&actionargs[]=67&actionargs[]=67 

----- Original Message -----

> From: "Xiaokui Qin" 
> To: yate@v...
> Sent: Saturday, December 24, 2011 3:06:25 PM
> Subject: Re: [yate] no ring when calling out using googlevoice

> Hi Yate programmer,

> Here are some more details.

> 1. The yate was compiled from svn on ubuntu 11.10.
> 2. The client is linphone.
> 3. Both sever and client are running on 192.168.1.10 behind a
> firewall.

> Here are the logs from Yate:

> ------
>  'udp:0.0.0.0:5060' received 887 bytes SIP message from
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> INVITE sip:6001234@1... SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1425508363
> From: ;tag=653474614
> To: 
> Call-ID: 615291946
> CSeq: 20 INVITE
> Contact: 
> Content-Type: application/sdp
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Max-Forwards: 70
> User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
> Subject: Phone call
> Content-Length: 404

> v=0
> o=asus 123456 654321 IN IP4 192.168.1.10
> s=A conversation
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
> a=rtpmap:112 speex/32000/1
> a=fmtp:112 vbr=on
> a=rtpmap:111 speex/16000/1
> a=fmtp:111 vbr=on
> a=rtpmap:110 speex/8000/1
> a=fmtp:110 vbr=on
> a=rtpmap:3 GSM/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000/1
> a=fmtp:101 0-11
> ------
>  'udp:0.0.0.0:5060' sending code 100 0x8d7edc0 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1425508363;received=192.168.1.10
> From: ;tag=653474614
> To: 
> Call-ID: 615291946
> CSeq: 20 INVITE
> Server: YATE/3.3.3
> Content-Length: 0

> ------
>  YateSIPConnection::YateSIPConnection(0x8d7f1e8,0x8d7ed40)
> [0x8d7f7a8]
>  Could not classify call from 'asus', wasted 59 usec
>  Got message 'call.route' for untracked id 'sip/1'
>  Routing call to '6001234' in context 'default' via '-' in 69
> usec
>  Call rejected error='noauth' reason='(null)' [0x8d7f7a8]
>  YateSIPConnection::hangup() state=0 trans=0x8d7ed40
> error='noauth' code=401 reason='(null)' [0x8d7f7a8]
>  YateSIPConnection::~YateSIPConnection() [0x8d7f7a8]
>  'udp:0.0.0.0:5060' sending code 401 0x8d81ff8 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1425508363;received=192.168.1.10
> From: ;tag=653474614
> To: 
> Call-ID: 615291946
> CSeq: 20 INVITE
> WWW-Authenticate: Digest realm="Yate",
> nonce="1f90cff54a0caa5b677bd1ce86b6ebcf.1324756677", stale=FALSE,
> algorithm=MD5
> Server: YATE/3.3.3
> Contact: 
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' received 237 bytes SIP message from
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> ACK sip:6001234@1... SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1425508363
> From: ;tag=653474614
> To: 
> Call-ID: 615291946
> CSeq: 20 ACK
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' received 1085 bytes SIP message from
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> INVITE sip:6001234@1... SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1058117490
> From: ;tag=653474614
> To: 
> Call-ID: 615291946
> CSeq: 21 INVITE
> Contact: 
> Authorization: Digest username="asus", realm="Yate",
> nonce="1f90cff54a0caa5b677bd1ce86b6ebcf.1324756677",
> uri="sip:6001234@1...",
> response="77f602664987b0ee47590890cd3dab58", algorithm=MD5
> Content-Type: application/sdp
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
> SUBSCRIBE, INFO
> Max-Forwards: 70
> User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
> Subject: Phone call
> Content-Length: 404

> v=0
> o=asus 123456 654321 IN IP4 192.168.1.10
> s=A conversation
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
> a=rtpmap:112 speex/32000/1
> a=fmtp:112 vbr=on
> a=rtpmap:111 speex/16000/1
> a=fmtp:111 vbr=on
> a=rtpmap:110 speex/8000/1
> a=fmtp:110 vbr=on
> a=rtpmap:3 GSM/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:101 telephone-event/8000/1
> a=fmtp:101 0-11
> ------
>  'udp:0.0.0.0:5060' sending code 100 0x8d7c540 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
> From: ;tag=653474614
> To: 
> Call-ID: 615291946
> CSeq: 21 INVITE
> Server: YATE/3.3.3
> Content-Length: 0

> ------
>  YateSIPConnection::YateSIPConnection(0x8d80e30,0x8d812c0)
> [0x8d81730]
>  Authenticating user asus with password length 8
>  Could not classify call from 'asus', wasted 28 usec
>  Got message 'call.route' for untracked id 'sip/2'
>  Routing call to '6001234' in context 'default' via
> 'jingle/2036001234@v...' in 105 usec
>  msgExecute. caller='wjgoogle@g.../D54DEA90'
> called='2036001234@v...' online=true filetransfer=false
>  Jingle version set to 0 from routing
>  Session flags set to 1 from ojingle_flags=noping
> [0x8d85888]
>  Outgoing. caller='wjgoogle@g.../D54DEA90'
> called='2036001234@v...' [0x8d85888]
>  Calling. caller=wjgoogle@g.../D54DEA90
> called=2036001234@v... [0x8d85888]
>  Added content='jingle/1_content_86247556' type=ice-udp
> initiator=true [0x8d85888]
>  Call(JG1_2036649621). Outgoing
> from=wjgoogle@g.../D54DEA90 to=2036001234@v...
> [0x8d886b0]
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v..." id="JG1_2036649621_1">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v..." id="JG1_2036649621">
> 
> 
> 
> 
>  clockrate="8000"/>
> 
> 
> 
> -----
>  Call(JG1_2036649621). Changing state from Idle to
> Pending [0x8d886b0]
>  Using audio content
> 'jingle/1_content_86247556' [0x8d85888]
>  No-transport message received
> 
> YRTPWrapper::YRTPWrapper('192.168.1.10',0x8d85888,'audio',bidir,0xb703b76c,false)
> [0x8d898c8]
>  YRTPWrapper::setupRTP("192.168.1.10",true) [0x8d898c8]
>  Session 'yrtp/1549148331' 0x8d88db0 bound to
> 192.168.1.10:21808 +RTCP [0x8d898c8]
>  YRTPSource::YRTPSource(0x8d898c8) [0x8d89af8]
>  YRTPConsumer::YRTPConsumer(0x8d898c8) [0x8d89be8]
>  YRTPWrapper::setupSRTP(false) [0x8d898c8]
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v..." id="JG1_2036649621_2">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v..." id="JG1_2036649621">
> 
>  port="21808" network="0" protocol="udp" username="1894068495178298"
> password="***" type="local" preference="1"/>
> 
> 
> 
> -----
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v..." id="JG1_2036649621_3">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v..." id="JG1_2036649621">
>  port="21808" network="0" protocol="udp" username="1894068495178298"
> password="***" type="local" preference="1"/>
> 
> 
> -----
>  Failed to start RTP for
> content='jingle/1_content_86247556' candidates local=true
> remote=false
> [0x8d85888]
> ringing targetid is sip/2 peerid is id is jingle/1 callto is
>  RTP/AVP message received
>  Guessed local IP '192.168.1.10' for remote '192.168.1.10'
> 
> YRTPWrapper::YRTPWrapper('192.168.1.10',0x8d81730,'audio',bidir,0x8d82280,false)
> [0x8d8a518]
>  YRTPWrapper::setupRTP("192.168.1.10",true) [0x8d8a518]
>  Session 'yrtp/1432917202' 0x8d7ed50 bound to
> 192.168.1.10:28240 +RTCP [0x8d8a518]
>  YRTPSource::YRTPSource(0x8d8a518) [0x8d81c90]
>  DataTranslator::attachChain [0x8d81c90] '(null)' ->
> [0x8d89be8]
> '(null)' not possible
>  YRTPConsumer::YRTPConsumer(0x8d8a518) [0x8d8a760]
>  DataTranslator::attachChain [0x8d89af8] '(null)' ->
> [0x8d8a760]
> '(null)' not possible
>  YRTPWrapper::startRTP("192.168.1.10",7078) [0x8d8a518]
>  RTP starting format 'speex/32000' payload 112 [0x8d8a518]
> >>> DataTranslator::detachChain(0x8d81c90,0x8d89be8)
> <<< DataTranslator::detachChain
>  DataTranslator::attachChain [0x8d81c90] 'speex/32000' ->
> [0x8d89be8] '(null)' not possible
> >>> DataTranslator::detachChain(0x8d89af8,0x8d8a760)
> <<< DataTranslator::detachChain
>  DataTranslator::attachChain [0x8d89af8] '(null)' ->
> [0x8d8a760]
> 'speex/32000' not possible
>  Choosing started 'audio' format 'speex/32000' [0x8d83db8]
> connected targetid is peerid is jingle/1 id is sip/2 callto is
> connected targetid is sip/2 peerid is sip/2 id is jingle/1 callto is
>  'udp:0.0.0.0:5060' sending code 180 0x8d8a2a0 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
> From: ;tag=653474614
> To: ;tag=1676262022
> Call-ID: 615291946
> CSeq: 21 INVITE
> Server: YATE/3.3.3
> Contact: 
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
> Content-Type: application/sdp
> Content-Length: 189

