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Enc(11): WARNING: Total dropped frames

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Dear all,

Under linux, I found it has a lot of warning message under /var/log/message as follows:

[root@server]log]# grep Enc messages

Sep 20 10:43:53 server ssp_x86Linux_boot: Enc(82): WARNING: Total dropped frames = 102

Sep 20 10:44:40 server ssp_x86Linux_boot: Enc(84): WARNING: Total dropped frames = 2967

Sep 20 10:47:11 server ssp_x86Linux_boot: Enc(80): WARNING: Total dropped frames = 16596

Sep 20 10:48:20 server ssp_x86Linux_boot: Enc(86): WARNING: Total dropped frames = 1607

Sep 20 10:50:23 server ssp_x86Linux_boot: Enc(88): WARNING: Total dropped frames = 7993

Sep 20 10:50:30 server ssp_x86Linux_boot: Enc(5): WARNING: Total dropped frames = 699

Sep 20 10:55:09 server ssp_x86Linux_boot: Enc(19): WARNING: Total dropped frames = 8135

What's the meaning ? Any hidden problem on this server ?

Regards

Eddie


Receive 488 Not Acceptable Here after receive Re-Invite without SDP on Avaya SES

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Hello,

Just wondering whether HMP will be able to handle RE-INVITE with no SDP?

At the moment it seems the application send back "488 Not Acceptable Here", i was thinking it is about codec mismatch.

I still don't understand why Avaya SIP Enable Service (SES) sending SIP RE-INVITE without SDP.

The first INVITE that get accepted by the application shown below (i bold and italics the changed):

=======

INVITE sip:33002@192.168.17.98:5060 SIP/2.0
Call-ID: 808e6f5f4e4cde1e24a2220d100
CSeq: 1 INVITE
From: "33003" <sip:33003@192.168.17.130>;tag=808e6f5f4e4cde1d24a2220d100
Record-Route: <sip:192.168.17.130:5060;lr>,<sip:192.168.17.129:5061;lr;transport=tls>
To: "33002" <sip:33002@sessim.avaya.com>
Via: SIP/2.0/UDP 192.168.17.130:5060;branch=z9hG4bK8383830303038383834846.0,SIP/2.0/TLS 192.168.17.129;psrrposn=2;received=192.168.17.129;branch=z9hG4bK808e6f5f4e4cde1f24a2220d100,SIP/2.0/UDP 192.168.17.156:7318;psrrposn=1;received=192.168.17.156;branch=z9hG4bK-d8754z-3c51a73c06005e0c-1---d8754z-;rport=7318
Content-Length: 162
Content-Type: application/sdp
Contact: "33003" <sip:33003@192.168.17.129:5061;transport=tls>
Max-Forwards: 67
User-Agent: Avaya CM/R015x.01.0.414.0
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
Accept-Contact: *;+avaya-cm-line=1
Supported: 100rel,timer,replaces,join,histinfo
Alert-Info: <cid:external@sessim.avaya.com>;avaya-cm-alert-type=external
Min-SE: 1200
Session-Expires: 1200;refresher=uac
History-Info: <sip:33002@sessim.avaya.com>;index=1,"33002" <sip:33002@sessim.avaya.com>;index=1.1

v=0
o=- 1 1 IN IP4 192.168.17.129
s=-
c=IN IP4 127.0.0.2
b=AS:64
t=0 0
m=audio 2376 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000

======

 

and the second INVITE has no SDP:

 

======

INVITE sip:33002@192.168.17.98:5060 SIP/2.0
Call-ID: 808e6f5f4e4cde1e24a2220d100
CSeq: 2 INVITE
From: "33003" <sip:33003@192.168.17.130>;tag=808e6f5f4e4cde1d24a2220d100
To: "33002" <sip:33002@sessim.avaya.com>;tag=ab08090-0-13c4-50022-30f776-8e76635-30f776
Via: SIP/2.0/UDP 192.168.17.130:5060;branch=z9hG4bK838383030303838383704d.0,SIP/2.0/TLS 192.168.17.129;psrrposn=1;received=192.168.17.129;branch=z9hG4bK808e6f5f4e4cde11424a2220d100
Content-Length: 0
Contact: "33003" <sip:33003@192.168.17.129:5061;transport=tls>
Max-Forwards: 69
User-Agent: Avaya CM/R015x.01.0.414.0
Allow: INVITE,CANCEL,BYE,ACK,PRACK,SUBSCRIBE,NOTIFY,REFER,OPTIONS,INFO,PUBLISH
Supported: 100rel,timer,replaces,join,histinfo
Min-SE: 1200
Session-Expires: 1200;refresher=uac
Record-Route: <sip:192.168.17.130:5060;lr>

======

I tried to look at rtf log (attached)  but could not find anything. Any help regarding this would be much appreciated.

 

Network topology:

- HMP Application : ext 33002 : 192.168.17.98

- SIP PROXY - Avaya SES - 192.168.17.130

- Caller Extension using X-Lite application: ext 33003: 192.168.17.156

the scenario is 33003 calling 33002.

Thank you very much.

Best Regards,

Budi

Call Troubleshooting

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What is the current recommended call troubleshooting methodology?

Back in the day we would turn on RTF tracking. What is recommended now?

And while I have your attention, my global call header files are missing after I upgraded to SR6.0  release  275.

Where are they now, or how do I get them?

Problems suddenly with gc_GETCALLINFO

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My mature outbound calling application has suddenly developed a problem. The call is answered, but it appears the gc_GETCALLINFO function never returns with a result.

Any ideas what could suddenly cause this behavior . Other sites seem to be functioning properly with the same lines of code.

