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Forum Post: RE: HMP3.0 SU372 defect processing "record-route"

So just to be on the same page about this one. There is "Record-Route:" in 200 OK reply... with 1 record According to RFC any content should be copied to outgound ACK. In our case should be "Route:"...

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Forum Post: RE: How to configure (STP) to Connect between two (SPs).

Hi, I am trying to do exactly the same thing as mentioned in your post.  I need to do it as a part of my project. I wanted to you if you found a way to successfully send a message from one SP to...

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Forum Post: RE: How to configure (STP) to Connect between two (SPs).

Hi ukernie, Can you please suggest some application that can help my central node behave like an STP?

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Forum Post: Not able to get calling number

A voice call has been placed from line 1 to line 2(loop back)of same Diva Board(Diva Analog 4 PCI v1 SN 40373). When I try to get the calling number on the termination side(Calliee), I am not getting...

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Forum Post: RE: How to configure (STP) to Connect between two (SPs).

Dialogic sell an STP product based on their stack - so that's probably what you need. We use it and find it to be very good - flexible, reliable and scalable....

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Forum Post: RE: How to configure (STP) to Connect between two (SPs).

Thank you Mikebromwich. So it is a software only solution? And I can install it on my virtual machine?

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Forum Post: RE: How to configure (STP) to Connect between two (SPs).

We have the hardware-based version. There is a VM available too - you'll need to talk to your local Dialogic reseller for pricing etc.

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Forum Post: RE: Not able to get calling number

Am I missing some thing above? Can any one please let me know?

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Forum Post: SIGTRAN - Linkset Priority

Hi, in case of traditional TDM-based SS7 assigning linkset priorities can be ensured that the shortest path is taken by a message. Do we have similar solution (i.e. configuration entries) for NA SS...

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Forum Post: Subscriber Data parameter in Insert Subscriber Data

On Page 55 of MAP Programmer's manual , the parameters of INSERT SUBSCRIBER DATA request are given. Among them, there is a parameter named ' Subscriber Data '. But then there is no explanation of this...

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Forum Post: Incoming G722 call "not acceptable here"(488) despite license

Hello, having trouble with the G722 codec in a software only SIP solution. We have a six channel license which includes G772. Yet I cannot answer a call that offers the G722 codec but no G711 codecs....

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Forum Post: RE: Incoming G722 call "not acceptable here"(488) despite license

Sorry for the typos, I always meant 722, not 772 :)

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Forum Post: RE: Incoming G722 call "not acceptable here"(488) despite license

More info: rhe GCEV_REQ_MODIFY_CALL has nothing to do with it though, I was misreading my own log. My only actions are gc_AcceptCall and gc_AnswerCall and handling the corresponding events.

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Forum Post: Busy signal

First please forgive that I know nothing about the Dialogic boards I am working with.  We have an old (runs on Windows XP...) application that no longer dials out like it should. Nobody is quite sure...

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Forum Post: RE: Incoming G722 call "not acceptable here"(488) despite license

Is this for windows or Linux? Also note support for HD codecs (AMR / GSM / G722 ) are only available when running the application is 3PCC mode in this case. Is you application running in 3PCC mode?...

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Forum Post: RE: Incoming G722 call "not acceptable here"(488) despite license

Thanks Jeff, learned as much in the meantime from a support call. I took a look at the documentation about 3cc mode and as I understand it, my application would act as a kind of mediator handling the...

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Forum Post: RE: Not able to get calling number

How did you configure CLIP for the lines in the Diva Configuration Manager? Caller ID (CLIP) Caller ID (CLIP) is a property of the analog line. With Caller ID (CLIP) you enable phone number detection....

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Forum Post: RE: HMP3.0 SU372 defect processing "record-route"

If anyone interested this issue fixed in SU375

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Forum Post: HMP linux Invite request line Number

Hi, Sometimes number in To: field and number in request line are not the same so i was wandering Is there any API i can call to get number from request Line in INVITE ? Thanks

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Forum Post: RE: Not able to get calling number

Thanks for the response. In our analog line we don’t have CLIP configuration. But we made change on line “Use Caller ID(CID)” to 'yes'. After that we are able to receive calling number on termination...

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