> v=0
> o=yate 1324756677 1324756677 IN IP4 192.168.1.10
> s=SIP Call
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 28240 RTP/AVP 112 101
> a=rtpmap:112 SPEEX/32000
> a=rtpmap:101 telephone-event/8000
> ------
>  Receiving from 'gmail.com' [0x8d59b70]
> -----
>  to="wjgoogle@g.../D54DEA90" id="JG1_2036649621_1">
>  responder="2036001234@v..." id="JG1_2036649621"
> xmlns="http://www.google.com/session">
> 
> 
> 
> 
>  clockrate="8000"/>
> 
> 
> 
>  xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=
>  xmlns:ses="http://www.google.com/session">xmpp:2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=
> 
> 
> -----
>  Processing event (0x8d89398,Iq)
>  Call(JG1_2036649621). Sent element with
> id=JG1_2036649621_1 confirmed by error. Terminating [0x8d886b0]
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v..." id="JG1_2036649621_4">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v..." id="JG1_2036649621"/>
> 
> -----
>  Call(JG1_2036649621). Changing state from Pending to
> Ending [0x8d886b0]
>  Call(JG1_2036649621). Changing state from Ending to
> Destroy [0x8d886b0]
>  Redirecting to
> '2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI='
> [0x8d85888]
>  msgExecute. caller='wjgoogle@g.../D54DEA90'
> called='2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI='
> online=true filetransfer=false
>  Jingle version set to 0 from routing
>  Session flags set to 1 from ojingle_flags=noping
> [0x8d8af70]
>  Outgoing. caller='wjgoogle@g.../D54DEA90'
> called='2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI='.
> Transferred from=2036001234@v... [0x8d8af70]
>  Calling. caller=wjgoogle@g.../D54DEA90
> called=2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=
> [0x8d8af70]
>  Added content='jingle/2_content_66470052' type=ice-udp
> initiator=true [0x8d8af70]
>  Call(JG2_339938560). Outgoing
> from=wjgoogle@g.../D54DEA90
> to=2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=
> [0x8d8bea0]
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560_1">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560">
> 
> 
> 
> 
>  clockrate="8000"/>
> 
>  from="2036001234@v..."/>
> 
> 
> -----
>  Call(JG2_339938560). Changing state from Idle to
> Pending
> [0x8d8bea0]
>  Using audio content
> 'jingle/2_content_66470052' [0x8d8af70]
>  No-transport message received
> 
> YRTPWrapper::YRTPWrapper('192.168.1.10',0x8d8af70,'audio',bidir,0xb762f36c,false)
> [0x8d8cbf8]
>  YRTPWrapper::setupRTP("192.168.1.10",true) [0x8d8cbf8]
>  Session 'yrtp/1912729886' 0x8d8cd18 bound to
> 192.168.1.10:21330 +RTCP [0x8d8cbf8]
>  YRTPSource::YRTPSource(0x8d8cbf8) [0x8d8c6a8]
>  YRTPConsumer::YRTPConsumer(0x8d8cbf8) [0x8d8c7a8]
>  YRTPWrapper::setupSRTP(false) [0x8d8cbf8]
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560_2">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560">
> 
>  port="21330" network="0" protocol="udp" username="1100049458102717"
> password="***" type="local" preference="1"/>
> 
> 
> 
> -----
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560_3">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560">
>  port="21330" network="0" protocol="udp" username="1100049458102717"
> password="***" type="local" preference="1"/>
> 
> 
> -----
>  Failed to start RTP for
> content='jingle/2_content_66470052' candidates local=true
> remote=false
> [0x8d8af70]
> >>> DataTranslator::detachChain(0x8d81c90,0x8d89be8)
> <<< DataTranslator::detachChain
> >>> DataTranslator::detachChain(0x8d89af8,0x8d8a760)
> <<< DataTranslator::detachChain
>  disconnected. final=0 reason=(null) [0x8d85888]
>  DataTranslator::attachChain [0x8d8c6a8] '(null)' ->
> [0x8d8a760]
> 'speex/32000' not possible
>  DataTranslator::attachChain [0x8d81c90] 'speex/32000' ->
> [0x8d8c7a8] '(null)' not possible
>  Session terminated with reason='redirect'
> text='xmpp:2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI='
> [0x8d85888]
> ringing targetid is sip/2 peerid is id is jingle/2 callto is
>  RTP/AVP message received
>  Wrapper 0x8d8a518 found by CallEndpoint 0x8d81730
>  YRTPWrapper::startRTP("192.168.1.10",7078) [0x8d8a518]
>  YRTPSource::~YRTPSource() [0x8d89af8] wrapper=0x8d898c8
> ts=0
>  YRTPConsumer::~YRTPConsumer() [0x8d89be8]
> wrapper=0x8d898c8
> ts=0
>  YRTPWrapper::~YRTPWrapper() bidir 'audio' [0x8d898c8]
>  Cleaning up RTP 0x8d88db0 [0x8d898c8]
>  Hangup. reason=failure [0x8d85888]
>  disconnected. final=1 reason=failure [0x8d85888]
>  Destroyed [0x8d85888]
> connected targetid is jingle/1 peerid is jingle/2 id is sip/2 callto
> is
> connected targetid is sip/2 peerid is sip/2 id is jingle/2 callto is
>  'udp:0.0.0.0:5060' sending code 180 0x8d844f8 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
> From: ;tag=653474614
> To: ;tag=1676262022
> Call-ID: 615291946
> CSeq: 21 INVITE
> Server: YATE/3.3.3
> Contact: 
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
> Content-Type: application/sdp
> Content-Length: 189

