What ports does the 7050 use in a sipaddr-to-sipaddr conversation?

zooster

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I've found that I'm able to make sip address-to-sip address call only if the 7050 is not behind the router, then I have to open on teh router ports used in that kind of call.
What ports do I have to open on the router for make a sip-to-sip call(no fixed lines)?
 
tcp 5060
udp 7078-7087

for SIP to work in general... no matter what kind of endpoint you call.
 
Ok, thanks it works great now!
But can you tell my why in the Anrufliste I dont have any sipaddress but only numbers? When I get an incoming call there is written unbekannt(unknown I think), is there anything wrong?
 
It's a feature, not a bug ;-) If you log onto the FBF using telnet and look at the voipd output, you'll find the SIP addresses there as well... AVM just apparently did not care to log that information... besides, some calls originate in the fixed-line space, so it's actually useful to have the caller id listed in the Anrufliste... Signalling a SIP address as caller id is a bit of a problem with classical fones ;-) However, I agree with you, finding that at least in the Anrufliste would be helpful... maybe somebody has a mod? ;-)

Cheers,
--gandalf.
 
Mod neeeded :) Plz tell me wht mode I have to upload, I dont understand german :)
 
Hmm... that's no mod I've seen so far. It would be somehow the equivalent of the callmessage script for telefon, only for voipd to catch certain log output mentioning call setup and termination with respective SIP addresses.

--gandalf.
 
But if I buy an isdn cordless with an alphabetic display, will be the sip address anyway shown??
 
Nope. You'll see the caller's numeric and alphanumeric id (if present) on the display. The FBF-internal log will include the address, of course.
 
Then this statement is wrong??
 
Good question... I've never seen a SIP address like [email protected] or such signalled on my ISDN phone... neither as a numeric id, nor as the alphanumeric data.

--gandalf.
 
Hi Zooster,
in that statement I'm speaking about a possibility. As some ISDN phones can display alphanumeric IDs, it is (at least technically) possible to display a sip address. The FBF (or any other adapter) just has to generate the correct signalling sequence. That would be a good wish on the feature list for the next firmware update.
 
Hi,

The FBF forwards the local part of the sip-uri as calling party number to ISDN phones. I think this makes sense as landline calls that terminate as VoIP calls also have an originating uri, but the caller cannot be reached with this uri (eg <sip:[email protected]> if a localized number from gmx is called via landline).
If the "display name" (= the string that precedes the uri in the "From:" part of the INVITE sequence) is "anonymous" then the calling party number is signalled as unknown.
The list of calls in the web interface however shows these calling party numbers only if they are purely numerical.

HTH
radio_junkie
 
Ok, thanks.
 
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