Hi,
Was this instance on a physical or Virtual Machine? (if VM what was the version)?
I guess the question is what type of call flow is going on here and what are the media operations (example use) of active calls?
How many active calls were going on?
Need a bit more info to understand the run-time environment.
Jeff
Hi everybody,
Several times there were system crashes in our HMP applications with the bugcheck 0x000000d1 (DRIVER_IRQL_NOT_LESS_OR_EQUAL) in the HMP system driver ssp.mlm.sys. It occurred in different servers Windows Server 2012 R2 (both physical and VM servers) with HMP 3.0 versions SU361 and SU372.
I made upgrading to SU375 for one of our customers about three months ago and defined detailed RTF logs. Recently we have received the same system crash there.
Attached file: ZIP file produced by the utility its_sysinfo + Windows minidump (included in the ZIp file).
I will be very grateful for the professional help.
Hi Jeff,
Thanks for your fast answer.
In the last case of the system crash there is Virtual Machine, version VMWare ESXi 5.1. It is a small contact center with no more than 20-30 active calls (waiting in queues and serviced by agents). SIP-based telephony (extensions and external trunks) with 1st party call control (1PCC) is used in this installation.
All other details you can find in the attached ZIP file. RTF detailed trace files contain 10 last minutes of call center activity till the system crash.
Regards,
Boris.
Actually the SDP content is correct as per RFC specification in this case:
"Specifically described here:
a=rtpmap: / [/ parameters>]
This attribute maps from an RTP payload type number (as used in
an "m=" line) to an encoding name denoting the payload format
to be used. It also provides information on the clock rate and
encoding parameters. It is a media-level attribute that is not
dependent on charset.
Although an RTP profile may make static assignments of payload
type numbers to payload formats, it is more common for that
assignment to be done dynamically using "a=rtpmap:" attributes.
As an example of a static payload type, consider u-law PCM
coded single-channel audio sampled at 8 kHz. This is
completely defined in the RTP Audio/Video profile as payload
type 0, so there is no need for an "a=rtpmap:" attribute, and
the media for such a stream sent to UDP port 49232 can be
specified as:
m=audio 49232 RTP/AVP 0"
Overall, this is why HMP does not send “a=” line info for basic codecs in the 200ok response or that of lies of INVITE message and any end-point should be able to understand it as per the mention in the "m=" line instead.
Jeff
Dear Ladies and Gentlemen,
We are trying to make our custom application to respond to OPTIONS request by means of the system default behavior. The problem is that the default behavior uses default codecs in the response (G723/G729). When we try to change default codecs (G711) for line devices by calling gc_SetUserInfo before calls to gc_WaitCall method the system starts responding without specifying any codec. How to force the system to respond with custom codecs and still using the default behavior.
