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Messages - xevyk

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1
SIP to H.323 proxy / No SDP in incoming call from CUCM - Yate - FS
« on: April 01, 2016, 02:42:01 AM »
Here is another problem with incomng H323 calls from CUCM to Yate.
Call is drops with error:

20160401151811.946537 <sip/1:INFO> Not sending chan.rtp for empty media audio [0x7f6500006170]
20160401151811.946543 <sip/1:MILD> Removing failed SDP media 'audio' format '' from offer [0x7f6500006170]

And looks like CUCM doesn't send SDP info to Yate because of calls in H323. So in that way Yate have to generate some SDP info to FS to establish calls.


Full log available here: http://pastebin.com/Nxvky2By

2
SIP to H.323 proxy / Re: No RBT during calls
« on: April 01, 2016, 02:23:34 AM »
I don't know it: I don't know much about H323.
Maybe remote party don't send early media?
This might be the reason why remote RTP ip/port is received on answer.

Thank you for help!

I'll try to solve this puzzle and write back to this topic.

3
SIP to H.323 proxy / Re: No RBT during calls
« on: April 01, 2016, 01:59:24 AM »
Yate is receiving remote party RTP addr/port after answer.
No ringback or any other early media is possible before it.

See log:
20160401104223.017318 <h323/4:INFO> YateH323Connection::OnEstablished() [0x7f55a0009ca0]
...
20160401104223.027413 <h323/4:INFO> Got remote RTP address 10.74.100.5:4000 [0x7f55a0009ca0]

It looks like I have to change order of receiveng remote party RTP addr/port before answer but how can I do it?

4
SIP to H.323 proxy / Re: No RBT during calls
« on: March 31, 2016, 09:59:54 PM »
Can you post a log with all call signalling from start to end?
Yate messages (sniffer) included.

Thank you for reply again!

Here is log http://pastebin.com/CuzTLYQ7

5
SIP to H.323 proxy / No RBT during calls
« on: March 31, 2016, 05:31:54 AM »
Hello everyone!

I've stuck with another problem. I trying to make outbounding calls from FS to CUCM through Yate (FS & Yate on one box) without RTP Forwarding because CUCM confs doesn't support faststart. Calls are successfull and media goes fine but there is no RBT.

My regexroute.conf:

${formats}^\([^,]*\)=;formats=\1
${module}^sip$=h323/${called}@10.74.100.5
${module}^h323$=sip/sip:${called}@10.74.33.70;line=freeswitch
.*=-;error=forbidden;reason=Protocol not allowed

With RTP Forwarding enabled RBT works fine, but calls dropped with error due to disabled support of faststart on CUCM.

Here is call log with RTP Forwarding disabled and no RBT:

 0160331182652.770165 <sip:INFO> 'udp:0.0.0.0:5068' received 1133 bytes SIP message from 10.74.33.70:5080 [0x279c500]
 -----
 NVITE sip:5142@10.74.33.70:5068 SIP/2.0
 ia: SIP/2.0/UDP 10.74.33.70:5080;rport;branch=z9hG4bK3aBcyHeD2y8vH
 ax-Forwards: 69
 rom: "TEST 1" <sip:5000@10.74.33.70:5068>;tag=32S3FBggXSNQa
 o: <sip:5142@10.74.33.70:5068>
 all-ID: 4f07ed6c-71d6-1234-7faf-aad026afe127
 Seq: 89375742 INVITE
 ontact: <sip:gw+yate@10.74.33.70:5080;transport=udp;gw=yate>
 ser-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit
 llow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY
 upported: timer, path, replaces
 llow-Events: talk, hold, conference, refer
 ontent-Type: application/sdp
 ontent-Disposition: session
 ontent-Length: 313
 -FS-Support: update_display,send_info
 emote-Party-ID: "TEST 1" <sip:5200@10.74.33.70:5068>;party=calling;screen=yes;privacy=off
 
