How to dial a sip address wiht the 7050?

zooster

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I have 5 voip accounts, but I can't find where and how I can dial a sip address like [email protected].
 
To do this, you have to use a little workaround, as ISDN- and PSTN-phones are usually not able to dial sip addresses.
Step 1: Create a quick-dial that dials the sip address. (Telefonie -> Kurzwahlen -> Neue Kurzwahl)
Step 2: On your phone, dial that quick-dial.
Step 3: Happy phoning.
 
But I don't understand how to dial that: for example I've set
Kurzwahl-01 Vanity-nickname [email protected]
How, what is suposed to be dialed?
If I try to dial 01 I have line busy, or should I dial something before? Shall I choose the provider or what?
Thanks.
 
Got your problem. Before dialing the 01 or the vanity code, you have to tell your FBF what you will dial next. :) if you want to dial the 01, punch **701# - if you want to dial a vanity, punch **8vanity#
 
Not working, after I dial(wihtout choosing any sip provider first) **701# I get line busy...
But don't I have to tell to the box what account I'm gonna use to make that sip call?? How?
 
Zuletzt bearbeitet:
No, you don't have to tell the box which account to use. The FBF uses the first account as caller ID. You can choose the account as usual with *12x# and dial **701# afterwards. Just replace x with the account's number as found in Telefonie -> Internettelefonie. Ijust checked it with my FBF and a softphone; it works fine.

Please have a look into the logfiles of your FBF (System -> Ereignisse) and post a screenshot of them, right after you placed such a call. Don't forget to black out any account- and server names.
 
Ok, I can't. I tried to dial **701# as well as *125#**701#, but all I get is a busy line... Here are screenshots of my configurations:
 

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That 503 error message is to be read as "Service unavailable -- Indicates that the server or gateway is unable to process the request due to an overload or maintenance problem." The FBF communicates the 503 error as a busy line on its FON ports.
Can you place outgoing calls to regular phones when chosing your *125# account?
 
By that account I can't call a regular fixed number, but I can call sip addresses. I use it in X-lite and it works well.
 
I just set up such a similar account in my FBF and dialing *126#**799# (99 resolves to a sip address as described above) works just fine. That's odd - and I don't have any more ideas on this topic, sorry.
 
I've just deleted the stun server of voipuser.org, and now it works fine! Bohhhh, how strange. Thank you very much though :)
 
OK ... are you using your FBF behind any router or is it your border router to the internet? If it's the border router, you don't need any STUN-server :) because the FBF already knows its public IP.
 
I'm using it behind a router... boh. :)
 
So you probably DO need STUN. Mabye it works if you use one of these servers?
stun.sipgate.net:10000
stun.gmx.net:3478

One other idea is to use your FBF as your border router.
 
Boh, now it works... I'll leave this working config atm :p
 
zooster schrieb:
Boh, now it works... I'll leave this working config atm :p

Just in case the problem returns:

Place "sip:" in front of the quick-dial; that tells your FB to call the number entered by using the SIP server after the @:

01 = sip:[email protected]


Raffke.
 
Just in case you are still interested. You used the example of sipphone which has numerical sip addresses. Thus you do not have to set up like above. You simple make a dial rule for this number to use your siphone account and the call will go through.

Remember though that this will not work where you have asked the box to automatically add country code or region code to the front of numbers (ie if you use a VoIp provider for both PSTN calls and SIP address calls)
 

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