> v=0
> o=yate 1324756677 1324756678 IN IP4 192.168.1.10
> s=SIP Call
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 28240 RTP/AVP 112 101
> a=rtpmap:112 SPEEX/32000
> a=rtpmap:101 telephone-event/8000
> ------
>  Receiving from 'gmail.com' [0x8d59b70]
> -----
>  to="wjgoogle@g.../D54DEA90" id="JG1_2036649621_2">
>  initiator="wjgoogle@g.../D54DEA90"
> responder="2036001234@v..." id="JG1_2036649621"
> xmlns="http://www.google.com/session">
> 
>  port="21808" network="0" protocol="udp" username="1894068495178298"
> password="***" type="local" preference="1"/>
> 
> 
> 
> 
> No such
> session
> 
> 
> -----
>  Processing event (0x8d8ac50,Iq)
>  Processing jabber.iq
> from=2036001234@v... to=wjgoogle@g.../D54DEA90
>  Receiving from 'gmail.com' [0x8d59b70]
> -----
>  from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560_1" type="result"/>
> -----
>  Processing event (0x8d89540,Iq)
>  Call(JG2_339938560). Sent element with
> id=JG2_339938560_1
> confirmed by result [0x8d8bea0]
>  Receiving from 'gmail.com' [0x8d59b70]
> -----
>  from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560_2" type="result"/>
> -----
>  Processing event (0x8d89540,Iq)
>  Call(JG2_339938560). Sent element with
> id=JG2_339938560_2
> confirmed by result [0x8d8bea0]
>  Receiving from 'gmail.com' [0x8d59b70]
> -----
>  from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="JG2_339938560_3" type="result"/>
> -----
>  Processing event (0x8d89540,Iq)
>  Call(JG2_339938560). Sent element with
> id=JG2_339938560_3
> confirmed by result [0x8d8bea0]
>  Receiving from 'gmail.com' [0x8d59b70]
> -----
>  from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> to="wjgoogle@g.../D54DEA90"
> id="jingle:10.12.239.27-20403969:1:57C7DBA9" type="set">
>  initiator="wjgoogle@g.../D54DEA90"
> xmlns:ses="http://www.google.com/session">
>  username="VQzIBL2ZSwe9wNbR" preference="1.0" protocol="udp"
> network="mediaproxy" generation="0" password="***" type="relay"/>
>  username="VQzIBL2ZSwe9wNbR" preference="0.6" protocol="tcp"
> network="mediaproxy" generation="0" password="***" type="relay"/>
>  username="VQzIBL2ZSwe9wNbR" preference="0.5" protocol="ssltcp"
> network="mediaproxy" generation="0" password="***" type="relay"/>
> 
> 
> -----
>  Processing event (0x8d89540,Iq)
>  Call(JG2_339938560). Candidates action set to
> candidates
> [0x8d8bea0]
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="jingle:10.12.239.27-20403969:1:57C7DBA9"/>
> -----
>  No-transport message received
>  Wrapper 0x8d8cbf8 found by CallEndpoint 0x8d8af70
>  YRTPWrapper::startRTP("74.125.91.126",19295) [0x8d8cbf8]
>  RTP starting format 'mulaw' payload 0 [0x8d8cbf8]
> >>> DataTranslator::detachChain(0x8d8c6a8,0x8d8a760)
> <<< DataTranslator::detachChain
> 
> SpeexCodec::SpeexCodec("slin/32000","speex/32000",encoding,2)
> [0x8d8b890]
>  Created DataTranslator 0x8d8b890 for 'mulaw' -> 'speex/32000'
> by
> factory 0x8c96e40 (len=3)
>  DataTranslator::attachChain [0x8d8c6a8] 'mulaw' -> [0x8d8a760]
> 'speex/32000' succeeded
> >>> DataTranslator::detachChain(0x8d81c90,0x8d8c7a8)
> <<< DataTranslator::detachChain
> 
> SpeexCodec::SpeexCodec("speex/32000","slin/32000",decoding,2)
> [0x8d8ac78]
>  Created DataTranslator 0x8d842e8 for 'speex/32000' -> 'mulaw'
> by
> factory 0x8c96e40 (len=3)
>  DataTranslator::attachChain [0x8d81c90] 'speex/32000' ->
> [0x8d8c7a8] 'mulaw' succeeded
>  RTP started for content='jingle/2_content_66470052'
> local='192.168.1.10:21330' remote='74.125.91.126:19295' [0x8d8af70]
>  Filter: Response authenticated for '74.125.91.126:19295'
> -
> notifying RTP. [0x8d83f50]
>  No-transport message received
>  Wrapper 0x8d8cbf8 found by ID 'yrtp/1912729886'
>  YRTPWrapper::startRTP("74.125.91.126",19295) [0x8d8cbf8]
>  Initial timeout in channel jingle/2 wrapper [0x8d8cbf8]
>  Receiving from 'gmail.com' [0x8d59b70]
> -----
>  from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> to="wjgoogle@g.../D54DEA90"
> id="jingle:10.12.239.27-20403969:1:57C7DC21" type="set">
>  initiator="wjgoogle@g.../D54DEA90"
> xmlns:ses="http://www.google.com/session">
> 
> 
> 
> 
> 
> 
> -----
>  Processing event (0x8d44b70,Iq)
>  Call(JG2_339938560). Changing state from Pending to
> Active [0x8d8bea0]
>  Sending to 'gmail.com' [0x8d59b70]
> -----
>  to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
> id="jingle:10.12.239.27-20403969:1:57C7DC21"/>
> -----
>  Remote peer answered the call [0x8d8af70]
>  Content 'jingle/2_content_66470052' removing media
> 8/alaw
> from offer [0x8d8af70]
>  RTP/AVP message received
>  Wrapper 0x8d8a518 found by CallEndpoint 0x8d81730
>  YRTPWrapper::setupSRTP(false) [0x8d8a518]
>  'udp:0.0.0.0:5060' sending code 200 0x8d7e4b8 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
> From: ;tag=653474614
> To: ;tag=1676262022
> Call-ID: 615291946
> CSeq: 21 INVITE
> Server: YATE/3.3.3
> Contact: 
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
> Content-Type: application/sdp
> Content-Length: 189

> v=0
> o=yate 1324756677 1324756679 IN IP4 192.168.1.10
> s=SIP Call
> c=IN IP4 192.168.1.10
> t=0 0
> m=audio 28240 RTP/AVP 112 101
> a=rtpmap:112 SPEEX/32000
> a=rtpmap:101 telephone-event/8000
> ------
>  'udp:0.0.0.0:5060' received 357 bytes SIP message from
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> ACK sip:6001234@1...:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK637423096
> From: ;tag=653474614
> To: ;tag=1676262022
> Call-ID: 615291946
> CSeq: 21 ACK
> Contact: 
> Max-Forwards: 70
> User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
> Content-Length: 0

