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RE: Retrieving P-Asserted-Identity and Remote-Party-ID from SIP Message headers

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Hi,

I guess the main question here could be are you running the application in 1PCC or 3PCC mode.

I know it works on 1PCC mode after connected event, and suspect 3PCC you need to call it on receipt of the GC SIP 200ok event and not the connected event as it would have been cleared (or I suspect).

Regards,

Jeff

Jeff


Retrieving P-Asserted-Identity and Remote-Party-ID from SIP Message headers

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Apologies for the length of this question.

Some background; I have an application in which the user for a telephone is emailed the call details (caller, length of call etc) after the call is completed. If a phone rings but the usual operator is not available, any other user may pick up the call. I would like the email to go to the user who picked up the call rather than the absent user who was called. However, I am struggling for a method to identify  the user who picked up the call, from the SIP message header. I had hoped to idenify the user from one of two fields in the SIP message header associated with the GCEV_CONNECTED event: namely P-Asserted-Identity or Remote-Party-ID

Part of a TCP dump associated with the 200 OK message:

...
> Status-Line SIP/2.0 200 OK
> Message Header
    > Via: SIP/2.0/UDP 192.168.80.50:5060;branch=z9hG4bk-36-d451-eb5ea87-f716a208
    > From: "Ext 2910"<sip:2910@192.168.80.50>;tag=f7180638-3250a8c0-13c4-65014-36-3f373d36-36
    > To: <sip:2902@192.168.100.10>;tag=749046~6986e0d9-4f9b-4be5-a059-e690e7e506af-18995125
      Date: Tue, 19 Nov 2019 12:28:48 GMT
      Call-ID: f7199e40-3250a8c0-13c4-65014-36-7e604f1-=36
    > CSeq: 1 INVITE
      Allow: INVITE, OPTIONS, INFO, BYE, CANCEL, ACK, PRACK, UPDATE, REFERE, SUBSCRIBE, NOTIFY
      Allow-Events: presence
      Supported: replaces
      Supported: X-csco-srtp-fallback
      Supported: Geolocation
    > P-Asserted-Identity: "John Smith" <sip:2902@192.168.100.10>
    > Remote-Party-ID: "John Smith" <sip:2902@192.168.100.10>;party=called;screen=yes;privacy=off
    > Contact: <sip:2902@192.168.80.50:5060>
      Content-Type: application/sdp
      Content-Length: 274


Some bits of source code which are called on opening the virtual board(s), GlobalCall device(s), and on processing the GCEV_CONNECTED event:

//
int virtualBoardCreate (int boardCount, IP_VIRTBOARD *pVirtBoards)
{
    ...
    for (int i = 0; i < boardCount; i++)
    {
        IP_VIRTBOARD    *pVirtBoard = pVirtBoardsIdea;

        ...
        pVirtBoard->sip_signaling_port = 5060;
        pVirtBoard->sip_msginfo_mask = IP_SIP_MSGINFO_ENABLE | IP_SIP_MIME_ENABLE;
    }

    return 0;
}

//
int GlobalCallDeviceOnOpenEx (unsigned long DEVH)
{
    ...
    GC_PARM_BLKP    parmblkp = NULL;

    if ((result = gc_util_insert_parm_ref (&parmblkp, IPSET_CONFIG, IPPARM_REGISTER_SIP_HEADER, strlen (pAllowEvents) + 1, pAllowEvents)) != GC_SUCCESS)
        return result;

    if ((result = gc_util_insert_parm_ref (&parmblkp, IPSET_CONFIG, IPPARM_REGISTER_SIP_HEADER, strlen (pPAssertedIdentity) + 1, pPAssertedIdentity)) != GC_SUCCESS)
        return result;

    if ((result = gc_util_insert_parm_ref (&parmblkp, IPSET_CONFIG, IPPARM_REGISTER_SIP_HEADER, strlen (pRemotePartyId) + 1, pRemotePartyId)) != GC_SUCCESS)
        return result;

    return 0;
}

//
const char *getSetIDText (int setID)
{
    // convert numeric setID to text
    ...
}

//
const char *getParmIDText (int parmID)
{
    // convert numeric parmID to text
    ...
}

//
int GlobalCallDeviceGcevConnected (const METAEVENT& metaevent)
{
    GC_PARM_BLKP    parmblkp ((GC_PARM_BLKP) metaevent.extevtdatap);

    int parmctr = 0;

