[Asterisk-Dev] Anyone tried Asterisk with Nokia PTT technology.

[Alex]

PTT is so called "PushToTalk" it is a kind of virtual 2-way walkie talkies
inmplemented with VoIP over GPRS. Nokia insists it is using "standarts
based" SIP technlogy. I have Googled, but found nothing.

When my phone (6230i) is trying to register to * it is always getting
"401 Unautorized".

Anyone got any experiences with PTT?
It would be nice if * will support it. I guess there are some
extensions that will be needed to handle PTT correctly.

This is "sip debug" of phone trying to register with *


Cheers, Alex.

------- cut --------

Sip read:
REGISTER sip:pbx.intercaf.ru SIP/2.0
Via: SIP/2.0/UDP 213.87.62.53;branch=z9hG4bK5v96bu
To: 210 <sip:210 at pbx.intercaf.ru>
From: 210 <sip:210 at pbx.intercaf.ru>;tag=hmgid4
Max-Forwards: 70
Call-ID: 6285372088210777 at 213.87.62.53
CSeq: 2 REGISTER
Contact: 210 <sip:210 at 213.87.62.53>
Expires: 3600
Content-Type: application/sdp
Content-Length: 350

v=0
o=210 0 0 IN IP4 213.87.62.53
s=logon
c=IN IP4 213.87.62.53
t=0 0
m=audio 49154 RTP/AVP 105
a=ptime:160
a=maxptime:160
a=rtpmap:105 AMR/8000
a=fmtp:105 mode-set=1; octet-align=0
m=control 49154 RTP/POC 104
a=poc-rtpmode:STP
a=poc-rtp-heartbeat:1200
a=poc-version:1
a=poc-dndmode:off
a=poc-callacceptance:accept
a=poc-urlvr-ind:0

11 headers, 17 lines
Using latest request as basis request
Sending to 213.87.62.53 : 5060 (non-NAT)
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 213.87.62.53;branch=z9hG4bK5v96bu
From: 210 <sip:210 at pbx.intercaf.ru>;tag=hmgid4
To: 210 <sip:210 at pbx.intercaf.ru>;tag=as13690c72
Call-ID: 6285372088210777 at 213.87.62.53
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:210 at 83.102.221.70>
Content-Length: 0


 to 213.87.62.53:5060
Transmitting (no NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 213.87.62.53;branch=z9hG4bK5v96bu
From: 210 <sip:210 at pbx.intercaf.ru>;tag=hmgid4
To: 210 <sip:210 at pbx.intercaf.ru>;tag=as13690c72
Call-ID: 6285372088210777 at 213.87.62.53
CSeq: 2 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: <sip:210 at 83.102.221.70>
WWW-Authenticate: Digest realm="asterisk", nonce="0d9708eb"
Content-Length: 0


 to 213.87.62.53:5060
Scheduling destruction of call '6285372088210777 at 213.87.62.53' in 15000 ms
Destroying call '6285372088210777 at 213.87.62.53'
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