> ------
>  RTP/AVP message received
>  Wrapper 0x8d8a518 found by CallEndpoint 0x8d81730
>  YRTPWrapper::startRTP("192.168.1.10",7078) [0x8d8a518]
>  Changing SSRC from 5CB0EAFB to 7682E186 in wrapper
> 0x8d8a518
>  'udp:0.0.0.0:5060' received 348 bytes SIP message from
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> REGISTER sip:192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1799621859
> From: ;tag=3712123
> To: 
> Call-ID: 612788096
> CSeq: 3 REGISTER
> Contact: 
> Max-Forwards: 70
> User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
> Expires: 100
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' sending code 100 0x8d7e4b8 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1799621859;received=192.168.1.10
> From: ;tag=3712123
> To: 
> Call-ID: 612788096
> CSeq: 3 REGISTER
> Server: YATE/3.3.3
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' sending code 401 0x8d86b88 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK1799621859;received=192.168.1.10
> From: ;tag=3712123
> To: 
> Call-ID: 612788096
> CSeq: 3 REGISTER
> WWW-Authenticate: Digest realm="Yate",
> nonce="866d6a8a378015c08a4940cf367dbbd5.1324756712", stale=FALSE,
> algorithm=MD5
> Server: YATE/3.3.3
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' received 537 bytes SIP message from
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> REGISTER sip:192.168.1.10 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK945362817
> From: ;tag=3712123
> To: 
> Call-ID: 612788096
> CSeq: 4 REGISTER
> Contact: 
> Authorization: Digest username="asus", realm="Yate",
> nonce="866d6a8a378015c08a4940cf367dbbd5.1324756712",
> uri="sip:192.168.1.10", response="86b3198a9969835f1fffcb4e80890148",
> algorithm=MD5
> Max-Forwards: 70
> User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
> Expires: 100
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' sending code 100 0x8d86488 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK945362817;received=192.168.1.10
> From: ;tag=3712123
> To: 
> Call-ID: 612788096
> CSeq: 4 REGISTER
> Server: YATE/3.3.3
> Content-Length: 0

> ------
>  Authenticating user asus with password length 8
>  Registered user asus via sip/sip:asus@1...:5063
>  Registered user 'asus' expires in 100 s
>  'udp:0.0.0.0:5060' sending code 200 0x8d7ab68 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK945362817;received=192.168.1.10
> From: ;tag=3712123
> To: ;tag=541259953
> Call-ID: 612788096
> CSeq: 4 REGISTER
> Expires: 100
> Contact: ;expires=100
> Server: YATE/3.3.3
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' received 357 bytes SIP message from
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> BYE sip:6001234@1...:5060 SIP/2.0
> Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK900340663
> From: ;tag=653474614
> To: ;tag=1676262022
> Call-ID: 615291946
> CSeq: 22 BYE
> Contact: 
> Max-Forwards: 70
> User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' sending code 100 0x8d83fe8 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 100 Trying
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK900340663;received=192.168.1.10
> From: ;tag=653474614
> To: ;tag=1676262022
> Call-ID: 615291946
> CSeq: 22 BYE
> Server: YATE/3.3.3
> Content-Length: 0

> ------
>  'udp:0.0.0.0:5060' sending code 401 0x8d80728 to
> 192.168.1.10:5063 [0x8c9cdf0]
> ------
> SIP/2.0 401 Unauthorized
> Via: SIP/2.0/UDP
> 192.168.1.10:5063;rport=5063;branch=z9hG4bK900340663;received=192.168.1.10
> From: ;tag=653474614
> To: ;tag=1676262022
> Call-ID: 615291946
> CSeq: 22 BYE
> WWW-Authenticate: Digest realm="Yate", domain="192.168.1.10",
> nonce="866d6a8a378015c08a4940cf367dbbd5.1324756712", stale=FALSE,
> algorithm=MD5
> Server: YATE/3.3.3
> Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
> Content-Length: 0

> ------

> On Sat, 2011-12-24 at 12:58 -0500, Xiaokui Qin wrote:
> > Hi Yate programmers,
> >
> > I am new to yate. I am trying to set up yate server that can
> > make/receive calls from google voice.
> >
> > So far I can call out and receive phone calls. But the problem is
> > that
> > when you make a outgoing call through google, in my sip client that
> > connects to yate doesn't play the ring tone. I can see the sip
> > client
> > says "remote rings", but it just doesn't make a ring tone. If I use
> > the
> > sip client calls an other sip client, I can hear the ring tone.
> >
> > Does anyone know how to fix this?
> >
> > Thanks
> >
> > Xiaokui Qin



I think you need this solution from last month's discussion. 180 Ringing without early media: http://yate.null.ro/archive/?action=show_msg&actionargs[]=67&actionargs[]=67



From: "Xiaokui Qin" <xkqin@y...>
To: yate@v...
Sent: Saturday, December 24, 2011 3:06:25 PM
Subject: Re: [yate] no ring when calling out using googlevoice

Hi Yate programmer,

Here are some more details.

1. The yate was compiled from svn on ubuntu 11.10.
2. The client is linphone.
3. Both sever and client are running on 192.168.1.10 behind a firewall.

Here are the logs from Yate:

------
<sip:INFO> 'udp:0.0.0.0:5060' received 887 bytes SIP message from
192.168.1.10:5063 [0x8c9cdf0]
------
INVITE sip:6001234@1... SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1425508363
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>
Call-ID: 615291946
CSeq: 20 INVITE
Contact: <sip:asus@1...:5063>
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   404

v=0
o=asus 123456 654321 IN IP4 192.168.1.10
s=A conversation
c=IN IP4 192.168.1.10
t=0 0
m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
a=rtpmap:112 speex/32000/1
a=fmtp:112 vbr=on
a=rtpmap:111 speex/16000/1
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000/1
a=fmtp:110 vbr=on
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
------
<sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x8d7edc0 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1425508363;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>
Call-ID: 615291946
CSeq: 20 INVITE
Server: YATE/3.3.3
Content-Length: 0

------
<sip/1:ALL> YateSIPConnection::YateSIPConnection(0x8d7f1e8,0x8d7ed40)
[0x8d7f7a8]
<INFO> Could not classify call from 'asus', wasted 59 usec
<cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/1'
<INFO> Routing call to '6001234' in context 'default' via '-' in 69 usec
<sip/1:MILD> Call rejected error='noauth' reason='(null)' [0x8d7f7a8]
<sip/1:ALL> YateSIPConnection::hangup() state=0 trans=0x8d7ed40
error='noauth' code=401 reason='(null)' [0x8d7f7a8]
<sip/1:ALL> YateSIPConnection::~YateSIPConnection() [0x8d7f7a8]
<sip:INFO> 'udp:0.0.0.0:5060' sending code 401 0x8d81ff8 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1425508363;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>
Call-ID: 615291946
CSeq: 20 INVITE
WWW-Authenticate: Digest realm="Yate",
nonce="1f90cff54a0caa5b677bd1ce86b6ebcf.1324756677", stale=FALSE,
algorithm=MD5
Server: YATE/3.3.3
Contact: <sip:6001234@1...:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' received 237 bytes SIP message from
192.168.1.10:5063 [0x8c9cdf0]
------
ACK sip:6001234@1... SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1425508363
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>
Call-ID: 615291946
CSeq: 20 ACK
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' received 1085 bytes SIP message from
192.168.1.10:5063 [0x8c9cdf0]
------
INVITE sip:6001234@1... SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1058117490
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>
Call-ID: 615291946
CSeq: 21 INVITE
Contact: <sip:asus@1...:5063>
Authorization: Digest username="asus", realm="Yate",
nonce="1f90cff54a0caa5b677bd1ce86b6ebcf.1324756677",
uri="sip:6001234@1...",
response="77f602664987b0ee47590890cd3dab58", algorithm=MD5
Content-Type: application/sdp
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE,
SUBSCRIBE, INFO
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Subject: Phone call
Content-Length:   404

