[Rtcomm] [Rtcomm] [Bug 6930] Incomming CallerID on VoIP account fails to associate with existing contact

[From: bugzilla-daemon at maemo.org]

https://bugs.maemo.org/show_bug.cgi?id=6930


spamsuxx at gmail.com changed:

           What    |Removed                     |Added
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                 CC|                            |spamsuxx at gmail.com




------- Comment #7 from spamsuxx at gmail.com  2009-12-28 09:45 GMT+3 -------
(In reply to comment #6)
> Hi, I have same problem. So, n900 does not display callerid name but accept
> callerid number.
> With following code on e75 i receiving name in format:
> +123456 at my-contact-sip-server and number just as +123456; while on n900 i
> receiving:
> +123456 at my-own-asterisk-domain-or-peer-name
> 
> exten =>
> _[+-_*#0-9a-zA-Z].,1,Set(CALLERID(num)=${CUT(CUT(SIP_HEADER(From),@,1),:,2)})
> exten => _[+-_*#0-9a-zA-Z]./_00Z!,2,Set(CALLERID(all)=+${CALLERID(num):2})
> exten => _[+-_*#0-9a-zA-Z]./_011Z!,2,Set(CALLERID(all)=+${CALLERID(num):3})
> exten => _[+-_*#0-9a-zA-Z]./_810Z!,2,Set(CALLERID(all)=+${CALLERID(num):3})
> exten =>
> _[+-_*#0-9a-zA-Z].,n,Set(CALLERID(name)=${CALLERID(num)}@${CUT(CUT(CUT(SIP_HEADER(From),@,2),>,1),\;,1)})
> exten => _[+-_*#0-9a-zA-Z].,n,Goto(dialin,${EXTEN},start)
> 
> It would be much better to enable in SIP advanced settings on n900 the way to
> setup callerid and let match it against phonebook.
> 
> Another SIP stack bugs:
> choppy audio on wifi, sometimes on 3G;
> does not support for amr codec (i am using amr on e75 and it is great when i
> call over 3G);
> no early-audio (i do not hear operator's messages and have to do trick: exten
> => _X!,1,Playback(silence/1,noanswer), exten => _X!,n,Dial(....) and with this
> audio is very choppy from first second;
> sip does not work under openvpn (to be correct, it uses ip address from ISP,
> not openvpn) - need option to specify bindings priority for different
> interfaces (tap, wlan, gprs).
> 

thanks for the info.

You might check my other bug for your second problem, it should be more
related. Otherwise open a new bug for your second problem.
--> Bug #6936 (Caller voice gets minced after some time when taking call via
SIP)

Cheers
Bjoern


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