// The startWaitCall method is called upon GCEV_UNBLOCKED event receiving
void startWaitCall(LINEDEV lineDevice)
{
IP_CAPABILITY ipCap;
ZeroMemory(&ipCap, sizeof(IP_CAPABILITY));
ipCap.capability = CODEC_AUDIO_G711ALAW64K;
ipCap.direction = IP_CAP_DIR_LCLTRANSMIT;
ipCap.type = GCCAPTYPE_AUDIO;
ipCap.extra.audio.frames_per_pkt = 20;
ipCap.extra.audio.VAD = GCPV_DISABLE;
GC_PARM_BLKP paramBlock = NULL;
gc_util_insert_parm_ref_ex(¶mBlock,
GCSET_CHAN_CAPABILITY, IPPARM_LOCAL_CAPABILITY, sizeof(IP_CAPABILITY), &ipCap);
ipCap.direction = IP_CAP_DIR_LCLRECEIVE;
gc_util_insert_parm_ref_ex(¶mBlock,
GCSET_CHAN_CAPABILITY, IPPARM_LOCAL_CAPABILITY, sizeof(IP_CAPABILITY), &ipCap);
gc_SetUserInfo(GCTGT_GCLIB_CHAN, lineDevice, paramBlock, GC_ALLCALLS);
gc_util_delete_parm_blk(paramBlock);
gc_WaitCall(lineDevice, NULL, NULL, 0, EV_ASYNC);
}
Below you may find request/response content for the default behavior after the line device codecs are changed. As you may see there are no codecs within SDP body
Sent 423 bytyes to 172.18.180.10:15060
OPTIONS sip:172.18.180.10:15060 SIP/2.0
Via: SIP/2.0/UDP 172.18.180.10:25060;branch=z9hG4bK3223my9dpm1mk12y23dl1to91;X-D
ptMsg=147
Call-ID: nn3o2tyod3p232t0d234pdppdyodj0j3@10.18.5.64
From: <sip:172.18.180.10:25060>;tag=tmpt2d2d
To: <sip:172.18.180.10:15060>
CSeq: 1 OPTIONS
Max-Forwards: 70
Allow: INVITE,ACK,CANCEL,OPTIONS,BYE,REGISTER,PRACK,INFO,UPDATE,SUBSCRIBE,NOTIFY
,MESSAGE,REFER,PUBLISH
Content-Length: 0
Received 637 bytes from 172.18.180.10:15060
SIP/2.0 200 OK
From: <sip:172.18.180.10:25060>;tag=tmpt2d2d
To: <sip:172.18.180.10:15060>;tag=5059470-ab412ac-3ad4-65014-b49c-5d3a1e2d-b49c
Call-ID: nn3o2tyod3p232t0d234pdppdyodj0j3@10.18.5.64
CSeq: 1 OPTIONS
Via: SIP/2.0/UDP 172.18.180.10:25060;branch=z9hG4bK3223my9dpm1mk12y23dl1to91;X-D
ptMsg=147
Allow: INVITE, CANCEL, ACK, BYE, OPTIONS, INFO
Accept-encoding:
Accept-language: en
Accept: application/sdp
Content-Type: application/sdp
Content-Length: 165
v=0
o=Dialogic_SIP_CCLIB 0 1 IN IP4 172.18.180.10
s=Dialogic_SIP_CCLIB
i=session information
c=IN IP4 172.18.180.10
t=0 0
m=audio 49152 RTP/AVP 8
a=ptime:20
Thank You.
Best Regards.
P.S. We see the codecs were changed by observing the traces on SIP INVITE requests during incoming calls.
Hello Jeff and thank you very much for the answer. However we still have a question - How to force the system to respond to OPTIONS request with custom codecs and still using the default behavior? We are ok with default OPTIONS handler we just want to change codecs. The code snippet would be of great help.
Thank You.
Best Regards.
Ok, just to clarify in this case so that I interpret the question correctly: You would like to change the codecs specified by HMP (from those listed by default) to only specific content in the SDP in this case. Meanwhile leaving the SIP stack handling the response (200ok) to the OPTIONS request received without application intervention for the OPTIONs handling, correct?
What specific codecs were you going to have listed in this case?
Are you saying that when you specify only one of two, other still show up? if so which ones?
Jeff
Hi,
From a quick look at the logs, I see some issues at the SIP stack level beforehand, but don't think they relate to the BSOD. But that is hard to say at this point based on the data available. When I checked the mini dump file itself, it was point towards an unexpected request received on an active data stream at the driver level (for media based operations). But that could be a resultant of incorrect behaviors happening before that.
Going back to the SIP side of things, I see that SIP call legs is maxed out as per following messages:
10/31/2017 10:11:10.543 4596 2464 sip_stack Error 000009A0 ERROR - CALL - CallLegMgrCheckObjCounters - already allocate 92 call-legs (hidden=1), maximum calls configuration 92.