 =0
 =FreeSWITCH 1459399404 1459399405 IN IP4 10.74.33.70
 =FreeSWITCH
 =IN IP4 10.74.33.70
 =0 0
 =audio 24208 RTP/AVP 0 8 18 9 101
 =rtpmap:0 PCMU/8000
 =rtpmap:8 PCMA/8000
 =rtpmap:18 G729/8000
 =fmtp:18 annexb=no
 =rtpmap:9 G722/8000
 =rtpmap:101 telephone-event/8000
 =fmtp:101 0-16
 =ptime:20
------
 0160331182652.775440 <sip:INFO> 'udp:0.0.0.0:5068' sending code 100 0x7f559800ce40 to 10.74.33.70:5080 [0x279c500]
 -----
 IP/2.0 100 Trying
 ia: SIP/2.0/UDP 10.74.33.70:5080;rport=5080;branch=z9hG4bK3aBcyHeD2y8vH;received=10.74.33.70
 rom: "TEST 1" <sip:5000@10.74.33.70:5068>;tag=32S3FBggXSNQa
 o: <sip:5142@10.74.33.70:5068>
 all-ID: 4f07ed6c-71d6-1234-7faf-aad026afe127
 Seq: 89375742 INVITE
 erver: YATE/5.5.1
 ontent-Length: 0
 