v=0
o=asus 123456 654321 IN IP4 192.168.1.10
s=A conversation
c=IN IP4 192.168.1.10
t=0 0
m=audio 7078 RTP/AVP 112 111 110 3 0 8 101
a=rtpmap:112 speex/32000/1
a=fmtp:112 vbr=on
a=rtpmap:111 speex/16000/1
a=fmtp:111 vbr=on
a=rtpmap:110 speex/8000/1
a=fmtp:110 vbr=on
a=rtpmap:3 GSM/8000/1
a=rtpmap:0 PCMU/8000/1
a=rtpmap:8 PCMA/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
------
<sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x8d7c540 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>
Call-ID: 615291946
CSeq: 21 INVITE
Server: YATE/3.3.3
Content-Length: 0

------
<sip/2:ALL> YateSIPConnection::YateSIPConnection(0x8d80e30,0x8d812c0)
[0x8d81730]
<regfile:ALL> Authenticating user asus with password length 8
<INFO> Could not classify call from 'asus', wasted 28 usec
<cdrbuild:INFO> Got message 'call.route' for untracked id 'sip/2'
<INFO> Routing call to '6001234' in context 'default' via
'jingle/2036001234@v...' in 105 usec
<jingle:ALL> msgExecute. caller='wjgoogle@g.../D54DEA90'
called='2036001234@v...' online=true filetransfer=false
<jingle/1:ALL> Jingle version set to 0 from routing
<jingle/1:ALL> Session flags set to 1 from ojingle_flags=noping
[0x8d85888]
<jingle/1:CALL> Outgoing. caller='wjgoogle@g.../D54DEA90'
called='2036001234@v...' [0x8d85888]
<jingle/1:CALL> Calling. caller=wjgoogle@g.../D54DEA90
called=2036001234@v... [0x8d85888]
<jingle/1:ALL> Added content='jingle/1_content_86247556' type=ice-udp
initiator=true [0x8d85888]
<jgengine:ALL> Call(JG1_2036649621). Outgoing
from=wjgoogle@g.../D54DEA90 to=2036001234@v...
[0x8d886b0]
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="set" from="wjgoogle@g.../D54DEA90"
to="2036001234@v..." id="JG1_2036649621_1">
  <session xmlns="http://www.google.com/session" type="initiate"
initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v..." id="JG1_2036649621">
    <description xmlns="http://www.google.com/session/phone">
      <payload-type id="0" name="PCMU" clockrate="8000"/>
      <payload-type id="8" name="PCMA" clockrate="8000"/>
      <payload-type id="101" name="telephone-event" clockrate="8000"/>
      <payload-type id="101" name="audio/telephone-event"
clockrate="8000"/>
    </description>
  </session>
</iq>
-----
<jgengine:INFO> Call(JG1_2036649621). Changing state from Idle to
Pending [0x8d886b0]
<jingle/1:ALL> Using audio content
'jingle/1_content_86247556' [0x8d85888]
<yrtp:ALL> No-transport message received
<yrtp:ALL>
YRTPWrapper::YRTPWrapper('192.168.1.10',0x8d85888,'audio',bidir,0xb703b76c,false) [0x8d898c8]
<yrtp:ALL> YRTPWrapper::setupRTP("192.168.1.10",true) [0x8d898c8]
<yrtp:INFO> Session 'yrtp/1549148331' 0x8d88db0 bound to
192.168.1.10:21808 +RTCP [0x8d898c8]
<yrtp:ALL> YRTPSource::YRTPSource(0x8d898c8) [0x8d89af8]
<yrtp:ALL> YRTPConsumer::YRTPConsumer(0x8d898c8) [0x8d89be8]
<yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x8d898c8]
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="set" from="wjgoogle@g.../D54DEA90"
to="2036001234@v..." id="JG1_2036649621_2">
  <session xmlns="http://www.google.com/session" type="transport-info"
initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v..." id="JG1_2036649621">
    <transport xmlns="http://www.google.com/transport/p2p">
      <candidate name="rtp" generation="0" address="192.168.1.10"
port="21808" network="0" protocol="udp" username="1894068495178298"
password="***" type="local" preference="1"/>
    </transport>
  </session>
</iq>
-----
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="set" from="wjgoogle@g.../D54DEA90"
to="2036001234@v..." id="JG1_2036649621_3">
  <session xmlns="http://www.google.com/session" type="candidates"
initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v..." id="JG1_2036649621">
    <candidate name="rtp" generation="0" address="192.168.1.10"
port="21808" network="0" protocol="udp" username="1894068495178298"
password="***" type="local" preference="1"/>
  </session>
</iq>
-----
<jingle/1:NOTE> Failed to start RTP for
content='jingle/1_content_86247556' candidates local=true remote=false
[0x8d85888]
ringing targetid is sip/2 peerid is  id is jingle/1 callto is
<yrtp:ALL> RTP/AVP message received
<yrtp:INFO> Guessed local IP '192.168.1.10' for remote '192.168.1.10'
<yrtp:ALL>
YRTPWrapper::YRTPWrapper('192.168.1.10',0x8d81730,'audio',bidir,0x8d82280,false) [0x8d8a518]
<yrtp:ALL> YRTPWrapper::setupRTP("192.168.1.10",true) [0x8d8a518]
<yrtp:INFO> Session 'yrtp/1432917202' 0x8d7ed50 bound to
192.168.1.10:28240 +RTCP [0x8d8a518]
<yrtp:ALL> YRTPSource::YRTPSource(0x8d8a518) [0x8d81c90]
<INFO> DataTranslator::attachChain [0x8d81c90] '(null)' -> [0x8d89be8]
'(null)' not possible
<yrtp:ALL> YRTPConsumer::YRTPConsumer(0x8d8a518) [0x8d8a760]
<INFO> DataTranslator::attachChain [0x8d89af8] '(null)' -> [0x8d8a760]
'(null)' not possible
<yrtp:ALL> YRTPWrapper::startRTP("192.168.1.10",7078) [0x8d8a518]
<yrtp:INFO> RTP starting format 'speex/32000' payload 112 [0x8d8a518]
>>> DataTranslator::detachChain(0x8d81c90,0x8d89be8)
<<< DataTranslator::detachChain
<INFO> DataTranslator::attachChain [0x8d81c90] 'speex/32000' ->
[0x8d89be8] '(null)' not possible
>>> DataTranslator::detachChain(0x8d89af8,0x8d8a760)
<<< DataTranslator::detachChain
<INFO> DataTranslator::attachChain [0x8d89af8] '(null)' -> [0x8d8a760]
'speex/32000' not possible
<NOTE> Choosing started 'audio' format 'speex/32000' [0x8d83db8]
connected targetid is  peerid is jingle/1 id is sip/2 callto is
connected targetid is sip/2 peerid is sip/2 id is jingle/1 callto is
<sip:INFO> 'udp:0.0.0.0:5060' sending code 180 0x8d8a2a0 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>;tag=1676262022
Call-ID: 615291946
CSeq: 21 INVITE
Server: YATE/3.3.3
Contact: <sip:6001234@1...:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 189