    GC_PARM_DATA_EXT    parm;
    INIT_GC_PARM_DATA_EXT(&parm);

    while ((gc_util_next_parm_ex (parmblkp, &parm)) == GC_SUCCESS)
    {
        printf ("\n");

        switch (parm.set_ID)
        {
            case IPSET_SIP_MSGINFO:
                printf ("parmctr : %i", parmctr++);
                printf ("set_ID  : %s", getSetIDText (parm.set_ID));
                printf ("parm_ID : %s",  getParmIDText (parm.parm_ID));
                printf ("pData   : %s", (char *) parm.pData);
                break;

            // all other known values of parm.set_ID catered for)
            ...
        }
    }

    return 0;
}

The output text generated by the function GlobalCallDeviceGcevConnected above:

parmctr : 0
set_ID  : IPSET_SDP
parm_ID : IPPARM_SDP_OFFER
pData   : v=0
 o=CiscoSystemsCCM-SIP 749223 1 IN IP4 192.168.100.10
 s=SIP Call
 c=IN IP4 192.168.100.10
 b=TIAS:64000
 b=AS:64
 t=0 0
 m=audio 24666 RTP/AVP 0 8 96 a=rtpmap:0 PCMU/8000
 a=ptime:20
 a=rtpmap:8 PCMA/8000
 a=rtpmap:96 telephone-event/8000
 a=fmtp:96 0-15

parmctr : 1
set_ID  : IPSET_SIP_MSGINFO
parm_ID : IPPARM_CONTACT_URI
pData   : sip:2902@192.168.100.10:5060^[0^P

parmctr : 2
set_ID  : IPSET_SIP_MSGINFO
parm_ID : IPPARM_CALLID_HDR
pData   : f7199e40-3250a8c0-13c4-65014-19-212ff6b3-19

parmctr : 3
set_ID  : IPSET_SIP_MSGINFO
parm_ID : IPPARM_FROM
pData   : "Ext 2910"<sip:2910@192.168.80.50>;tag=f7180638-3250a8c0-13c4-65014-19-afd3c09-19^[0^S

parmctr : 4
set_ID  : IPSET_SIP_MSGINFO
parm_ID : IPPARM_TO
pData   : <sip:2902@192.168.100.10>;tag=749223~6986e0d9-4f9b-4be5-a059-e690e7e506af-18995131

As can be seen, the fields I am interested in (P-Asserted-Identity and Remote-Party-ID are present in the SIP message header, but are not extracted into the IP parameter buffers.

I have referred to the manual globalcall_for_ip_hmp_v12.pdf, section 4.9 Setting and retrieving SIP Message Header Fields and to my alarm, in Section 4.9.4, Registering SIP Header Fields to be retrieved,

(pages 183-184), the two fields I want do not seem to be listed.

Have I missed something, or is there another method of retrieving a user's id from the SIP Message headers (or any other part of HMP / GlobalCall) ?

Regards

RE: Error while starting DM IP601-2E1-PCI-100BT

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Yes, indeed this go back a ways and I recall caveats at the time which Jaron had alluded to back then:

1. Make sure to use the host based mediaload load (starting with "ipvs_evr", such as ipvs_evr_2isdn_net5_311.pcd.)

2. Check your TDM bus settings and make sure the card is set to Alaw. In some builds of the 5.1.1 Feature Pack line, the bus settings will default to Mulaw, even though the card in the system might require Alaw.

I remember the second one being an issue back then.

Regards,

Jeff M.

RE: D/600JCT-2E1 Power Req

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Here is some information from technote written on this topic geared towards HMP, but applicable to the SR boards as well.

www.dialogic.com/.../JCT

Jeff M.

The SR 6.0 dependency on MSVCR80.dll

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Hi,

We've been having some problems with our SR 6.0 system, and traced the issue to a call by Dialogic into MSVCR80.dll

This is a fairly old version of the C++ Redistributable; from VS 2005

Are there any plans to update this in the next version of SR 6.0?

Thanks,

Brendan

RE: The SR 6.0 dependency on MSVCR80.dll

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Hi Brendan,

As of January 9th, 2018, the following the JCT and DM3 product lines as well as the SR6.0 software is now carried by Sangoma Technologies.  Sangoma now provides technical support and services for those products.  Please visit Sangoma's web site at www.sangoma.com/support for available support options and/or discussion forums.

Dan

RE: The SR 6.0 dependency on MSVCR80.dll

Download HMP driver now change, requires support or license.to access.

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We/Dialogic used to be able to allow our clients the ability to download and run two HMP ports for a short eval period with no constraints other than the eval period.

The new download link requires either a service contract or a purchase license ID.

This is unacceptable, seems that Montreal does not understand developers that must prove a a proof of concept.


RE: Download HMP driver now change, requires support or license.to access.

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We do offer different trial options.  If you contact your Dialogic sales channel, they can review available options.

Does Dialogic® HMP Interface Boards (• Dialogic® DNI/1210TEPEHMP Digital Network Interface Board) work with Diva SDK ?