Dialogic SR6.0 build 275, springware boards

Problems again with GC_GetCallInfo

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Well, the problem is back.

My mature outbound calling application has developed a problem. The call is answered, but it appears the gc_GETCALLINFO function never returns with a result.

What timeout options are available?

Dialogic SR6.0 build 275, springware boards

No GCEV_UNBLOCKED after gc_OpenEx()

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We are using Dialogic SR 6.0 Build 271 on Windows using NI2 protocol.

We have an intermittent problem with our application first starts. What we see happening is that after the gc_OpenEx() call we do not receive the GCEV_UNBLOCKED event. When we attempt to start the service again, things come up cleanly.

The following are the errors that we see when this happens: 

10/09/2017 08:26:45.792 8880 13384 dm3low EXCE Exception at LibDm3Dti.cpp:578 (0x6001)
10/09/2017 08:26:45.792 8880 13384 spwrndi ERR1 ntftools LoadAllFormatters: Failed to load Formatter - sdntf.dll
10/09/2017 08:26:45.792 8880 17916 spwrndi ERR1 ntftools LoadAllFormatters: Failed to load Formatter - sdntf.dll
10/09/2017 08:26:45.792 8880 17904 spwrndi ERR1 ntftools LoadAllFormatters: Failed to load Formatter - sdntf.dll
10/09/2017 08:26:45.854 8880 17904 gc ERR1 gcdb ----- gcdb_ChangeCCLibFlags(flags:2, action:2) - gc_alarmdb_change_cclib_flags() returns: 68
10/09/2017 08:26:45.854 8880 17904 gc ERR1 gcdb ----- gcdb_InsertTimeSlotObject(devname:dtiB1T1, ldev:4) - gc_alarmdb_insert_time_slotEx() returns: 68
10/09/2017 08:26:45.854 8880 17904 gc ERR1 gcams_db ====> find_board_or_timeslot_by_app_handle() - _sr_claim(boardh:4) returns NULL

Attaching full RTF log.

DlgcHost_Utility::SystemException

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DlgcHost_Utility::SystemException issue

A simple program (opening voice device and successfully playing vox file) gives “terminate called after throwing on instance of ‘DlgcHost_Utility::SystemException’” on exit. This behavior is observed when program is compiled with third party library – Voice Recognition library.

The system is ISDN on DMV1200BTEP , Linux-CentOS, R4 app.

Any help on this issue would be highly appreciated.

Regards,

Halina

DIALOGIC HMP 18X RESPONSE RETREIVE VALUE

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Hi,

I am using HMP Linux 4.1 release .

When i use gc_MakeCall (SIP Technology ) when i get GCALL event (18x )  how to retirve value in the application to determine if it is 183 Session In progress  or 180 Rinigng event  ?


HMP 3.0 SU375 and ssp.mlm.sys bugcheck 0x000000d1

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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.

SIP: How to set default codecs for OPTIONS response

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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(&paramBlock,
GCSET_CHAN_CAPABILITY, IPPARM_LOCAL_CAPABILITY, sizeof(IP_CAPABILITY), &ipCap);

ipCap.direction = IP_CAP_DIR_LCLRECEIVE;

gc_util_insert_parm_ref_ex(&paramBlock,
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.

License Activation Failure

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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

HMP 3.0 Win GSM codec Support?

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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

Serial number of HMP virtual device

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Being that HMP does not rely on physical hardware, where does its serial number come from?

Where is the serial number stored?

Thanks.

Use Fax on D/600 JCT

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I use fx_open() to open fax device on D/600 JCT2E1 PCIe with channelname 'dxxxB1C1'

but it fails. how can I open fax Device on D/600 and how to get fax channelname if it differs with voice channelname.

Open Fax Device on D/600 JCT

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I am using fx_open() to open fax device on D/600 with channelname = dxxxB1C1

But it failes. what is channel name for fax device and how can I get fax channelnames.


Running HMP Linux with a user different than root

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Hi all,

I'd like to know if HMP 4.1 installed on RHEL 7, can run with a user different than root. Generally, running third-party software through the root account is a key aspect for customers who are very careful about security issues.

Would it be possible using a root account just for installation phase, and then configure the software to run with a different user account?

I'm able to start/stop HMP software using a user account (editing /etc/sudoers), but what I'm looking for is running processes with owner different by root.

Thanks,

Roberto

Cannot start dialogic service after switching to paid license

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I was testing HMP with a trial license.

Everything worked fine: I started the dialogic successfully service and was able to receive calls.

However,

The moment I switched to paid license (8-channel) suddenly it stopped working -- the dialog service does not start anymore:

I restarted the machine several times but no improvement.

I've attached the relevant items from the event viewer for your review:

(Please visit the site to view this file)

Please advise.

Thank you.

Different behavior between SU347 and SU375

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Hi,

I would to update release HMP from SU 347 to SU375 but I have a problem when whit this new release.

Avec un meme appel, le comportement de HMP est différent.
Les traces des 2 version sont en fichiers attachés.

thank you in advance for your help

Best regards

Yvan

Global call APIs sample C code?

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I am looking for a sample C source code that demonstrates the use of global call APIs and HMP? Simple code that includes:

Dialing

Playing a voice file

Getting DTMF keys

etc...

Thank you.

Michael

JCT Realtime streaming on Linux

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Is there any way to send a call to a host computer for realtime processing? I was thing about using the ec_stream() function to stream audio to a program that performs conferencing on the host. With the minimal driver and firmware buffer settings though, it appears to be impossible to get latency below 200 milliseconds for this task.

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