10/31/2017 10:11:10.543 4596 2464 sip_stack Error 000009A0 ERROR - CALL - CallLegMgrCreateCallLeg - Cannot create new CallLeg due to CallLegMgr counters check
When I looked around this area, I see that the system is getting SUBSCRIBE requests. They are coming in every couple minutes:
10/31/2017 10:11:10.541 4596 2464 sip_stack Debug 000009A0 INFO - TRANSPORT - <-- SUBSCRIBE sip:5748@192.168.202.3 SIP/2.0
Are these to be expected? Does the application have handling for these?
If not, then it could be that the call-legs are getting leaked since you are not handling this at the application level potentially, in order to properly clean up the leg at the stack level. Thus leading to problems with overall Sip message handling.
I would suggest you look into the subscribe part first and if that was expected or not.
Jeff
Ok, so this is really a CPA issue then with respect to getting that result information after the call connected. How do you have the CPA based settings set in the cdp file in this case:
PSL_MakeCall_MediaDetect and PSL_MAKECALL_CALLPROGRESS
As these determine what events and control will be given to the application in this case when implementing CPA for out-dialing. Based on this setting it determine when the app would be calling gc_getcallinfo with respect to that termination event (GCEV_CONNECT vs GCEV_MEDIA_DETECTED). At what point are you calling the getcallinfo call?
Jeff
I am facing the issue in starting the Media for SIP in VM environment. I am using HMP 4.1 - 213 update.
CLI> conf system ipmedia start
updated- This action can take a few minutes to complete. Please wait
CLI>
EVENT-- SYSTEM Run State [Initializing]
CLI>
EVENT-- SYSTEM Run State [Fault]
CLI>
IpMedia start complete
In RTF logs this is the error printing
#Timestamp #PID #TID #Module #Client #Label #Message
11/09/2017 13:09:53.640 6953 4151634144 OAMIPC Warning OAMOSSL - OAMOSSL::OSSL_Socket::ClientConnect connect udom failed (111 - Connection refused)
11/09/2017 13:09:53.640 6953 4126145392 OAMIPC Warning OAMOSSL - OAMOSSL::OSSL_Socket::ClientConnect connect udom failed (111 - Connection refused)
11/09/2017 13:10:38.003 7163 127142768 OAMSYSLOG Error LicenseMangerAPI: - [CLicense::GetActiveLockCode()] failed to get lock code
11/09/2017 13:10:42.456 7163 127142768 OAMSYSLOG Error LicenseMangerAPI: - CLicense::GetNewJob(), Failed to get lockcode
11/09/2017 13:10:42.456 7163 127142768 OAMSYSLOG Error LicenseMangerAPI: - Method: CLicense::ActivateFull(), Exception: Error getting license feature list
11/09/2017 13:10:42.456 7163 127142768 OAMSYSLOG Error LicenseMangerAPI: - [CLicense::Activate()] Activate->Activate throwing exception
I have installed “Dialogic(R) Host Media Processing Software Release 4.1 LIN” and am trying activate Demo license.