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Does Dialogic® HMP Interface Boards (• Dialogic® DNI/1210TEPEHMP Digital Network Interface Board) work with Diva SDK ?

i mean does Does Dialogic® HMP Interface Boards (• Dialogic® DNI/1210TEPEHMP Digital Network Interface Board) work with software that develop Diva SDK , and does software detect DNI/1210TEPEHMP  Card.

Actual Release Info about PowerMedia HMP Linux development kit

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Where can I find actual release information about PowerMedia HMP Linux?

What are the current system requirements?

A few years ago we did a some SIP development on RedHat 6.6 - 32 bit with lnxHMP_4_1_165.tgz.

I tried to download the last version, but the product registrations at my account seems to have some problems, so downloads are not possible for me. 

How can I get the downloads working?

RE: Actual Release Info about PowerMedia HMP Linux development kit

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You can find the documentation to the PowerMedia HMP 4.1 release here www.dialogic.com/.../hmp41lin.  In particular you would want to look at the 'System Configuration Guide', 'Software Installation Guide' and 'Release Update' documents.  The Release Updates has the latest information.

The HMP 4.1 Linux software can be downloaded from www.dialogic.com/.../hmp-linux.   (Please Note: Users are required to have a valid support agreement to obtain the PowerMedia HMP Software).

RE: Actual Release Info about PowerMedia HMP Linux development kit

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Thanks Daniel,

In the release update I see:

Document Rev 31 - published January 8, 2016

Updates for Service Update 213.

...

1.17 Updated Operating System Support

With Service Update 236, Dialogic® PowerMedia™ HMP for Linux Release 4.1 supports

the following Red Hat Enterprise Linux (RHEL) and Community ENTerprise Operating

System (CentOS) versions in both 32-bit and 64-bit:

• Red Hat 7.1 or greater

• CentOS 7.1 or greater

...

1.31 Red Hat Enterprise Linux Release 7 Update 1

Support

Service Update 213 announces support for Red Hat Enterprise Linux Release 7 Update 1.

I do not see explicitly mentioned that Red Hat 8.1 is supported so far.

Is it planned to support Red Hat 8.1 soon?

RE: Actual Release Info about PowerMedia HMP Linux development kit

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Hi Teo -- adding Red Hat 8 support is being looked at my the product teams for 2020.  Right now it is in the planning stages.

RE: Actual Release Info about PowerMedia HMP Linux development kit

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Okay, that's clear.

Thanks for the quick response.


HMP for Windows 2012R2 with SU 395 crashing

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Hi there 

We are facing an application crashing issue with HMP 3 for Windows 2012-R2 (running on VMware Exsi 5.5) with SU 395, the application was successfully running on SU 320 on Windows 2008 however when we are running it on an updated SU 395 /windows 2012 the application started crashing (fault in: ucrtbase.dll) after a while… 

 Please note that we are able to regenerate this issue in our Lab, Kindly assist how we can fix this…

 

Thanks

Regards

RE: HMP for Windows 2012R2 with SU 395 crashing

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After doing several adjustments in the application, the application is now running OK ...

Thanks

Unable to start "Dialogic System Service"

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Hi,

I have installed HMP release 3.0 (build no. 241) in my Windows Server 2019 but I'm not able to start "Dialogic System Service" as the Media Server & HMP Media Server services are depended on this service?
Is HMP compatible with Win 2019? Is there any solution to get Dialogic System Service work? Find the attachments!

Thanks in advance!

BR

Sinan

RE: Unable to start "Dialogic System Service"

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Support for Windows Server 2019 Operating System was added in HMP 3.0 Service Update 393.

How to record Inbound and outboung call in HMP Windows

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Hi,

I have installed HMP Windows SU 395 in windows 2019 server. I am looking for how can I record the conversation between two calls. i.e. one inbound and one outbound call.

1. As of now I have an application which will record the inbound call which is fine where I am routing IPT & VOX channels and latter recording the call using dx_reciottdata.

2. But now in the same application I have to make outbound call specified by the caller (DTMF input).

I have couple of following questions.

1. for this outbound call, still do I need to route IPT & VOX channels for inbound call?

2. How do I connect inbound and outbound calls?

3. once connected, to record the conversation (may be called transaction) what channels need to be routed using nr_scroute. I think I need to use dx_mreciottdata().

4. Am I correct to think I need to have following resources

to record  -> 1 VOX channel,

inobund call -> 1 VOX (play prompts), 1 IPT,

outbound call -> 1 VOX (again for prompts as I need to inform the user the call being recorded), 1 IPT 

If I am missing something other than above, please guide me.

Thanks,

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