I have used parts of the demo program including the gc_set_channel_codecs() function. When I call gc_SetUserInfo() fails. It returns with -1. The parameters seems to be OK...
Any idea why? I am using SIP.
Thanks.
Mike
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gc_SetUserInfo() fails
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SIP: How to change or remove QoS attributes in the SDP OK status
Dear Ladies and Gentlemen,
We are creating an application to handle incoming calls by means of the system default behavior. Everything goes ok until some callers send us QoS attributes in the INVITE request. In these cases, API just replies with the same attributes and the call is canceled/dropped before initiating. One way to make it work. according to this RFC https://tools.ietf.org/html/rfc3312 , is to send 183 back to caller thus initiating a sequence of request/responses to reserve resources. All that to met preconditions for session establishment. But according to https://www.dialogic.com/~/media/manuals/docs/globalcall_for_ip_hmp_v11.pdf the only way to do it is by means of gc_SipSessionProgress( ) which only can be used in 3PCC mode.
The gc_SipSessionProgress( ) function indicates to the originator that the call will be answered. This function provides a “183 Session Progress” to the destination party request acknowledging that the call has been received but is not yet answered. Upon successful completion of this function, the call state changes from the Offered state to the Accepted state. If the call state is already in the Accepted state, then it will remain in the Accepted state.
Notes:
1. In 1PCC mode, the SDP cannot be controlled by the user application. Therefore, gc_SipSessionProgress( ) cannot be used to change any SDP information.
2. The SDP can be set and sent when operating in 3PCC mode only
While this solution is the correct one there are time and resource limitations to implement it right now.
Another way to do it is to change SDP body part of "OK status" sent back to a caller where 'a=curr:qos' and 'a=des:qos' are replicated automatically by GC API itself. Is there any function to change that SDP body or simply remove QoS attributes that don't require 3PCC mode? Is there a function to disable replication of QoS attributes?
Appreciate your help.
Best Regards.
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gc_MakeCall() fails IPERR_BAD_PARAM
Hi,
I have a problem while calling the gc_MakeCall() API. It fails with the following error.
12/14 11:52:52.811 [GC_APIERR]: gc_MakeCall(linedev=1, numberstr=SIP:0526726565@82.166.90.3, mode=EV_ASYNC) Failed
GC ERROR: Invalid parameter
CC Name: GC_H3R_LIB
CC ERROR: IPERR_BAD_PARAM
Any idea why?
Thank you.
Mike
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gc_MakeCall() API fails when using the EV_SYNC,parameter
Hi,
When I run the gc_basic_call_model demo project and the gc_MakeCall() API is called using the EV_ASYNC parameter, the outgoing call is successful, however, when I use EV_SYNC, the API fails.
Any idea why?
Thanks.
Mike
gc_MakeCall(pline->ldev, &pline->call[callindex].crn, numberstr, makecallblkp, pline->makecall_timeout, EV_ASYNC)
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The GCEV_OPENEX event is not received in the process_event() function
Hi,
I have used the demo project as a skeleton for my program.
Any idea why the GCEV_OPENEX is not received in the process_event() function?
As far as I can see the device was opened successfully .
Thank you.
Mike
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gc_basic_call_model - how can I generate an outgoing call?
Hi,
I am using the gc_basic_call_model demo project using SIP technology.
Can you tell me what is required to generate a phone call? The demo project automatically generates a phone call. Assuming I would like to generate more phone calls, what is required?
Thank you.
Regards,
Mike
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Global call - dx_play() sample?
Hi,
Is there a sample code that demonstrates dialing and playing a wave file?
Thanks.
Mike
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gc_hmp_hold_retrieve demo project - the call is successful but the voice file is not heard
Hi,
I am running the gc_hmp_hold_retrieve demo project using SIP technology. The call is initiated, the phone rings, I pick-up the phone but I don't hear the voice file - music.pcm.
See the log below:
[GC_APICALL]: dx_getxmitslot() Success (ts=4096)
[GC_APICALL]: gc_Listen() Success (ts=4096)
[GC_APICALL]: gc_GetXmitSlot() Success (ts=4098)
[GC_APICALL]: dx_listen() Success (ts=4098)
[GC_APICALL]: Starting Play (playfile=music.pcm)
[STATE]: GCST_CONNECTED is the new GC call state after processing the event
Any idea why?
Thanks.
Mike
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HMP 4.1 su213 ssp_x86Linux_boot: null_cfsp.c: losing data.
Hi all,
we're experiencing the message “ssp_x86Linux_boot: null_cfsp.c: losing data.” in var/log/messages with HMP 4.1 SU213 installed on Linux RHEL 7.2. The system is installed on a ESXi virtual machine.
Did someone encounter the same message?
Thanks,
Roberto
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NET5 protocol on DMV1200 problem
I have two E1 links , one from PSTN and one from Ericson PBX.
