For the first one, was the call disconnected by your own digression at the application level, or by the remote side?
I know in some cases, the value may not be populated correctly for GC SS7 variant in comparison to other protocol (ie ISDN, VOIP). So that may be ok.
As opposed to real reason like the second one would seem more legit giving a factor as congestion sent by the remote switch. Meaning it cant handle call right now.
Jeff
Thank you Jeff,
Calls disconnected by remote site.
I will double check then feedback you
Hi,
I use Host license bicc, m3ua. I make call almost successfull. But sometimes I face this error on all channels when make call out going
GCEV_DISCONNECTED
gc_ResultInfo() successfully called
a_Info->gcValue = 0x506
a_Info->gcMsg = Event caused by call control library specific failure
a_Info->ccLibId = 5
a_Info->ccLibName = GC_SS7_LIB
a_Info->ccValue = 0x2c
a_Info->ccMsg = No description available
a_Info->additionalInfo =
�
or
GCEV_DISCONNECTED
gc_ResultInfo() successfully called
a_Info->gcValue = 0x547
a_Info->gcMsg = Congestion on the line
a_Info->ccLibId = 5
a_Info->ccLibName = GC_SS7_LIB
a_Info->ccValue = 0x2a
a_Info->ccMsg = No description available
a_Info->additionalInfo =
�
�
Are there any idea?
Hello Jeff Mason! Thank you for your reply and sorry for the delay to answer!
The SR installed is the last one 271 and the board is D/600-JCT-2E1.
Hi Jeff,
I attach more information ./gctload -t1, -t2, -t3
http://www.mediafire.com/download/yaz2k1ywt3b7rfv/gctload-t2.txt
There type of error above are an indication that HMP driver is not running properly or having issue on the system. When you reinstall, do you perform full uninstall first?
Are you saying now, that DCM does not show the HMP device?
Could be some problem at system or license level. Did you reactivate the license, did that work?
Jeff
We installed lates HMP for Windows 2012 Server. First everything worked well for a while, but then system stopped answer calls.
HMP Log:
03/19/2016 07:37:41.754 4488 5616 dm3low ERR1 Dm3MsgDispatcher *ERROR_SEM_TIMEOUT* Not an empty message
03/19/2016 07:37:41.754 4488 5616 dm3low ERR1 Dm3MsgDispatcher ERROR 0x79 on command message (C=0xc7)
03/19/2016 07:37:41.754 4488 5616 dm3low ERR1 Dm3MsgDispatcher *ERROR_SEM_TIMEOUT* Not an empty message
03/19/2016 07:37:41.754 4488 5616 dm3low ERR1 Dm3MsgDispatcher ERROR 0x79 on command message (C=0xc5)
Then I try to reinstall HMP and now I cant even setup HMP_software device? Licenses are good.
Any ideas?
HMP uninstall / install failed so OS is now rebuild and reconfigured.
Everything seems to work so far, but getting lines above on error log on incoming call
03/23/2016 09:11:13.933 1796 4124 libipm_ipvsc ERR1 Resource ::ReserveResource()-> All available Resource Reservations are in use.
03/23/2016 09:11:13.933 1796 4124 libipm_ipvsc ERR1 ReservationDBase ::ReserveResource()-> Reservation of Resource 1 failed.
03/23/2016 09:11:13.933 1796 5184 libipm_ipvsc ERR1 CIPVscChannel ipmB1C9 --- ::ReserveResource-> Resource Reservation Failed.
03/23/2016 09:11:13.933 1796 5184 libipm_ipvsc EXCE Ipmedialib ipmB1C9 ===> ipm_ReserveResource(): Exception=CIPMException,Line=6464,File=ipvscchannel.cpp
03/23/2016 09:11:30.473 1796 5732 sip_stack Error 00001664 ERROR - CALL - CallLegLock - Call 0x06760050: CallLeg object was destructed
The above will happen when you don't specify correct codecs in GC level for capabilities in respect to what is available in the license.
What is being set at the app level for coder capabilities.
Jeff
My Card support 2 E1 I worked With 1 E1 in the past Now I need add another Channel in My Card
Hi Sir I have Problem When I Want to start 2 E1
When I use 1 E1 I runing from my Application Ok
But when I running 2 E1 First one is Runing second One return error Can not set Handler System error.error=0
Can Help
We have quite frequent gc_DropCall timeouts in our system, and I'm hoping I could get some help here in fixing these issues.