------
20160331182652.775516 <sip/3:ALL> YateSIPConnection::YateSIPConnection(0x7f559800bdb0,0x7f55ac02d830) [0x7f5598002fe0]
20160331182652.775591 <sip/3:ALL> NAT address is '(null)' [0x7f5598002fe0]
20160331182652.775746 <sip/3:ALL> Set media: audio=g729 [0x7f5598002fe0]
20160331182652.778955 <INFO> Could not classify call from '5000', wasted 4 usec
20160331182652.779095 <INFO> Routing call to '5142' in context 'default' via 'h323/5142@10.74.100.5' in 78 usec
20160331182652.779106 <sip/3:ALL> NAT address is '(null)' [0x7f5598002fe0]
20160331182652.779155 <h323:INFO> Found call to H.323 target='5142@10.74.100.5'
20160331182652.783909 <h323:ALL> YateH323Connection::YateH323Connection(0x26d56f0,5784,0x7f5598001d20) [0x7f55b4000c10]
20160331182652.783922 <h323/3:ALL> YateH323Chan::YateH323Chan(0x7f55b4000c10,(null)) outgoing [0x7f55b4004b50]
20160331182652.783945 <h323/3:INFO> Setting H.323 caller: number='5000' name='TEST 1'
20160331182652.783952 <h323/3:ALL> YateH323Connection::rtpForward(0x7f5598001d20,0) [0x7f55b4000c10]
20160331182652.783956 <h323/3:NOTE> Formats changed to 'g729'
20160331182652.783980 <h323/3:ALL> Removing capability 'G.729B' (g729b) not in remote 'g729'
20160331182652.790985 <h323/3:ALL> YateH323Connection::OnSetLocalCapabilities() external [0x7f55b4000c10]
20160331182652.791025 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external [0x7f55b4000c10]
20160331182652.791041 <h323/3:ALL> Capability 'G.729A/B{sw}' format 'g729' session 1 receive
20160331182652.791069 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160331182652.791082 <yrtp:ALL> No-transport message received
20160331182652.791097 <yrtp:ALL> YRTPWrapper::YRTPWrapper('10.74.33.70',0x7f55b4004b50,'audio',receive,0x7f55bb7d6dd0,false) [0x7f55b4009b60]
20160331182652.791108 <yrtp:ALL> YRTPWrapper::setupRTP("10.74.33.70",true,true) [0x7f55b4009b60]
20160331182652.791130 <yrtp:INFO> Session 'yrtp/1910091913' 0x7f55b4009ee0 bound to 10.74.33.70:22190 +RTCP [0x7f55b4009b60]
20160331182652.791137 <yrtp:ALL> YRTPSource::YRTPSource(0x7f55b4009b60) [0x7f55b400a320]
20160331182652.791143 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x7f55b4009b60]
20160331182652.791168 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external [0x7f55b4000c10]
20160331182652.791178 <h323/3:ALL> Capability 'G.729A/B{sw}' format 'g729' session 1 send
20160331182652.791189 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160331182652.791198 <yrtp:ALL> No-transport message received
20160331182652.791202 <yrtp:ALL> Wrapper 0x7f55b4009b60 found by CallEndpoint 0x7f55b4004b50
20160331182652.791206 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x7f55b4009b60) [0x7f55b400a880]
20160331182652.791210 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x7f55b4009b60]
20160331182652.791232 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external [0x7f55b4000c10]
20160331182652.791240 <h323/3:ALL> Capability 'G.729A{sw}' format 'g729' session 1 receive
20160331182652.791250 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160331182652.791258 <yrtp:ALL> No-transport message received
20160331182652.791262 <yrtp:ALL> Wrapper 0x7f55b4009b60 found by CallEndpoint 0x7f55b4004b50
20160331182652.791265 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x7f55b4009b60]
20160331182652.791281 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external [0x7f55b4000c10]
20160331182652.791288 <h323/3:ALL> Capability 'G.729A{sw}' format 'g729' session 1 send
20160331182652.791297 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160331182652.791305 <yrtp:ALL> No-transport message received
20160331182652.791308 <yrtp:ALL> Wrapper 0x7f55b4009b60 found by CallEndpoint 0x7f55b4004b50
20160331182652.791312 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x7f55b4009b60]
20160331182652.791328 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external [0x7f55b4000c10]
20160331182652.791335 <h323/3:ALL> Capability 'G.729{sw}' format 'g729' session 1 receive
20160331182652.791344 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160331182652.791352 <yrtp:ALL> No-transport message received
20160331182652.791355 <yrtp:ALL> Wrapper 0x7f55b4009b60 found by CallEndpoint 0x7f55b4004b50
20160331182652.791358 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x7f55b4009b60]
20160331182652.791377 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external [0x7f55b4000c10]
20160331182652.791385 <h323/3:ALL> Capability 'G.729{sw}' format 'g729' session 1 send
20160331182652.791394 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160331182652.791401 <yrtp:ALL> No-transport message received
20160331182652.791405 <yrtp:ALL> Wrapper 0x7f55b4009b60 found by CallEndpoint 0x7f55b4004b50
20160331182652.791409 <yrtp:ALL> YRTPWrapper::setupSRTP(false) [0x7f55b4009b60]
20160331182653.263866 <h323/3:INFO> YateH323Connection::OnAlerting '5142' [0x7f55b4000c10]
20160331182653.265510 <yrtp:ALL> RTP/AVP message received
20160331182653.265547 <yrtp:INFO> Guessed local IP '10.74.33.70' for remote '10.74.33.70'
20160331182653.265553 <yrtp:ALL> YRTPWrapper::YRTPWrapper('10.74.33.70',0x7f5598002fe0,'audio',bidir,0x27ab130,false) [0x27b1a20]
20160331182653.265560 <yrtp:ALL> YRTPWrapper::setupRTP("10.74.33.70",true,true) [0x27b1a20]
20160331182653.265578 <yrtp:INFO> Session 'yrtp/1139616103' 0x27b2930 bound to 10.74.33.70:27396 +RTCP [0x27b1a20]
20160331182653.265584 <yrtp:ALL> YRTPSource::YRTPSource(0x27b1a20) [0x27b6ab0]
20160331182653.265590 <INFO> DataTranslator::attachChain [0x27b6ab0] '(null)' -> [0x7f55b400a880] '(null)' not possible
20160331182653.265594 <yrtp:ALL> YRTPConsumer::YRTPConsumer(0x27b1a20) [0x27b6c40]
20160331182653.265598 <INFO> DataTranslator::attachChain [0x7f55b400a320] '(null)' -> [0x27b6c40] '(null)' not possible
20160331182653.265602 <yrtp:ALL> YRTPWrapper::startRTP("10.74.33.70",24208) [0x27b1a20]
20160331182653.265607 <yrtp:INFO> RTP starting format 'g729' payload 18 [0x27b1a20]
20160331182653.265613 >>> DataTranslator::detachChain(0x27b6ab0,0x7f55b400a880)
20160331182653.265616 <<< DataTranslator::detachChain
20160331182653.265621 <INFO> DataTranslator::attachChain [0x27b6ab0] 'g729' -> [0x7f55b400a880] '(null)' not possible
20160331182653.265624 >>> DataTranslator::detachChain(0x7f55b400a320,0x27b6c40)
20160331182653.265627 <<< DataTranslator::detachChain
20160331182653.265630 <INFO> DataTranslator::attachChain [0x7f55b400a320] '(null)' -> [0x27b6c40] 'g729' not possible
 0160331182653.268333 <sip:INFO> 'udp:0.0.0.0:5068' sending code 180 0x27af4f0 to 10.74.33.70:5080 [0x279c500]
 -----
 IP/2.0 180 Ringing
 ia: SIP/2.0/UDP 10.74.33.70:5080;rport=5080;branch=z9hG4bK3aBcyHeD2y8vH;received=10.74.33.70
 rom: "TEST 1" <sip:5000@10.74.33.70:5068>;tag=32S3FBggXSNQa
 o: <sip:5142@10.74.33.70:5068>;tag=1646484832
 all-ID: 4f07ed6c-71d6-1234-7faf-aad026afe127
 Seq: 89375742 INVITE
 erver: YATE/5.5.1
 ontact: <sip:5142@10.74.33.70:5068>
 llow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
 ontent-Type: application/sdp
 ontent-Length: 204
 