v=0
o=yate 1324756677 1324756677 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 28240 RTP/AVP 112 101
a=rtpmap:112 SPEEX/32000
a=rtpmap:101 telephone-event/8000
------
<c2s/GoogleVoice:INFO> Receiving from 'gmail.com' [0x8d59b70]
-----
<iq type="error" from="2036001234@v..."
to="wjgoogle@g.../D54DEA90" id="JG1_2036649621_1">
  <session type="initiate" initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v..." id="JG1_2036649621"
xmlns="http://www.google.com/session">
    <description xmlns="http://www.google.com/session/phone">
      <payload-type id="0" name="PCMU" clockrate="8000"/>
      <payload-type id="8" name="PCMA" clockrate="8000"/>
      <payload-type id="101" name="telephone-event" clockrate="8000"/>
      <payload-type id="101" name="audio/telephone-event"
clockrate="8000"/>
    </description>
  </session>
  <error code="302" type="modify">
    <redirect
xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">xmpp:2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=</redirect>
    <ses:redirect
xmlns:ses="http://www.google.com/session">xmpp:2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=</ses:redirect>
  </error>
</iq>
-----
<jbclientengine:INFO> Processing event (0x8d89398,Iq)
<jgengine:NOTE> Call(JG1_2036649621). Sent element with
id=JG1_2036649621_1 confirmed by error. Terminating [0x8d886b0]
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="set" from="wjgoogle@g.../D54DEA90"
to="2036001234@v..." id="JG1_2036649621_4">
  <session xmlns="http://www.google.com/session" type="terminate"
initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v..." id="JG1_2036649621"/>
</iq>
-----
<jgengine:INFO> Call(JG1_2036649621). Changing state from Pending to
Ending [0x8d886b0]
<jgengine:INFO> Call(JG1_2036649621). Changing state from Ending to
Destroy [0x8d886b0]
<jingle/1:CALL> Redirecting to
'2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=' [0x8d85888]
<jingle:ALL> msgExecute. caller='wjgoogle@g.../D54DEA90'
called='2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI='
online=true filetransfer=false
<jingle/2:ALL> Jingle version set to 0 from routing
<jingle/2:ALL> Session flags set to 1 from ojingle_flags=noping
[0x8d8af70]
<jingle/2:CALL> Outgoing. caller='wjgoogle@g.../D54DEA90'
called='2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI='.
Transferred from=2036001234@v... [0x8d8af70]
<jingle/2:CALL> Calling. caller=wjgoogle@g.../D54DEA90
called=2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=
[0x8d8af70]
<jingle/2:ALL> Added content='jingle/2_content_66470052' type=ice-udp
initiator=true [0x8d8af70]
<jgengine:ALL> Call(JG2_339938560). Outgoing
from=wjgoogle@g.../D54DEA90
to=2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=
[0x8d8bea0]
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="set" from="wjgoogle@g.../D54DEA90"
to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560_1">
  <session xmlns="http://www.google.com/session" type="initiate"
initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560">
    <description xmlns="http://www.google.com/session/phone">
      <payload-type id="0" name="PCMU" clockrate="8000"/>
      <payload-type id="8" name="PCMA" clockrate="8000"/>
      <payload-type id="101" name="telephone-event" clockrate="8000"/>
      <payload-type id="101" name="audio/telephone-event"
clockrate="8000"/>
    </description>
    <transfer xmlns="urn:xmpp:jingle:transfer:0"
from="2036001234@v..."/>
  </session>
</iq>
-----
<jgengine:INFO> Call(JG2_339938560). Changing state from Idle to Pending
[0x8d8bea0]
<jingle/2:ALL> Using audio content
'jingle/2_content_66470052' [0x8d8af70]
<yrtp:ALL> No-transport message received
<yrtp:ALL>
YRTPWrapper::YRTPWrapper('192.168.1.10',0x8d8af70,'audio',bidir,0xb762f36c,false) [0x8d8cbf8]
<yrtp:ALL> YRTPWrapper::setupRTP("192.168.1.10",true) [0x8d8cbf8]
<yrtp:INFO> Session 'yrtp/1912729886' 0x8d8cd18 bound to
192.168.1.10:21330 +RTCP [0x8d8cbf8]
<yrtp:ALL> YRTPSource::YRTPSource(0x8d8cbf8) [0x8d8c6a8]
<yrtp:ALL> YRTPConsumer::YRTPConsumer(0x8d8cbf8) [0x8d8c7a8]
<yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x8d8cbf8]
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="set" from="wjgoogle@g.../D54DEA90"
to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560_2">
  <session xmlns="http://www.google.com/session" type="transport-info"
initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560">
    <transport xmlns="http://www.google.com/transport/p2p">
      <candidate name="rtp" generation="0" address="192.168.1.10"
port="21330" network="0" protocol="udp" username="1100049458102717"
password="***" type="local" preference="1"/>
    </transport>
  </session>
</iq>
-----
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="set" from="wjgoogle@g.../D54DEA90"
to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560_3">
  <session xmlns="http://www.google.com/session" type="candidates"
initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560">
    <candidate name="rtp" generation="0" address="192.168.1.10"
port="21330" network="0" protocol="udp" username="1100049458102717"
password="***" type="local" preference="1"/>
  </session>
</iq>
-----
<jingle/2:NOTE> Failed to start RTP for
content='jingle/2_content_66470052' candidates local=true remote=false
[0x8d8af70]
>>> DataTranslator::detachChain(0x8d81c90,0x8d89be8)
<<< DataTranslator::detachChain
>>> DataTranslator::detachChain(0x8d89af8,0x8d8a760)
<<< DataTranslator::detachChain
<jingle/1:CALL> disconnected. final=0 reason=(null) [0x8d85888]
<INFO> DataTranslator::attachChain [0x8d8c6a8] '(null)' -> [0x8d8a760]
'speex/32000' not possible
<INFO> DataTranslator::attachChain [0x8d81c90] 'speex/32000' ->
[0x8d8c7a8] '(null)' not possible
<jingle/1:INFO> Session terminated with reason='redirect'
text='xmpp:2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI=' [0x8d85888]
ringing targetid is sip/2 peerid is  id is jingle/2 callto is
<yrtp:ALL> RTP/AVP message received
<yrtp:ALL> Wrapper 0x8d8a518 found by CallEndpoint 0x8d81730
<yrtp:ALL> YRTPWrapper::startRTP("192.168.1.10",7078) [0x8d8a518]
<yrtp:ALL> YRTPSource::~YRTPSource() [0x8d89af8] wrapper=0x8d898c8 ts=0
<yrtp:ALL> YRTPConsumer::~YRTPConsumer() [0x8d89be8] wrapper=0x8d898c8
ts=0
<yrtp:ALL> YRTPWrapper::~YRTPWrapper() bidir 'audio' [0x8d898c8]
<ALL> Cleaning up RTP 0x8d88db0 [0x8d898c8]
<jingle/1:CALL> Hangup. reason=failure [0x8d85888]
<jingle/1:CALL> disconnected. final=1 reason=failure [0x8d85888]
<jingle/1:CALL> Destroyed [0x8d85888]
connected targetid is jingle/1 peerid is jingle/2 id is sip/2 callto is
connected targetid is sip/2 peerid is sip/2 id is jingle/2 callto is
<sip:INFO> 'udp:0.0.0.0:5060' sending code 180 0x8d844f8 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>;tag=1676262022
Call-ID: 615291946
CSeq: 21 INVITE
Server: YATE/3.3.3
Contact: <sip:6001234@1...:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 189

