Hi.
We need a solution to connect an IVR using DNI board into another SIP Server with VoIP technology. Is it possible to connect it using rtp license and IP call control license? Please help me ASAP.
Armin
↧
Connecting to another SIP Server
↧
DM/IP SIP and NAT Traversal
We have a Global Call app (using a DI/0408-LS-A-R2) that accepts inbound calls from a telco service provider via SIP. Everything works fine when we have our box listen to a public IP. However, we'd like to have it sitting behind symmetric NAT. Since the inbound requests are coming from a public IP and all of the SIP header IP addresses coming from the telco are globally routable, it should work.
The issue is with the IP addresses we send back in the SDP information to the telco for RTP stream establishment in the 200 OK message when we call gc_AnswerCall. The IP address is the private one. It needs to be the public one. For example, the owner and connection information we are sending is:
o=Intel_IPCCLib 73715088 73715089 IN IP4 192.168.0.2
c=IN IP4 192.168.0.2
Instead, 192.168.0.2 needs to be replaced with our pubic IP or host name. Is there a way to set this in Global Call without binding the box to the public IP?
Thank you in advance,
Mark
The issue is with the IP addresses we send back in the SDP information to the telco for RTP stream establishment in the 200 OK message when we call gc_AnswerCall. The IP address is the private one. It needs to be the public one. For example, the owner and connection information we are sending is:
o=Intel_IPCCLib 73715088 73715089 IN IP4 192.168.0.2
c=IN IP4 192.168.0.2
Instead, 192.168.0.2 needs to be replaced with our pubic IP or host name. Is there a way to set this in Global Call without binding the box to the public IP?
Thank you in advance,
Mark
↧
↧
RE: DM/IP SIP and NAT Traversal
09/25/2015
Dear Dialogic
I'm working with a DM/IP241-100 board with the latest version of SR 6.0. My client has 20 of these boards on several systems that we'd like to reconfigure from a static IP to the other side of a router without a static IP.
I just read the above post about DM/IP SIP and NAT Traversal written 7 years ago. Is this still true today ? As of Oct 2007 the drivers did not support NAT through a router and OpenSER http://openser.org/ was the best answer.
With the latest Dialogic drivers for Windows and/or the Dialogic API now support this ?
Is OpenSER still the best answer today ?
Dan
St Louis, MO
↧
RE: SIP Device names
Operating system: Red Hat Enterprise Linux Server release 6.3 (Santiago)
HMP details:
System Release ProductName = Dialogic(R) Host Media Processing Software Release 4.1 LIN
System Release Platform = RMS
I ran the demo application CnfConferencingDemo, which failed for exactly the same reason. As a red herring, it suggested a license failure, which isn't the case, but also suggested ("service started? other APP running HMP?)" How do I check for these conditions?
I have a single source file, test1.cpp, which merely tries to execute gc_start (). The build options are:
Compile: g++ -v -g -c -Wall -DLINUX -Dlint -D_HMP15 -o ./test1.o -I/usr/dialogic/inc
Link: g++ -v -g -o ./test1 ./test1.o -L/usr/dialogic/lib -L/usr/lib -lgc
NB I have tried compiling with -D_HMP30
↧
SIP Device names
Apologies if this is a newbie question but:
int mode = SR_POLLMODE;
if (sr_setparm (SRL_DEVICE, SR_MODEID, &mode) != -1)
{
// Enable the event handler
if (sr_enbhdlr (EV_ANYDEV, EV_ANYEVT, (long (*) (void *)) evt_hdlr) != -1)
{
LINEDEV ldev;
if (gc_OpenEx (":P_SIP:N_iptB1T1:M_ipmB1C1", EV_SYNC, NULL]) != GC_SUCCESS)
{
GC_INFO info;
gc_ErrorInfo (&info);
// info. gcValue = -x8e;
// info.gcMsg = "Invalid Device Name"
}
}
}
I have tried various permutations and combinations of the name, including dispensing with the IP Media device, using ":P_IP" instead of "P_SIP" etc. The names I have tried to use have been lifted from Page 476 ("8.3.18 gc_OpenEx( ) Variances for IP") of /www.dialogic.com/~/media/manuals/docs/globalcall_for_ip_hmp_v8.pdf
"Invalid device name" is not very helpful without some indication of what is invalid about it. Is there a mechanism by which I can discover what is wrong with the name, and what device name formats and / or content would be valid?
↧
↧
RE: SIP Device names
If it helps, I have the RTF output for the last attempt (timestamp and PID fields omitted for brevity
#TID #Module
3078355648 sm_main.cpp:340 ! 0 ! gc_h3r:>> h3r_Start : gc_start_structp->version=513, media_operational_mode=EmbeddedMedia
3078355648 decoder.cpp:72 ! 0 ! gc_h3r:Sharon - Decoder Send socket binding on IP=0x100007f : Port=51335
3078355648 decoder.cpp:85 ! 0 ! gc_h3r:Sharon - Listening on IP=0x7f000001 : Port=34760
3078355648 virt_board.cpp:181 ! 0 ! gc_h3r:MIME pool allocation complete size 0x5c4 number 5.