/usr/dialogic/log/ has these failures:
-------------------------------------------------
10/05/2011 17:41:56.817 16167 117660528 OAMSYSLOG Info LicenseFileParser - ERROR: LicenseFileParser: FEATURE token verification failed. Read buffer:
10/05/2011 17:41:56.817 16167 117660528 OAMSYSLOG Info LicenseFileParser - ERROR: LicenseFileParser: FEATURE token verification failed. Read buffer:
10/05/2011 17:42:26.928 16167 117660528 OAMSYSLOG Info LicenseFileParser - ERROR: LicenseFileParser: FEATURE token verification failed. Read buffer:
10/05/2011 17:42:26.928 16167 117660528 OAMSYSLOG Info LicenseFileParser - ERROR: LicenseFileParser: FEATURE token verification failed. Read buffer:
10/05/2011 17:42:28.936 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - CLicense::GetNewJob(), Failed to get lockcode
10/05/2011 17:42:28.939 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - CLicense::GetNewJob(), Failed to get lockcode
10/05/2011 17:42:28.939 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - [CLicense::Checkout()] { failed in GetNewJob(), It maybe due to Node-Specific Information (NSI) mismatch}
10/05/2011 17:42:28.939 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - Method: CLicense::ActivateFull(), Exception: Error checking out feature 'Voice': error=22
10/05/2011 17:42:28.939 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - [CLicense::ActivateFull()], Exception: Error checking out feature 'Voice': Method: CLicense::ActivateFull(), Exception: Error checking out feature 'Voice': error=22
10/05/2011 17:42:28.939 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - [CLicense::Activate()] Activate->Activate throwing exception
10/05/2011 17:42:28.940 16167 117660528 OAMSYSLOG Info LicenseFileParser - ERROR: LicenseFileParser: FEATURE token verification failed. Read buffer:
10/05/2011 17:42:28.940 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - [CLicense::CLicense()] Exception: License file does not exist ->
10/05/2011 17:42:28.940 16167 117660528 OAMSYSLOG Error LicenseMangerAPI: - [CLicense::CLicense()] } Leave (Exception)
SHOW LICENCE gives these details:
--------------------------------------------------
CLI> show license
Selected License:
License Type: Verification
License Category: Node locked specific
License Seed:
Expiration Date:
License Options:
OEM License
Function:
Library:
License Directory: /usr/dialogic/data/
File Name: 2r2v2e2c2s2f2i2m2a2u2n2g_host_eva_000000000000.lic
License Type: Evaluation
License Category: Node locked specific
Expiration Date: 19-nov-2011
License Options: Voice 2, Enhanced_RTP 2, Conferencing 2, Speech_Integration 2, Fax 2, RTP_G_711 2, IP_Call_Control 2, Multimedia 2, AMR_NB 2, Native_Audio 4, Native_RTP 2, 3G-324M 2,
HMP and HMP Interface Boards
Serial Number: LK3D206F
Description: Host Media Processor
[root@localhost log]# uname -a
Linux localhost 2.6.32-131.0.15.el6.x86_64 #1 SMP Tue May 10 15:42:40 EDT 2011 x86_64 x86_64 x86_64 GNU/Linux
In previous Discussions pertaining to Licence activation failures, 2 fixes were suggested.
- installing compat-libstdc++-33
- and verifying `ping $(hostname)` works.
the above checks are verified, but still we face the"Node-Specific Information (NSI) mismatch" problem.
Kindly suggests.
thanks and regards,
Nayeem
Here is my code
System::String^ E1;
int i;
int rc;
IntPtr ptr = Marshal::StringToCoTaskMemAnsi(this->dataGridView1[3,channel-1]->Value->ToString());
char *DialString = (char*)ptr.ToPointer();
//enable call progress
GC_MAKECALL_BLK gcMakeCallBlk;
memset( &gcMakeCallBlk, 0, sizeof( gcMakeCallBlk));
PDK_MAKECALL_BLK pdkMakeCallBlk;
memset( &pdkMakeCallBlk, 0, sizeof( pdkMakeCallBlk));
pdkMakeCallBlk.flags = MEDIA_TYPE_DETECT;
gcMakeCallBlk.cclib = &pdkMakeCallBlk;
rc = gc_SetCallProgressParm(chinfo[channel].ldev,&capp);
if (rc != GC_SUCCESS)
{
gc_ErrorInfo(&t_Info);
E1 = gcnew System::String(t_Info.ccMsg);
}
// failsafe time test
DateTime t = System::DateTime::Now::get();
TimeSpan MinTime = TimeSpan(6,59,0);//no calls befor 6:59am
TimeSpan MaxTime = TimeSpan(22,1,0);//no calls after 10:01pm
