Summary:	ASTERISK-27138: ooh323, no audio from Cisco CallManager Express ver.11.5 to asterisk
Reporter:	Dmitry Melekhov (slesru)	Labels:
Date Opened:	2017-07-18 01:43:39	Date Closed:	2017-08-15 02:42:07
Priority:	Major	Regression?
Status:	Closed/Complete	Components:	Addons/chan_ooh323
Versions:	13.13.1	Frequency of Occurrence
Related Issues:
Environment:	Centos 6	Attachments:	( 0) ast-27138.patch ( 1) asterisk.cap ( 2) h323_log ( 3) sladzar.cap ( 4) sladzar-notun.cap ( 5) sladzar-notun-notun.cap ( 6) sladzar-patched.cap
Description:	We need to establish h323 trunk from asterisk to cisco call manager express. Here is scheme PBX--isdn pri--asterisk--h323--ast-neftisa--h323--cisco Here is config from asterisk side: [sladzar] type=friend ;type=peer context=sladzar ip=10.56.6.1 port=1720 ;e164=101 disallow=all allow=alaw allow=ulaw ;allow=g729 fastStart=no ;fastStart=yes h245tunneling=yes ;canreinvite=no directmedia=yes directrtpsetup=yes dtmfmode=inband ;nat=yes When I as PBX user place call from asterisk to cisco call manager user eveything is fine. But if the same user calls me- call passes, but there is no audio at all. Looks like bug, because they say they can call cisco-cisco. Thank you!
Comments:	By: Asterisk Team (asteriskteam) 2017-07-18 01:43:40.143-0500 Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report. Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process]. By: Dmitry Melekhov (slesru) 2017-07-18 01:46:07.043-0500 Here is calls log, my number is 6401, call is from 761555 Thank you! By: Dmitry Melekhov (slesru) 2017-07-18 01:48:05.017-0500 Oops, sorry for attaching too large file- it contains more calls... btw, log is from ast-neftisa Thank you! By: Dmitry Melekhov (slesru) 2017-07-18 02:14:42.720-0500 btw, here is what I see on console:    -- OOH323/asterisk-17 is making progress passing it to OOH323/sladzar-16    -- OOH323/asterisk-17 is ringing       > 0x7f4520068310 -- Probation passed - setting RTP source address to 10.18.0.19:26094    -- Remote UNIX connection    -- Remote UNIX connection disconnected    -- Remote UNIX connection    -- Remote UNIX connection disconnected    -- OOH323/asterisk-17 answered OOH323/sladzar-16 10.18.0.19 is asterisk (connected to pbx) address, but there is no address for cisco callmanager :-( By: Rusty Newton (rnewton) 2017-07-19 13:29:13.492-0500 Assigning to the chan_ooh323 maintainer Alexander Anikin for triage. By: Alexander Anikin (may213) 2017-07-21 18:40:33.717-0500 Dmitry, could you please attach here capture files from ast-nefisa per both of legs of the call (ast-neftisa <-> asterisk and ast-neftisa <-> sladzar) it will more easiest to understand the issue. By: Dmitry Melekhov (slesru) 2017-07-24 01:18:21.294-0500 Hello! These files contains data gathered by tcpdump -s0 -vvv -i any -n host 10.56.6.1 -w sladzar.cap and tcpdump -s0 -vvv -i any -n host 10.18.0.19 -w asterisk.cap on ast-neftisa (10.18.0.6) I (from number 6401) called 761555 , this call is OK, and then I was called from 761555, at this call there is no audio. btw, we have the same scheme working for other endpoints, namely avaya and cisco cucm (not express), and it works. I.e. I don't think there is problem between 10.18.0.6 and 10.18.0.19. Thank you! By: Dmitry Melekhov (slesru) 2017-07-24 01:21:47.381-0500 btw, may be  this dumps can help with  ASTERISK-27137 , as you can see there is g729 traffic, although g729 is disabled ... By: Alexander Anikin (may213) 2017-08-02 05:41:37.977-0500 Hi Dmitry, I did some analysis of the issue. First thing is strange working of H.245 tunneling from CCME. You have enabled it on CCME peer (sladzar) but