 =0
 =yate 1459423613 1459423613 IN IP4 10.74.33.70
 =SIP Call
 =IN IP4 10.74.33.70
 =0 0
 =audio 27396 RTP/AVP 18 101
 =rtpmap:18 G729/8000
 =fmtp:18 annexb=no
 =rtpmap:101 telephone-event/8000
------

Help me please.

6
SIP to H.323 proxy / Re: Proxy Authentication Required
« on: March 24, 2016, 01:57:10 AM »
Please pay attention to routing rules: first match is handled.
You've added the new rule after the old one!

Problem solved! Thank you for your help!

7
SIP to H.323 proxy / Re: Proxy Authentication Required
« on: March 23, 2016, 09:17:42 PM »
You don't provide credentials to outgoing sip call.
Set the account when routing:

${module}^h323$=line/${called};line=freeswitch

Thank you for reply!
I've modified regexroute.conf to

${rtp_forward}possible=;rtp_forward=yes
${formats}^\([^,]*\)=;formats=\1
${module}^h323$=sip/sip:${called}@10.74.33.70;line=freeswitch
.*=-;error=forbidden;reason=Protocol not allowed

then: service yate restart

And error gone!
Thanks again!

8
SIP to H.323 proxy / [SOLVED] Proxy Authentication Required
« on: March 23, 2016, 05:52:31 AM »
Greetings everyone!

I trying to use Yate as H323 signalling proxy for Freeswitch but with no luck. Both of them installed on one box with IP 10.74.33.70. Yate configuration based on tutorial http://docs.yate.ro/wiki/H323_To_SIP_Signalling_Proxy

Then I've modified accfile.conf to

[freeswitch]
enabled=yes
protocol=sip
username=1002
description=Freeswitch account
interval=600
authname=1002
password=123
number=1002
domain=10.74.33.70
registrar=10.74.33.70
formats=alaw,mulaw,gsm
;outbound=10.0.0.1:5061
;localaddress=192.168.0.1:5062

regexroute.conf:

${rtp_forward}possible=;rtp_forward=yes
${formats}^\([^,]*\)=;formats=\1
${module}^h323$=sip/sip:${called}@10.74.33.70
.*=-;error=forbidden;reason=Protocol not allowed