if ((t.TimeOfDay <MinTime)){
MessageBox::Show("A call has been scheduled during the night."+Environment::NewLine+"This is not permited."+Environment::NewLine+"CallManager will need to be restarted.");
this->~Form1();
}
if ((t.TimeOfDay >MaxTime)){
MessageBox::Show("A call has been scheduled during the night."+Environment::NewLine+"This is not permited."+Environment::NewLine+"CallManager will need to be restarted.");
this->~Form1();
}
this->label12->Text = "LastDialChannel - "+channel;
rc = gc_MakeCall( chinfo[channel].ldev,&chinfo[channel].crn,DialString,&gcMakeCallBlk,60,EV_ASYNC);
----------------------------------------------------------------------------------------------------------------------------
rc = sr_waitevt(-1);
//MessageBox::Show("We have a board event - ");
if (gc_GetMetaEvent(&metaevent) != GC_SUCCESS) {
// process error
}
dev = sr_getevtdev(0);
// we have a gc event
for (channel = 1; channel <= ActiveLines; channel++) {
if (chinfo[channel].ldev == dev)
break;
}
gc_ResultInfo(&metaevent, &gc_info);
evttype = metaevent.evttype;
//this->dataGridView1[7,channel+ActiveLines+1]->Value = ":"+evttype;
switch (evttype)
case GCEV_CONNECTED:
this->dataGridView1[1,channel-1]->Value = "call answered";
chinfo[channel].CpaActive = false;
rc = gc_GetCallInfo(metaevent.crn, CONNECT_TYPE, &ConnectType);
I see a variable call pamd_timeout
Any info about implementation?
Ok, just to clarify in this case so that I interpret the question correctly: You would like to change the codecs specified by HMP (from those listed by default) to only specific content in the SDP in this case. Meanwhile leaving the SIP stack handling the response (200ok) to the OPTIONS request received without application intervention for the OPTIONs handling, correct?
Yes Sir we want to do precisely that. We want to change capabilities and/or codecs but still using default OPTIONS handler. As it stated in the www.dialogic.com/.../globalcall_for_ip_hmp_v12
"If a channel is available to handle the incoming request, Global Call sends a 200 OK message that includes an SDP message body (Content-Type: application/sdp) which indicates the same
capabilities that the library would report in an outgoing INVITE request." We want to override this behavior in the most simple way possible and do it for the OPTIONS request.
What specific codecs were you going to have listed in this case?
G711 and probably other but this one is for sure.
When I try to change capabilities I see the system doesn't show any codecs in response.
Best Regards.
Hi,
Is the RTP Enhanced licence enough to make a call with RTP encoded as GSM ?
Are is this not supported in the Win version?
Kind Regards,
Filip Hoste
I'm guessing I do the following
capp.ca_pamd_timeout = 50;
rc = gc_SetCallProgressParm(chinfo[channel].ldev,&capp);
before
rc = gc_MakeCall( chinfo[channel].ldev,&chinfo[channel].crn,DialString,&gcMakeCallBlk,60,EV_ASYNC);
then I'm guessing
gc_ResultInfo(&metaevent, &gc_info);
evttype = metaevent.evttype;
returns some special event type or is it something under
GCEV_CONNECTED:
=================================================================================
example code would be great
A few documents that may be helpful:
Call Progress Analysis: Global Call API Usage and Protocol Configuration Application Note -- this is an application that describes GC usage and also has demo associated with it.
all three links are no functioning
Hello Jeff,
If you need more information regarding this issue we can send you the slice of code where we try to setup the codecs.
Best Regards.
Hi Jeff,
Thanks a lot for your attention and very effective assistance. You are right that there is no appropriate processing of SIP SUBSCRIBE requests. These requests are sent, for example, by Snom IP Phones (used as agent extensions). I really didn't expect it and didn't notice their receiving in trace files.
Although I don't think that it is a serious reason for BSOD crashing, I agree with you that, first of all, it is worth handling SUBSCRIBE messages correctly.
Best regards,
Boris