I used CTR4 protocol for both links and D/600 board worked fine.
Now I changed my board to DMV1200 (DM3 board) and changed protocol to NET5.
E1 from PSTN link works fine, but E1 link from Ericson PBX doesnt sync.
I checked CRC (disabled) but did not help.
anyone can help what other settings may change from CTR4 to NET5 ?
how can I check if what is wrong , any tools?
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Send and Receive Fax on HMP
Hi
Is there any fax resource license for HMP?
can I send and receive fax with HMP and no physical board ?
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KB4056892 Causes Access Denied for HMP
Installing KB4056892 causes DCM to report errors:
Failed to get installed families(s)
Access is denied
Failed to get the system mode
Access is denied
Windows 10.0.16299.192
Dialogic SU375
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DCM error - The system is in start/stop pending mode...
Hi,
When I start the DCM program, I get the following error. Any idea why and how to fix?
Thanks.
Mike
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Windows update
Hi all,
Could anybody help me with 4 port Dialogic analogue module. Problem appears after windows update. After Windows update Dialogic sometime stop open ports and answer to calls. After restart of Dialogic modules it working about 1-2 hours then again stop open ports. Modules are in running state, but I we got continuous errors in Dialogic log file "5768 3424 spwrvoice ERR1 dxpoll dxxxB2C2 ----- _dx_wtringEx(): dx_getevt() Failed! returned -1." I have downloaded and installed latest software from Dialogic site. Maybe someone meet same problem.
Thanks,
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DTMF bounce?
I've been having a few problems with a D/480JCT-2T1 card; for some reason, there is an unrealistically high bounce rate on DTMF detection with dx_getdig(). For example, if I tell it to receive 11 digits, it'll terminate after five or six, and return back, for example if I dial 1-800-223, 11880022233. For whatever it's worth, this seems to be significantly less of an issue if I ask the board to only collect four digits. Is there anything obvious I could be missing here?
EDIT: I should mention the DTMF amplitude level is fine; I tried making a quick recording of them from the board. They're show up as -8.1 dB and undistorted.
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after GCEV_CONNECTED, how to send DTMF in HMP windows
Currently I have an HMP SIP application based on C++ which makes an outbound call. But Is there any way to dial an extension number or some authentication for example user id once I got GCEV_CONNECTED event?
In normal phone If I want to connect to phone number(1111111111) and dial some digits(2222) after connection, we could do by "1111111111,,2222".
Is it possible to do same with gc_MakeCall itself?
Thanks in advance.
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Including User-Agent header on ACK during outbound call setup
Hello,
I have a 1PCC application using HMP Windows. We are interfacing to a PBX which requires that all SIP messages generated by HMP include a User-Agent header.
I set a User-Agent header using gc_SetUserInfo() as described in section 4.9 of the GlobalCall IP for HMP Technology guide:
char * ua = "User-Agent: Support";
gc_util_insert_parm_ref(&gcParmBlk,IPSET_SIP_MSGINFO,IPPARM_SIP_HDR,(unsigned char)strlen(ua)+1,ua);
But not all messages generated from HMP show the User-Agent header.
On incoming calls (Invite-OK-ACK), the OK response from HMP includes the User agent. This works fine.
On outgoing calls (Invite-OK-ACK), the Invite from HMP includes the user agent, but the ACK does not. This causes the PBX to ignore the Ack and call setup fails.
Is there a way to add the User-Agent header to the ACK?
Thanks,
Tony
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DTMF in 183 Session in Progress
Hi,
When somebody makes a call to HMP (linux relase ) after i send 183 Session in Progress is it possible to get DTMF before 200 OK ?
After i answer the call with 200 OK it works ,but i would like to be able to detect DTMF before i send 200 OK because after i answer with 200 OK billing takes place.
I would appreciate any advice on this subject.
Thanks in advance
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H.100 bus between different cards?
Can an H.100 bus be shared between JCT cards and other card types, such as a NMS CG6565e? If so, can anyone share a configuration example? The Natural Microsystems card seems to see the A clock coming from the JCT card, but still insists the primary master isn't ready. This is my first configuration with multiple cards for whatever it's worth, and I assume I'm doing something stupid.
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MRCP v2.0 client library use
Hi,
Could you help please?
I have to implement support for ASR and TTS. Knowing that HMP 3.0 does not have such library and we have to make MRCP client based on third party MRCP v2.0 library, but what library and how it is used.
What is not clear for me how to incorporate gc_API, dx_API and third party MRCP API how voice channel is routed with MRCP library channel and so on
So after GCEV_CONNECTED how to connect MRCP channel and HMP voice channel?
From some articles and libraries it seems that HMP application and MRCP client are separate application is this a must or it is possible to have in one application HMP with MRCP client in it?
Is there some example HMP application with MRCP support?
Thank you in advance.
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