1002 number is created and succefully tested in FS.

fs_cli show registrations

1001,10.74.33.70,0006538f-16b90003-587df546-41c99a8c@10.74.33.144,sofia/internal/sip:1001@10.74.33.144:5060;transport=udp,1458735743,10.74.33.144,50818,udp,freeswitch,
1002,10.74.33.70,1884231429@10.74.33.70,sofia/internal/sip:1002@10.74.33.70:5061,1458733753,10.74.33.70,5061,udp,freeswitch,
1000,10.74.33.70,122302058,sofia/internal/sip:1000@10.74.33.106:5062,1458735309,10.74.33.106,5062,udp,freeswitch,

3 total.


Also I have production Asterisk supporting SIP and H323 with IP 10.74.33.37 and registred ip-phones on it. When I trying to make a call from Asterisk with number 5142 in h323 to FS through Yate, I getting next messages in yate log:

------
20160323183109.004764 <sip:ALL> Sending UDP keepalive to 10.74.33.70:5060 for 'freeswitch'
20160323183135.005094 <sip:ALL> Sending UDP keepalive to 10.74.33.70:5060 for 'freeswitch'
 20160323183148.704142 <h323:ALL> YateH323Connection::YateH323Connection(0x1e508e0,15499,(nil)) [0x7fc03c045c00]
20160323183148.704174 <h323/3:ALL> YateH323Chan::YateH323Chan(0x7fc03c045c00,ip$10.74.33.37:3594) incoming [0x7fc03c04d040]
20160323183148.704262 <h323/3:ALL> YateH323Connection::OnSetLocalCapabilities() external passtrough [0x7fc03c045c00]
20160323183148.704454 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external passtrough [0x7fc03c045c00]
20160323183148.704480 <h323/3:ALL> Capability 'G.729A{sw}' format 'g729' session 1 send
20160323183148.704495 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160323183148.704508 <h323/3:ALL> YateH323_ExternalRTPChannel::OnReceivedPDU [0x7fc03c03c290]
20160323183148.704521 <h323:ALL> Remote RTP address 10.74.33.37:18726
20160323183148.704536 <h323/3:INFO> Got remote RTP address 10.74.33.37:18726 [0x7fc03c045c00]
20160323183148.704583 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external passtrough [0x7fc03c045c00]
20160323183148.704595 <h323/3:ALL> Capability 'G.729A{sw}' format 'g729' session 1 receive
20160323183148.704612 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160323183148.704624 <h323/3:ALL> YateH323_ExternalRTPChannel::OnReceivedPDU [0x7fc03c03af40]
20160323183148.704665 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external passtrough [0x7fc03c045c00]
20160323183148.704676 <h323/3:ALL> Capability 'G.729{sw}' format 'g729' session 1 send
20160323183148.704687 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160323183148.704698 <h323/3:ALL> YateH323_ExternalRTPChannel::OnReceivedPDU [0x7fc03c03d0c0]
20160323183148.704737 <h323/3:ALL> H323Connection::CreateRealTimeLogicalChannel external passtrough [0x7fc03c045c00]
20160323183148.704747 <h323/3:ALL> Capability 'G.729{sw}' format 'g729' session 1 receive
20160323183148.704759 <h323/3:ALL> Logical control channel address '10.74.33.70'
20160323183148.704770 <h323/3:ALL> YateH323_ExternalRTPChannel::OnReceivedPDU [0x7fc03c04cc90]
20160323183148.704852 <h323/3:INFO> YateH323Connection::OnAnswerCall caller='Dyakonov E.V. [10.74.33.37]' chan=0x7fc03c04d040 [0x7fc03c045c00]
20160323183148.704866 <h323/3:INFO> GetRemotePartyNumber()='5142'
20160323183148.704871 <h323/3:ALL> adr[0]='1000]
20160323183148.704874 <h323/3:INFO> Called number (alias) is '1000'
20160323183148.705068 <INFO> Could not classify call from '5142', wasted 3 usec
20160323183148.705110 <cdrbuild:INFO> Got message 'call.route' for untracked id 'h323/3'
20160323183148.705247 <INFO> Routing call to '1000' in context 'default' via 'sip/sip:1000@10.74.33.70' in 112 usec
20160323183148.705311 <sip/3:ALL> YateSIPConnection::YateSIPConnection(0x7fc03c03aba0,'sip:1000@10.74.33.70') [0x7fc0380010d0]
20160323183148.705345 <sip:ALL> YateSIPEndPoint::buildParty(0x7fc038001f30,'(null)',0,(nil))
20160323183148.705383 <sip/3:ALL> NAT address is '(null)' [0x7fc0380010d0]
20160323183148.705446 <sip/3:MILD> No formats for 'audio', excluding from SDP [0x7fc0380010d0]
20160323183148.705452 <sip/3:ALL> Set media: audio=g729 [0x7fc0380010d0]
20160323183148.705491 <h323/3:ALL> YateH323Connection::rtpExecuted(0x7fc03c03aba0) [0x7fc03c045c00]
20160323183148.705495 <h323/3:INFO> H323 Peer accepted RTP forward
 0160323183148.707515 <sip:INFO> 'udp:0.0.0.0:5061' sending 'INVITE sip:1000@10.74.33.70' 0x7fc038001f30 to 10.74.33.70:5060 [0x1f18220]
 -----
 NVITE sip:1000@10.74.33.70 SIP/2.0
 ax-Forwards: 20
 ia: SIP/2.0/UDP 10.74.33.70:5061;rport;branch=z9hG4bK2126407802
 rom: "TEST [10.74.33.37]" <sip:5142@10.74.33.70:5061>;tag=1131378280
 o: <sip:1000@10.74.33.70>
 all-ID: 1031467016@10.74.33.70:5061
 Seq: 4 INVITE
 ser-Agent: YATE/5.5.1
 ontact: <sip:5142@10.74.33.70:5061>
 llow: ACK, INVITE, BYE, CANCEL, REGISTER, REFER, OPTIONS, INFO
 ontent-Type: application/sdp
 ontent-Length: 95
 