3078355648 virt_board.cpp:209 ! 0 ! gc_h3r:NSControlData pool allocation complete size 0x1000 number 0.
3078355648 sm.cpp:2458 ! 0 ! gc_h3r: Allocation: Board 1 - Sharon = 1 , H323 = 0 , SIP = 1
3078355648 sm.cpp:2465 ! 0 ! gc_h3r: Allocation: CRNs = 1 , Extension buffers = 16
3078355648 gc_h3r ERR1 sm.cpp:1195 ! 0 ! << genDefaultSigalStartParams Setting Max Subscription to 0.
3078355648 gc_h3r ERR1 sm.cpp:698 ! 0 ! << genSigalStartParams: MAx Subscription = 0: .
3078355648 sm.cpp:1091 ! 0 ! gc_h3r:Initialization: delimiter: ',', IP_VIRTBOARD version: 0x114
3078355648 sm.cpp:486 ! 0 ! gc_h3r:Initialization: SIP Board 0: IPv4 transport address: 192.168.0.86:5060
3078355648 ssm.cpp:488 ! 0 ! gc_h3r:Initialization: SIP Board 0: IPv6 transport address: :::5060
3078355648 sm.cpp:490 ! 0 ! gc_h3r:Initialization: SIP Board 0: sip_msginfo_mask: 0x3, sup_serv_mask: 0x0, dynamic_outbound_proxy_enable: 0x0
86293360 sip_decoder.cpp:63 ! 0 ! gc_h3r: Board 1 - Listening on IP=0x7f000001 : Port=35052
86293360 sip_encoder.cpp:164 ! 0 ! gc_h3r:Board 1 - Sharon on IP=0x7f000001 : Port=34760
86293360 sip_stack Error ERROR - SOCKET - RvSocketBind(sock=15,addr=192.168.0.86:5060,scopeId=0,range=(nil),useRange=0,errno=98)=-2145371137
86293360 sip_stack Error ERROR - TRANSPORT - TransportUDPOpen (Local Address 0x0xb5d3ab68): can't bind UDP socket -1 (crv=-2145371137,rv=-1:UNKNOWN)
86293360 sip_stack Error ERROR - TRANSPORT - LocalAddressOpen - Failed to open local UDP address 192.168.0.86:5060 (0x0xb5d3ab68). rv=-1
86293360 sip_stack Error ERROR - TRANSPORT - OpenLocalAddressesFromList - Failed to open local UDP address 192.168.0.86:5060 (0x0xb5d3ab68). rv=-1
86293360 sip_stack Error ERROR - TRANSPORT - OpenLocalAddresses - Failed to open UDP addresses (rv=-1)
86293360 sip_stack Error ERROR - TRANSPORT - TransportMgrInitialize - Failed to open local addresses
86293360 sip_stack Error ERROR - TRANSPORT - SipTransportConstruct - Failed - destructing the module
86293360 sip_stack Error ERROR - STACK - StackConstructTransportModule - The transport module couldn't be constructed
86293360 sip_stack Error ERROR - STACK - StackConstructModules - The Transport module couldn't be constructed, rv=-1
86293360 sip_stack Error ERROR - STACK - RvSipStackConstruct - Failed to construct stack modules. The supplied configuration was:
3078355648 gc_h3r ERR1 sm.cpp:2500 ! 0 ! starting Sigals failed
3078355648 gc_h3r ERR1 sm_main.cpp:387 ! 0 ! >> h3r_Start : caught unknown exception while creating Sharon Manager!
3078355648 gc ERR1 gcprod ----- _gp_StartAllCCLibs() - Library libgch3r.so start procedure returned error: -1
3078355648 gc ERR1 gclib ::::> gc_Start() - _gp_StartAllCCLibs() failed:-1
Lines 8 and 9 (ERR1 in gc_h3r) may well be significant.
↧
RE: SIP Device names
As an aside:
License Type: Trial
Expiration Date: 25-oct-2015
License Options: Voice 10, Conferencing 10, Speech Integration 10, RTP_G_711 10, IP_Call_Control 10, Native_RTP 10
License Type: Verification
Expiration Date: Permanent
License Options: RTP_G_711 1, Voice 1, IP_Call_Control 1, Mutlimedia 1
↧
RE: SIP Device names
Hello all
I'm still smashing my head against a brick wall with this one. I have until 25th October to produce a demo. application and so far can't get past the first gc function call.