Here are the specs:
Quite frequently under heavy load gc_DropCall will timeout (called in SYNC mode) and rendering the line unusable (gc_ResetLineDev will timeout too) and the only way to recover it is by restarting the boards.
Thank you in advance!
Claudiu
Hi All,
we're experiencing the same problem in our application, I mean we start recording and we get no audio recorded, the record ends with reason 0x00002 (TM_MAXSIL).
In our application we have tried both direct recording in a file and ec_stream function with user callback.
In order to understand what was going wrong we've made a test using the IPM_demo source code provided above, we've started two instances and opened two devices (device #1 and #2), then we've configured each instance RTP session with the ip and port used by the other instance, finally on one instance we started the play of Music_long.pcm and on the other a recording operation (file stream.pcm)... the result was silence in the recorded file.
Any suggestion?
The system used for tests was configured with HMP 3.0 SU361 and Windows 2008 (build 6002) SP2.
Regards,
Alberto Navatta
Alberto,
If you run wireshark on the recording side, will you see audio coming from the player?
I'd also recommend to run 2 instances on same machine and see if in this case you see audio in the file - this will eliminate any external network elements, since in such configuration RTP is not leaving the server, and is routed on the local NIC
Hi there.
The WWW-Authenticate header is included to UAS 401 challenge response upon REGISTER from UAC. This response does not generate any GlobalCall event; instead, GlobalCall compares this header against authorization data the app provides via gc_SetAuthenticationInfo( ) before calling gc_ReqService() and sends digest of authentication info in Authorization header of the second REGISTER; the app has no control of it. The GCEV_SERVICERESP is generated when REGISTAR subscription is terminated, whether with success or failure. If the final response from UAS contains WWW-Authenticate, you can retrieve it as described in 4,.9.2 and 4.9.4 at the location you specified above. However, I don't see how you can use this info: according to RFC2617, this header is a property of UAS, and UAC includes Authorization header as a response to 401/407. In other words, the app cannot set or retrieve WWW-Authenticate header in most cases, and it does not need to do that. All the app needs to take care of is to provide right quadruplets before gc_ReqService() is called, and if match between the quadruplets and WWW-Authenticate content is found, the second REGISTER with Authorze header is sent automatically by GlobalCall
Hi Leonid,
I have initially run the test exactly as you described: two instances on the same machine, and, in order to avoid errors in typing the IP and ports, I have copy & pasted the info from one instance on the other.
After that test just to confirm the streaming (the play side of the test) was ok I started the streaming again sending it towards another host (my workstation) where I have captured the stream with wireshark: output stream is ok, I've made the playout of the captured RTP stream and audio was present and clear.
My understanding is that there's a problem in the receiving path (even because in our application I can also confirm we ear the audio so dx/play --> ipm/streaming RTP is definitely ok), is there any way to perform low level debug on the system or do you have any other suggestion?
Thanks,
Alberto
Alberto,
Did I understand correctly that the wireshark on the recording side shows good audio?
If so, the only thing I can think of now is firewall or other service which can block UDP ports HMP is using. Please do the following:
1. Locate RtfConfigWin.xml (I guess it is Windows, otherwise the file is RtfConfigLin.xml)
2. Open the file in editor and find <!-- Voice Library -->, <!-- DM3 VOICE -->, <!-- ipvsc library --> and <!-- ipm library --> sections and set all labels to "1" (leave "Info" in 2 latter sections with "0").
3. Start DebugAngel by typing from command line "debugangel -start" or "debugangel -install", if the first command fails
Make a test. Collect DebugAngel.log and all recent rtf*.txt from hmp/log, zip it up and attach here, please.
Windows firewall is off.
I have not found any RtfConfigWin.xml file, there is a rtflog-LOCAL-xxx file, somne RtfService.lck and .pid and an RtpServerStatus but no config in dialogic/HMP directory.
In our application the users SIP phones are connected continuously.
An user can select a seconds incoming or outgoing SIP connection. The select second call is connected to the first call with gc_Listen() and disconnected with gc_UnListen().
This work fine for a few hours, but the longer the SIP phones are connected, the bigger becomes the delay in de SIP audio stream. When the SIP phone of the user is reconnected, then it works fine again for a few hours.
Can somebody explain me what is causing de delay in de SIP voice stream of the long connected SIP connection?
Is it possible to prevent/reset the delay without reconnecting the first connection?
Technical info:
version: lnxHMP_4_1_165
codec: GCCAP_AUDIO_g711Alaw64k
sip_max_calls = 180
RTPTIMEOUT Alarms enabled