 =0
 =yate 1458732708 1458732708 IN IP4 10.74.33.37
 =SIP Call
 =IN IP4 10.74.33.37
 =0 0
------
 0160323183148.708406 <sip:INFO> 'udp:0.0.0.0:5061' received 352 bytes SIP message from 10.74.33.70:5060 [0x1f18220]
 -----
 IP/2.0 100 Trying
 ia: SIP/2.0/UDP 10.74.33.70:5061;rport=5061;branch=z9hG4bK2126407802
 rom: "TEST [10.74.33.37]" <sip:5142@10.74.33.70:5061>;tag=1131378280
 o: <sip:1000@10.74.33.70>
 all-ID: 1031467016@10.74.33.70:5061
 Seq: 4 INVITE
 ser-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit
 ontent-Length: 0
 
------
 0160323183148.708453 <sip:INFO> 'udp:0.0.0.0:5061' received 854 bytes SIP message from 10.74.33.70:5060 [0x1f18220]
 -----
 IP/2.0 407 Proxy Authentication Required
 ia: SIP/2.0/UDP 10.74.33.70:5061;rport=5061;branch=z9hG4bK2126407802
 rom: "TEST [10.74.33.37]" <sip:5142@10.74.33.70:5061>;tag=1131378280
 o: <sip:1000@10.74.33.70>;tag=1pjgZ80rS6NjB
 all-ID: 1031467016@10.74.33.70:5061
 Seq: 4 INVITE
 ser-Agent: FreeSWITCH-mod_sofia/1.6.6+git~20160111T201612Z~d2d0b3283a~64bit
 ccept: application/sdp
 llow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
 upported: timer, path, replaces
Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, mes age-summary, refer
 roxy-Authenticate: Digest realm="10.74.33.70", nonce="ce7e5a26-6984-45de-bed8-63a4a866d7d7", algorithm=MD5, qop="auth"
 ontent-Length: 0
 