v=0
o=yate 1324756677 1324756678 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 28240 RTP/AVP 112 101
a=rtpmap:112 SPEEX/32000
a=rtpmap:101 telephone-event/8000
------
<c2s/GoogleVoice:INFO> Receiving from 'gmail.com' [0x8d59b70]
-----
<iq type="error" from="2036001234@v..."
to="wjgoogle@g.../D54DEA90" id="JG1_2036649621_2">
  <session type="transport-info" initiator="wjgoogle@g.../D54DEA90"
responder="2036001234@v..." id="JG1_2036649621"
xmlns="http://www.google.com/session">
    <transport xmlns="http://www.google.com/transport/p2p">
      <candidate name="rtp" generation="0" address="192.168.1.10"
port="21808" network="0" protocol="udp" username="1894068495178298"
password="***" type="local" preference="1"/>
    </transport>
  </session>
  <error code="404" type="cancel">
    <item-not-found xmlns="urn:ietf:params:xml:ns:xmpp-stanzas"/>
    <text xmlns="urn:ietf:params:xml:ns:xmpp-stanzas">No such
session</text>
  </error>
</iq>
-----
<jbclientengine:INFO> Processing event (0x8d8ac50,Iq)
<jbserverengine:ALL> Processing jabber.iq
from=2036001234@v... to=wjgoogle@g.../D54DEA90
<c2s/GoogleVoice:INFO> Receiving from 'gmail.com' [0x8d59b70]
-----
<iq to="wjgoogle@g.../D54DEA90"
from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560_1" type="result"/>
-----
<jbclientengine:INFO> Processing event (0x8d89540,Iq)
<jgengine:ALL> Call(JG2_339938560). Sent element with id=JG2_339938560_1
confirmed by result [0x8d8bea0]
<c2s/GoogleVoice:INFO> Receiving from 'gmail.com' [0x8d59b70]
-----
<iq to="wjgoogle@g.../D54DEA90"
from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560_2" type="result"/>
-----
<jbclientengine:INFO> Processing event (0x8d89540,Iq)
<jgengine:ALL> Call(JG2_339938560). Sent element with id=JG2_339938560_2
confirmed by result [0x8d8bea0]
<c2s/GoogleVoice:INFO> Receiving from 'gmail.com' [0x8d59b70]
-----
<iq to="wjgoogle@g.../D54DEA90"
from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="JG2_339938560_3" type="result"/>
-----
<jbclientengine:INFO> Processing event (0x8d89540,Iq)
<jgengine:ALL> Call(JG2_339938560). Sent element with id=JG2_339938560_3
confirmed by result [0x8d8bea0]
<c2s/GoogleVoice:INFO> Receiving from 'gmail.com' [0x8d59b70]
-----
<iq from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
to="wjgoogle@g.../D54DEA90"
id="jingle:10.12.239.27-20403969:1:57C7DBA9" type="set">
  <ses:session type="candidates" id="JG2_339938560"
initiator="wjgoogle@g.../D54DEA90"
xmlns:ses="http://www.google.com/session">
    <ses:candidate name="rtp" address="74.125.91.126" port="19295"
username="VQzIBL2ZSwe9wNbR" preference="1.0" protocol="udp"
network="mediaproxy" generation="0" password="***" type="relay"/>
    <ses:candidate name="rtp" address="74.125.91.126" port="19294"
username="VQzIBL2ZSwe9wNbR" preference="0.6" protocol="tcp"
network="mediaproxy" generation="0" password="***" type="relay"/>
    <ses:candidate name="rtp" address="74.125.91.126" port="443"
username="VQzIBL2ZSwe9wNbR" preference="0.5" protocol="ssltcp"
network="mediaproxy" generation="0" password="***" type="relay"/>
  </ses:session>
</iq>
-----
<jbclientengine:INFO> Processing event (0x8d89540,Iq)
<jgengine:ALL> Call(JG2_339938560). Candidates action set to candidates
[0x8d8bea0]
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="result" from="wjgoogle@g.../D54DEA90"
to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="jingle:10.12.239.27-20403969:1:57C7DBA9"/>
-----
<yrtp:ALL> No-transport message received
<yrtp:ALL> Wrapper 0x8d8cbf8 found by CallEndpoint 0x8d8af70
<yrtp:ALL> YRTPWrapper::startRTP("74.125.91.126",19295) [0x8d8cbf8]
<yrtp:INFO> RTP starting format 'mulaw' payload 0 [0x8d8cbf8]
>>> DataTranslator::detachChain(0x8d8c6a8,0x8d8a760)
<<< DataTranslator::detachChain
<speexcodec:ALL>
SpeexCodec::SpeexCodec("slin/32000","speex/32000",encoding,2)
[0x8d8b890]
<ALL> Created DataTranslator 0x8d8b890 for 'mulaw' -> 'speex/32000' by
factory 0x8c96e40 (len=3)
<ALL> DataTranslator::attachChain [0x8d8c6a8] 'mulaw' -> [0x8d8a760]
'speex/32000' succeeded
>>> DataTranslator::detachChain(0x8d81c90,0x8d8c7a8)
<<< DataTranslator::detachChain
<speexcodec:ALL>
SpeexCodec::SpeexCodec("speex/32000","slin/32000",decoding,2)
[0x8d8ac78]
<ALL> Created DataTranslator 0x8d842e8 for 'speex/32000' -> 'mulaw' by
factory 0x8c96e40 (len=3)
<ALL> DataTranslator::attachChain [0x8d81c90] 'speex/32000' ->
[0x8d8c7a8] 'mulaw' succeeded
<jingle/2:ALL> RTP started for content='jingle/2_content_66470052'
local='192.168.1.10:21330' remote='74.125.91.126:19295' [0x8d8af70]
<stun:NOTE> Filter: Response authenticated for '74.125.91.126:19295' -
notifying RTP. [0x8d83f50]
<yrtp:ALL> No-transport message received
<yrtp:ALL> Wrapper 0x8d8cbf8 found by ID 'yrtp/1912729886'
<yrtp:ALL> YRTPWrapper::startRTP("74.125.91.126",19295) [0x8d8cbf8]
<yrtp:WARN> Initial timeout in channel jingle/2 wrapper [0x8d8cbf8]
<c2s/GoogleVoice:INFO> Receiving from 'gmail.com' [0x8d59b70]
-----
<iq from="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
to="wjgoogle@g.../D54DEA90"
id="jingle:10.12.239.27-20403969:1:57C7DC21" type="set">
  <ses:session type="accept" id="JG2_339938560"
initiator="wjgoogle@g.../D54DEA90"
xmlns:ses="http://www.google.com/session">
    <pho:description xmlns:pho="http://www.google.com/session/phone">
      <pho:payload-type id="0" name="PCMU" clockrate="8000"/>
      <pho:payload-type id="101" name="telephone-event"/>
    </pho:description>
  </ses:session>
</iq>
-----
<jbclientengine:INFO> Processing event (0x8d44b70,Iq)
<jgengine:INFO> Call(JG2_339938560). Changing state from Pending to
Active [0x8d8bea0]
<c2s/GoogleVoice:INFO> Sending to 'gmail.com' [0x8d59b70]
-----
<iq type="result" from="wjgoogle@g.../D54DEA90"
to="2036001234@v.../srvres-MTAuMTIuMjM5LjI3Ojk4OTI="
id="jingle:10.12.239.27-20403969:1:57C7DC21"/>
-----
<jingle/2:CALL> Remote peer answered the call [0x8d8af70]
<jingle/2:ALL> Content 'jingle/2_content_66470052' removing media 8/alaw
from offer [0x8d8af70]
<yrtp:ALL> RTP/AVP message received
<yrtp:ALL> Wrapper 0x8d8a518 found by CallEndpoint 0x8d81730
<yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x8d8a518]
<sip:INFO> 'udp:0.0.0.0:5060' sending code 200 0x8d7e4b8 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1058117490;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>;tag=1676262022
Call-ID: 615291946
CSeq: 21 INVITE
Server: YATE/3.3.3
Contact: <sip:6001234@1...:5060>
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Type: application/sdp
Content-Length: 189