If no libraries are passed to gc_start () and call:
GC_CCLIB_STATUSALL states;
gc_CCLibStatusEx ("GC_ALL_LIB", &states);
I get:
29-09-2015|12:39:58.73505|getLibraryStates|name: GC_ICAPI_LIB, state: GC_CCLIB_FAILED
29-09-2015|12:39:58.73533|getLibraryStates|name: GC_ISDN_LIB, state: GC_CCLIB_FAILED
29-09-2015|12:39:58.73547|getLibraryStates|name: GC_ANAPI_LIB, state: GC_CCLIB_FAILED
29-09-2015|12:39:58.73560|getLibraryStates|name: GC_PDKRT_LIB, state: GC_CCLIB_FAILED
29-09-2015|12:39:58.73573|getLibraryStates|name: GC_SS7_LIB, state: GC_CCLIB_AVAILABLE
29-09-2015|12:39:58.73586|getLibraryStates|name: GC_DM3CC_LIB, state: GC_CCLIB_AVAILABLE
29-09-2015|12:39:58.73598|getLibraryStates|name: GC_IPM_LIB, state: GC_CCLIB_AVAILABLE
29-09-2015|12:39:58.73611|getLibraryStates|name: GC_H3R_LIB, state: GC_CCLIB_FAILED
29-09-2015|12:39:58.73624|getLibraryStates|name: GC_DIVAISDN_LIB, state: GC_CCLIB_FAILED
29-09-2015|12:39:58.73637|getLibraryStates|name: GC_CUSTOM1_LIB, state: GC_CCLIB_CONFIGURED
Note state of GC_H3R_LIB.
I have attached the RTF output for this exercise (as before, I have trimmed the Timestamp and PID columns for brevity).
↧
RE: SIP Device names
Ok, I suspect some else is binding to port 5060 on the system in this case. As per:
0x7f000001 : Port=34760
86293360 sip_stack Error ERROR - SOCKET - RvSocketBind(sock=15,addr=192.168.0.86:5060,scopeId=0,range=(nil),useRange=0,errno=98)=-2145371137
86293360 sip_stack Error ERROR - TRANSPORT - TransportUDPOpen (Local Address 0x0xb5d3ab68): can't bind UDP socket -1 (crv=-2145371137,rv=-1:UNKNOWN)
This is O/S socket call failure. Check for netstat output to see what else might be using that port on the system.
Jeff
↧
↧
RE: DM/IP SIP and NAT Traversal
No changes would have been done since DMIP boards were EOL'ed many years (maybe 2010?) ago and are no longer supported. Also, 3PCC as mentioned above is only supported on HMP in this case.
Jeff
↧
RE: Dialogic D120 causing gateway module to short out
As I figured the issue with their gateway and media modules crashing was not a Dialogic card issue nor was it my server. One of the media modules does not seem to support the 24 ports connected and crashes.
↧
RE: Dialogic D120 causing gateway module to short out
Interesting -- the D120s have a 600 ohm impedance which is standard for analog handset terminating from the telco line. I wouldn't think it would cause an issue with the switch assuming it is configured to handle that impedance level and density.
↧
RE: SIP Device names
Rebooting and killing a lot of processes has at least got me past the first GlobalCall operation. Thanks.
↧
↧
RE: Dialogic D120 causing gateway module to short out
Sorry - didn't finish my post...dragged away by another customer.
Anywho - it turned out to be an issue with their amphenol cable connected to one of their media modules. Seems that when they use all 24 ports supported by the analog media module (connected to 2 D120's via patch panel) and reboot the server, it causes the media module to crash. If they only use 20 of 24 ports on it all it fine. I can't explain it and since it's not my issue I'll leave it up to them.
↧
Dialogic D120 causing gateway module to short out
I have a customer using an Avaya PBX (Avaya Communications Manager (CM) 6). They have their analog lines connecting to modules in a gateway. They were initially running one D120 without issue. They upgraded to 4 D120's and a total of 44 lines. They are telling me that when the server that houses the D120's is rebooted, it causes the gateway to short out. I've never heard of such a thing and since I'm the vendor, it's my problem. Any thoughts? If you need more info, please ask.
↧
RE: HMP 4.1 Linux Playing Vox File before gc_AnswerCall
I solved the issue .
↧
HMP 4.1 Linux Playing Vox File before gc_AnswerCall
Hi,
I am trying to set up IVR response on my SIP server before calling gc_AnswerCall.
I connected ip devices with voice device and called dx_play and it works if i call gc_AnswerCall function .
If i don't call gc_AnswerCall ,i get 180 RINGING in my softphone and i don' hear any vox even thoug i connected the devices and called dx_play .
I assume this is ok because 180 Ringing message doesn't allow any media to be heard but only plays ringtone localy on calling side .
On my SIP server i changed gc_AcceptCall to answer with 183 Session Progress and now on my softphone i don't hear ringtone but no RTP is received eventhough i connected ip and voice devices and called dx_play .Then after i call gc_AnswerCall IVR is heared .
So my question is is it possible to hear some IVR on HMP without calling gc_AnswerCall ?
If so i would appreciate if anyone can help .
↧
↧
RE: Standard Runtime Library on Windows Server 2012 32bit OK ?
Yes, it should be fine, as long as it built on 32-bit platform. Normally, no rebuild is required.
↧
Standard Runtime Library on Windows Server 2012 32bit OK ?
We use D120JCTLSEW boards.
I need to port from Windows 2008 to Windows 2012 OS
Our product runs on 64bit 2008 from app built on 32bit Windows 2003 OS (32bit SRL lib)
Can i simply run this on 2012, or must i relink my app to a 64bit SRL lib
Many Thanks !
Done
↧
RE: Best way to find the end of voicemail greeting
I also need to be able to leave voicemails when I call someone. We are using Powermedia XMS.
↧