he reply with tunneling off in call from asterisk to CCME, you can see it on first call in the attached dump. Second call initiated by CCME and contain h245tunneling is on, but he don't reply to facility packet which tunnel H.245 TCS signal. It's reason of silence in that call. And it's could be CCME don't understand PROGRESS signal and block call completeley due to it. I recommend switch off h245tunneling on asterisk and retest this case. Second thing is about codec and can be adressed to ASTERISK-27137. First call inititated with G.729 codec which is disabled on sladzar peer and it's subject of 27137. But some time later call switch G.711U and work ok. I think CCME switch call codec to codec configured on final subscriber of CCME. Normal way for that is to send empty tcs packet and renegotiate full TCS/MSD procedure but CCME just replace codec on existing rtp session. Asterisk allow that so there are no sound problem. By: Dmitry Melekhov (slesru) 2017-08-02 06:19:37.409-0500 Hello! I attached two files notun is for call from asterisk to call manager express and back with: ooh323 show peer sladzar Name:          sladzar FastStart/H.245 Tunneling:no,no DirectRTP      yes EarlyDirectRTP yes DTMF Mode:     inband T.38 Mode:     faxgw/chan_sip compatible FAX Detect:         Cng AccountCode:   ast_h323 AMA flags:     Unknown IP:Port:       10.56.6.1:1720 OutgoingLimit: 0 rtptimeout:    0 nat:           no There is no voice :-( notun-notun is when tunneling is disabled on ccme  side tooo, result is the same, this file contains call only from ccme to asterisk. By: Dmitry Melekhov (slesru) 2017-08-02 06:20:57.397-0500 about g729, I heard the same complains while calling avaya pbx, but we just ignored this, because everything works, this is just, well, improvement... Thank you! By: Dmitry Melekhov (slesru) 2017-08-03 00:30:46.539-0500 About progress, I tried to add Answer to incoming context: {noformat} [sladzar] exten =>_XXXX,1,Set(CALLERID(number)=91${CALLERID(number)}) exten =>_XXXX,n,Answer() exten =>_XXXX,n,Dial(OOH323/${EXTEN}@asterisk) exten =>_XXXX,n,Hangup() {noformat} Unfortunately, still no voice :-( Thank you! By: Alexander Anikin (may213) 2017-08-06 16:23:20.687-0500 attached patch add send_progress parameter to [general] and user/friend section. By: Alexander Anikin (may213) 2017-08-06 16:26:14.169-0500 Hi Dmitry, please try with attached patch and add send_progress=no to [sladzar] or [general] section. If no success please attach capture of call here. By: Dmitry Melekhov (slesru) 2017-08-07 04:10:27.689-0500 Hello! Unfortunately, there is no voice , dump file sladzar-patched.cap contains call from me (6401) to 761000 and back.. Thank you! By: Dmitry Melekhov (slesru) 2017-08-07 04:18:11.075-0500 btw, I added parameter to peer [sladzar] as send_progress=no there is no progress in traffic dump, so I hope that I did everything right... Thank you! By: Dmitry Melekhov (slesru) 2017-08-14 23:21:08.363-0500 Hello! Alexander, your are absolutely right that cisco does not really works with slow start, enabling fast start solved problem. I.e. this is not asterisk problem at all, i.e. not a bug. Thank you! By: Alexander Anikin (may213) 2017-08-15 02:42:07.386-0500 Many tests with Cisco shown this is not an issue in asterisk. Modern Cisco IOS doesn't interact properly with other vendor devices and between two Cisco even when fast start is disabled. There are no problems when fast start is enabled. Tested on IOS 15.2 and  IOS XE 16.04, another versions of IOS may be affected also. Anyway there are no issue in asterisk, so close this ticket. Thanks Dmitry to massive testing.