v=0
o=yate 1324756677 1324756679 IN IP4 192.168.1.10
s=SIP Call
c=IN IP4 192.168.1.10
t=0 0
m=audio 28240 RTP/AVP 112 101
a=rtpmap:112 SPEEX/32000
a=rtpmap:101 telephone-event/8000
------
<sip:INFO> 'udp:0.0.0.0:5060' received 357 bytes SIP message from
192.168.1.10:5063 [0x8c9cdf0]
------
ACK sip:6001234@1...:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK637423096
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>;tag=1676262022
Call-ID: 615291946
CSeq: 21 ACK
Contact: <sip:asus@1...:5063>
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Content-Length: 0

------
<yrtp:ALL> RTP/AVP message received
<yrtp:ALL> Wrapper 0x8d8a518 found by CallEndpoint 0x8d81730
<yrtp:ALL> YRTPWrapper::startRTP("192.168.1.10",7078) [0x8d8a518]
<yrtp:INFO> Changing SSRC from 5CB0EAFB to 7682E186 in wrapper 0x8d8a518
<sip:INFO> 'udp:0.0.0.0:5060' received 348 bytes SIP message from
192.168.1.10:5063 [0x8c9cdf0]
------
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK1799621859
From: <sip:asus@1...>;tag=3712123
To: <sip:asus@1...>
Call-ID: 612788096
CSeq: 3 REGISTER
Contact: <sip:asus@1...:5063>
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Expires: 100
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x8d7e4b8 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1799621859;received=192.168.1.10
From: <sip:asus@1...>;tag=3712123
To: <sip:asus@1...>
Call-ID: 612788096
CSeq: 3 REGISTER
Server: YATE/3.3.3
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' sending code 401 0x8d86b88 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK1799621859;received=192.168.1.10
From: <sip:asus@1...>;tag=3712123
To: <sip:asus@1...>
Call-ID: 612788096
CSeq: 3 REGISTER
WWW-Authenticate: Digest realm="Yate",
nonce="866d6a8a378015c08a4940cf367dbbd5.1324756712", stale=FALSE,
algorithm=MD5
Server: YATE/3.3.3
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' received 537 bytes SIP message from
192.168.1.10:5063 [0x8c9cdf0]
------
REGISTER sip:192.168.1.10 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK945362817
From: <sip:asus@1...>;tag=3712123
To: <sip:asus@1...>
Call-ID: 612788096
CSeq: 4 REGISTER
Contact: <sip:asus@1...:5063>
Authorization: Digest username="asus", realm="Yate",
nonce="866d6a8a378015c08a4940cf367dbbd5.1324756712",
uri="sip:192.168.1.10", response="86b3198a9969835f1fffcb4e80890148",
algorithm=MD5
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Expires: 100
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x8d86488 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK945362817;received=192.168.1.10
From: <sip:asus@1...>;tag=3712123
To: <sip:asus@1...>
Call-ID: 612788096
CSeq: 4 REGISTER
Server: YATE/3.3.3
Content-Length: 0

------
<regfile:ALL> Authenticating user asus with password length 8
<regfile:ALL> Registered user asus via sip/sip:asus@1...:5063
<sip:NOTE> Registered user 'asus' expires in 100 s
<sip:INFO> 'udp:0.0.0.0:5060' sending code 200 0x8d7ab68 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 200 OK
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK945362817;received=192.168.1.10
From: <sip:asus@1...>;tag=3712123
To: <sip:asus@1...>;tag=541259953
Call-ID: 612788096
CSeq: 4 REGISTER
Expires: 100
Contact: <sip:asus@1...:5063>;expires=100
Server: YATE/3.3.3
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' received 357 bytes SIP message from
192.168.1.10:5063 [0x8c9cdf0]
------
BYE sip:6001234@1...:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.10:5063;rport;branch=z9hG4bK900340663
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>;tag=1676262022
Call-ID: 615291946
CSeq: 22 BYE
Contact: <sip:asus@1...:5063>
Max-Forwards: 70
User-Agent: Linphone/3.3.2 (eXosip2/3.3.0)
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' sending code 100 0x8d83fe8 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 100 Trying
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK900340663;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>;tag=1676262022
Call-ID: 615291946
CSeq: 22 BYE
Server: YATE/3.3.3
Content-Length: 0

------
<sip:INFO> 'udp:0.0.0.0:5060' sending code 401 0x8d80728 to
192.168.1.10:5063 [0x8c9cdf0]
------
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
192.168.1.10:5063;rport=5063;branch=z9hG4bK900340663;received=192.168.1.10
From: <sip:asus@1...>;tag=653474614
To: <sip:6001234@1...>;tag=1676262022
Call-ID: 615291946
CSeq: 22 BYE
WWW-Authenticate: Digest realm="Yate", domain="192.168.1.10",
nonce="866d6a8a378015c08a4940cf367dbbd5.1324756712", stale=FALSE,
algorithm=MD5
Server: YATE/3.3.3
Allow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
Content-Length: 0

------


On Sat, 2011-12-24 at 12:58 -0500, Xiaokui Qin wrote:
> Hi Yate programmers,
>
> I am new to yate. I am trying to set up yate server that can
> make/receive calls from google voice.
>
> So far I can call out and receive phone calls. But the problem is that
> when you make a outgoing call through google, in my sip client that
> connects to yate doesn't play the ring tone. I can see the sip client
> says "remote rings", but it just doesn't make a ring tone. If I use the
> sip client calls an other sip client, I can hear the ring tone.
>
> Does anyone know how to fix this?
>
> Thanks
>
> Xiaokui Qin