------
20160323183148.708677 <sip/3:ALL> YateSIPConnection::hangup() state=1 trans=0x7fc038004660 error='(null)' code=407 reason='Proxy Authentication Required' [0x7fc0380010d0]
20160323183148.708701 >>> YateH323Chan::disconnected() 'Proxy Authentication Required' [0x7fc03c04d040]
20160323183148.708713 <<< YateH323Chan::disconnected()
20160323183148.708718 <sip/3:ALL> YateSIPConnection::~YateSIPConnection() [0x7fc0380010d0]
 0160323183148.708740 <sip:INFO> 'udp:0.0.0.0:5061' sending 'ACK sip:1000@10.74.33.70' 0x7fc03c050fc0 to 10.74.33.70:5060 [0x1f18220]
 -----
 CK sip:1000@10.74.33.70 SIP/2.0
 ia: SIP/2.0/UDP 10.74.33.70:5061;rport;branch=z9hG4bK2126407802
 rom: "TEST [10.74.33.37]" <sip:5142@10.74.33.70:5061>;tag=1131378280
 o: <sip:1000@10.74.33.70>;tag=1pjgZ80rS6NjB
 all-ID: 1031467016@10.74.33.70:5061
 Seq: 4 ACK
 ax-Forwards: 20
 ontact: <sip:5142@10.74.33.70:5061>
 ser-Agent: YATE/5.5.1
 ontent-Length: 0
 
------
20160323183148.713894 <h323/3:ALL> YateH323Chan::~YateH323Chan() accepted h323/3 [0x7fc03c04d040]
20160323183148.714005 <h323/3:ALL> YateH323Connection::CleanUpOnCallEnd() [0x7fc03c045c00]
20160323183148.715978 <h323/3:INFO> YateH323Connection::OnCleared() error: '(null)' reason: EndedByLocalUser (0) [0x7fc03c045c00]
20160323183148.715996 <h323/3:ALL> YateH323Connection::~YateH323Connection() [0x7fc03c045c00]

In FS next log:

2016-03-23 18:31:48.692249 [NOTICE] switch_channel.c:1101 New Channel sofia/internal/5142@10.74.33.70:5061 [81075522-fe65-4a74-9814-872f3b75a1cc]
2016-03-23 18:31:48.692249 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/5142@10.74.33.70:5061) Running State Change CS_NEW
2016-03-23 18:31:48.692249 [DEBUG] sofia.c:9248 sofia/internal/5142@10.74.33.70:5061 receiving invite from 10.74.33.70:5061 version: 1.6.6 git d2d0b32 2016-01-11 20:16:12Z 64bit
2016-03-23 18:31:48.692249 [DEBUG] sofia.c:9415 IP 10.74.33.70 Rejected by acl "domains". Falling back to Digest auth.
2016-03-23 18:31:48.692249 [DEBUG] sofia.c:2147 detaching session 81075522-fe65-4a74-9814-872f3b75a1cc
2016-03-23 18:31:48.692249 [DEBUG] switch_core_state_machine.c:492 (sofia/internal/5142@10.74.33.70:5061) State NEW
2016-03-23 18:31:58.712167 [WARNING] switch_core_state_machine.c:576 81075522-fe65-4a74-9814-872f3b75a1cc sofia/internal/5142@10.74.33.70:5061 Abandoned
2016-03-23 18:31:58.712167 [NOTICE] switch_core_state_machine.c:579 Hangup sofia/internal/5142@10.74.33.70:5061 [CS_NEW] [WRONG_CALL_STATE]
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/5142@10.74.33.70:5061) Running State Change CS_HANGUP
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:739 (sofia/internal/5142@10.74.33.70:5061) Callstate Change DOWN -> HANGUP
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/5142@10.74.33.70:5061) State HANGUP
2016-03-23 18:31:58.712167 [DEBUG] mod_sofia.c:431 Channel sofia/internal/5142@10.74.33.70:5061 hanging up, cause: WRONG_CALL_STATE
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:60 sofia/internal/5142@10.74.33.70:5061 Standard HANGUP, cause: WRONG_CALL_STATE
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:741 (sofia/internal/5142@10.74.33.70:5061) State HANGUP going to sleep
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:508 (sofia/internal/5142@10.74.33.70:5061) State Change CS_HANGUP -> CS_REPORTING
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:473 (sofia/internal/5142@10.74.33.70:5061) Running State Change CS_REPORTING
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/5142@10.74.33.70:5061) State REPORTING
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:104 sofia/internal/5142@10.74.33.70:5061 Standard REPORTING, cause: WRONG_CALL_STATE
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:827 (sofia/internal/5142@10.74.33.70:5061) State REPORTING going to sleep
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:499 (sofia/internal/5142@10.74.33.70:5061) State Change CS_REPORTING -> CS_DESTROY
2016-03-23 18:31:58.712167 [DEBUG] switch_core_session.c:1646 Session 19 (sofia/internal/5142@10.74.33.70:5061) Locked, Waiting on external entities
2016-03-23 18:31:58.712167 [NOTICE] switch_core_session.c:1664 Session 19 (sofia/internal/5142@10.74.33.70:5061) Ended
2016-03-23 18:31:58.712167 [NOTICE] switch_core_session.c:1668 Close Channel sofia/internal/5142@10.74.33.70:5061 [CS_DESTROY]
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:630 (sofia/internal/5142@10.74.33.70:5061) Running State Change CS_DESTROY
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/5142@10.74.33.70:5061) State DESTROY
2016-03-23 18:31:58.712167 [DEBUG] mod_sofia.c:341 sofia/internal/5142@10.74.33.70:5061 SOFIA DESTROY
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:111 sofia/internal/5142@10.74.33.70:5061 Standard DESTROY
2016-03-23 18:31:58.712167 [DEBUG] switch_core_state_machine.c:640 (sofia/internal/5142@10.74.33.70:5061) State DESTROY going to sleep

Please help me to find a way to soltuion.

9
I solved this problem by the next way - as I thought problem is in new versions of ptlib an h323plus, but other (older) versions doesn't compile with Jessie.
First I installed Wheezy and successfully compiled ptlib-2.10.1 and h323plus-v1_23_0 and with this versions I built Yate from SVN.
After that I updated Wheezy to Jessie and installed Freeswitch 1.6.5

10
Greetings everyone!
I trying to complile Yate 5.5.0-1 with h323 support as h323-sip signaling proxy to Freeswitch but with no luck.
Vurtual Machine in Xen Server, OS Debian Jessie 64 bit clean default installation.
After compiling ptlib (v 2.12.8 ) and h323plus (v 1.26.5) I made

./configure --with-openh323=/usr/local/

as result:

...
checking for Pwlib in /usr/local... installed  RTTI: none
checking for OpenH323 in /usr/local/... installed 1.26.5
...

Seems OK, but compiling Yate stops with large amounts of errors:

...
In file included from /usr/local/include/ptlib.h:96:0,
                 from h323chan.cpp:28:
/usr/local/include/ptlib/unix/ptlib/thread.h: At global scope:
/usr/local/include/ptlib/unix/ptlib/thread.h:62:19: warning: ‘void* PX_ThreadMain(void*)’ declared ‘static’ but never defined [-Wunused-function]
     static void * PX_ThreadMain(void *);
                   ^
/usr/local/include/ptlib/unix/ptlib/thread.h:63:17: warning: ‘void PX_ThreadEnd(void*)’ declared ‘static’ but never defined [-Wunused-function]
     static void PX_ThreadEnd(void *);
                 ^
Makefile:306: ошибка выполнения рецепта для цели «h323chan.yate»
make[1]: *** [h323chan.yate] Error 1
make[1]: quit from directory «/usr/src/yate/modules»
Makefile:186: ошибка выполнения рецепта для цели «modules»
make: *** [modules] Errors 2

Please help.

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