Call number transmission with sipgate & telephone system - VoIP General, VoIP News, VoIP Providers, VoIP hardware, VoIP software, VoIP Technology

Hello together.

With sipgate gives it the possibility of letting the outgoing Telefonummer of the terminal set. Unfortunately does not function with me.

I operate a Lancom 1722 VoiP gateway at a AGFEO AS 40 P telephone system.

Gateway is attached at an external S0 of the telephone system (as additional office). The VoIP-S0 is as multi-device connection configures and has a MSN in the format 4923xxyyyyyy.

Now I can determine at my system telephone (ST30) over a key whether over VoiP or ISDN is to be telephoned. If I lead a ISDN discussion, the call number is not conveyed, with VoIP.

Unfortunately I do not continue to know any longer, since I cannot determine exactly, whether the error arises during the call number transmission, in the configuration of the Routers or the telephone lies.

Here thus my question: Operate someone (successfully ) a similar system and can give itself a reference?
Where is those set going off call number (telephone - telephone system - rout)?

Thanks for your answers.

HP: Also with a test with a similar telephone the call number was not conveyed.

Erstmal: The MSN 4923… surely the wrong format has. Here may only the call number, without any ONKz'en or international preselections registered its, at least in as 40.

Then: The “call number-setting terminal” might be not the plant, but that Lancom routing. Above all there you should look for the problem.

And finally: If it should be always the same number, which you want to convey, then you can at the Website of Sipgate with your attitudes also purposefully “set number 4923xxxxxxx” to register. Then it is complete sausage, which rauswurstelt that and which terminal at numbers '.

Greeting,

Hendrik

Hendrik thanks for your answer.

It should not be always the same call number, which is conveyed. To as40 has I 35 telephones, which should convey their “own” call number in each case. If I stop the outgoing call number to the sipgate Website, that funktioniet perfectly, but evenly with only one call number.

The MSN in as40 can I freely adjust, all the same whether with land before whale or without or only in two digits, it makes unfortunately during the transmission obviously no difference.

That rout gives the following message raus:

...
Call: 492xxyyyyy@intern (492xxyyyyy) -> WIZ_SIPGATE: 02aabbbbb@sipgate.de (bells)
...

Nevertheless Sip packing trace the following results in:

[SIP packing] 2009/03/18 14:30: 48,570 [PACK]:
Sending datagram with length 899 from 79.140.180.68: 17040 tons of 217.10.79.9: 5060
INVITE sip: 023xxyyyyyy@sipgate.de SIP/2.0 \ r \ n
Via: SIP/2.0/UDP 79.140.180.68: 17040; branch=z9hG4bK-0e31e176-fb04f12a \ r \ n
From: “anonymous”< sip: 168xxxx@sipgate.de; user=phone>; day =-1416901190--1842750732 \ r \ n
Ton: <sip: 023xxyyyyyy@sipgate.de; user=phone> \ r \ n
Call ID: 2695882267@00a05712edd7 \ r \ n
...

Where does the call ID come? No notion.

The call number remains suppressed.

Greeting dock.

Quotation:

Quotation of kai123

To as40 has I 35 telephones, which should convey their “own” call number in each case.

Again to understand: Under normal conditions you have (bundled) a ISDN Anlagenanschluss, i.e. with trunk number and direct-inward dialings, correctly?

And: Which firmware is on your AS 40?

Quotation:

Quotation of kai123

The MSN in as40 can I freely adjust, all the same whether with land before whale or without or only in two digits, it makes unfortunately during the transmission obviously no difference.

Hmm. Do you really register MSNs? Or DDIs?

Greeting,

Hendrik

Quotation:

Under normal conditions you have (bundled) a ISDN Anlagenanschluss, i.e. with trunk number and direct-inward dialings, correctly?

I have a plant connection with trunk number and direct-inward dialings, correctly.

Quotation:

Which firmware is on your AS 40?

Firmware of as40 is 6,2, completely again up-played.

Quotation:

Do you really register MSNs? Or DDIs?

I defined the S0, at which that is attached Lancom routing, as multi-device connection, therefore I register MSN's.

Greeting dock.

Everything clearly.

Did you solve how exactly with the “Routing” for your 35 telephones? Are the telephones for outgoing connections so adjusted that that represents a Lancom routing the preferential line? Dropbake on the carrion with lane-occupied? (That might rather often happen, depending upon number of outgoing connections….) Or each caller before each discussion must select the Router-S0 by hand?

Independently of it: For Sipgate that is Lancom routing the actual terminal that “the call number sets”. For the moment can't I say to you, whether the Lancom is able, the outgoing different MSNs of the individual participants (all 35??) thus to convert that that is set off correctly TIE-CLIP at Sipgate. Exactly there the problem will be supposed (and to be solved not to probably also be able).

Which Sipgate account do you have actually? Basic or plus?

Greeting,

Hendrik

Thanks only times for your assistance.

Quotation:

Did you solve how exactly with the “Routing” for your 35 telephones?

In the telephone system I can separately adjust the preferential S0 for each individual telephone.

Quotation:

Are the telephones for outgoing connections so adjusted that that represents a Lancom routing the preferential line? Dropbake on the carrion with lane-occupied? (That might rather often happen, depending upon number of outgoing connections….)

Dropbake etc. leave I now times outside forwards.

Quotation:

… whether the Lancom is able, the outgoing different MSNs of the individual participants (all 35??) thus to convert that that is set off correctly TIE-CLIP at Sipgate.

The Lancom recognizes the individual MSN's; these are transferred by the telephone system and to be able into rout all to be administered individually and configured.

Quotation:

Or does each caller before each discussion have to select the Router-S0 by hand?

Is for the moment only so to the test operation. Even if I change the S0 over firmly in the telephone system, nothing changes in the result.

Quotation:

Which Sipgate account do you have actually? Basic or plus?

Basic. Should no difference make.

Which variable (in SIP minutes) has to be handed over for the call number transmission into which format? Perhaps we continue to come such a piece….

Greeting dock.

Quotation:

Quotation of kai123

In the telephone system I can separately adjust the preferential S0 for each individual telephone.

Ah… thank you. With Agfeo plants am I well trust, therefore we assume in principle that I am informed about the possibilities.

Quotation:

Quotation of kai123

The Lancom recognizes the individual MSN's; these are transferred by the telephone system and to be able into rout all to be administered individually and configured.

That is, you registered and convey really 35 MSNs these to the Lancom?

Quotation:

Quotation of kai123

Basic. Should no difference make.

I am not completely safe.

Quotation:

Quotation of kai123

Which variable (in SIP minutes) has to be handed over for the call number transmission into which format? Perhaps we continue to come such a piece….

Somehow the Lancom, in each case different numbers would have to be persuaded on the Sipgate account to routes. Whether it can do that, extracts itself from my knowledge.

Greeting,

Hendrik

Greeting dock. [/quote]