Nokia Push-to-talk


#1

Hello.
We have been using ASterisk as out company PBX for a while. I just recently aquired a Nokia 3230 GSM mobile phone which supports “push-to-talk”, which no GSM operator supports… But, extensve research indicates that “push-to-talk” is actually SIP(!). And I can partly register my mobile phone with asterisk… Password is accepted, but the phone never finishes “turing on push-to-talk”, and a process on the phone “PocServer” dies.

All my searching of asterisk-sites has revealed me nothing.

Asterisk log:
– Registered SIP ‘jm’ at 212.169.96.218 port 17618 expires 3600

Questions:

  • Has anybody else tried this? Did it work?
  • Anybody know any documentation?
  • Other comments? Is it worth looking into?

Marius


#2

Before anyone can comment much further, we really need to see the SIP debug. From the CLI:

*CLI> sip debug
… test it out …
*CLI> sip no debug

And let’s take a look at the output. If you have multiple things registering to the box, then you may want to do:

*CLI> sip debug ip w.x.y.z
(Where w.x.y.z is the IP addy of the phone).

Good luck!

-jbn


#3

www2CLI> sip debug ip 212.169.96.218
SIP Debugging Enabled for IP: 212.169.96.218
www2
CLI>

Sip read:
REGISTER sip:easynet.no SIP/2.0
Route: sip:217.77.32.167;lr
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKTgOUKf1QCReFg
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no
Contact: Jm sip:jm@10.247.255.148;expires=3600
Supported: sec-agree
CSeq: 44546338 REGISTER
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
Max-Forwards: 70
Content-Type: application/sdp
Content-Length: 354

v=0
o=jm 0 0 IN IP4 10.247.255.148
s=logon
c=IN IP4 10.247.255.148
t=0 0
m=audio 49154 RTP/AVP 105
a=maxptime:180
a=ptime:180
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:1500
a=poc-version:1
a=poc-dndmode:off
a=poc-callacceptance:normal
a=poc-urlvr-ind:10

12 headers, 17 lines
Using latest request as basis request
Sending to 10.247.255.148 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKTgOUKf1QCReFg;received=212.16
9.96.218;rport=43310
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no;tag=as6d278e8a
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
CSeq: 44546338 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:jm@217.77.32.167
Content-Length: 0

to 212.169.96.218:43310
Transmitting (NAT):
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKTgOUKf1QCReFg;received=212.16
9.96.218;rport=43310
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no;tag=as6d278e8a
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
CSeq: 44546338 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:jm@217.77.32.167
WWW-Authenticate: Digest realm=“asterisk”, nonce="199bc224"
Content-Length: 0

to 212.169.96.218:43310
Scheduling destruction of call ‘Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq’ in 15000 ms
www2*CLI>

Sip read:
REGISTER sip:easynet.no SIP/2.0
Route: sip:217.77.32.167;lr
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKIZQ5IMmPUNWso
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no
Contact: Jm sip:jm@10.247.255.148;expires=3600
Supported: sec-agree
CSeq: 44546339 REGISTER
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
Max-Forwards: 70
Authorization: digest username=“jm”,realm=“asterisk”,nonce=“199bc224”,uri=“sip:e
asynet.no”,response="19b02096378c9e4e72b0ddec88988faf"
Content-Type: application/sdp
Content-Length: 354

v=0
o=jm 0 0 IN IP4 10.247.255.148
s=logon
c=IN IP4 10.247.255.148
t=0 0
m=audio 49154 RTP/AVP 105
a=maxptime:180
a=ptime:180
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:1500
a=poc-version:1
a=poc-dndmode:off
a=poc-callacceptance:normal
a=poc-urlvr-ind:10

13 headers, 17 lines
Using latest request as basis request
Sending to 10.247.255.148 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKIZQ5IMmPUNWso;received=212.16
9.96.218;rport=43310
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no;tag=as6d278e8a
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
CSeq: 44546339 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact: sip:jm@217.77.32.167
Content-Length: 0

to 212.169.96.218:43310
– Registered SIP ‘jm’ at 212.169.96.218 port 43310 expires 3600
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKIZQ5IMmPUNWso;received=212.16
9.96.218;rport=43310
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no;tag=as6d278e8a
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
CSeq: 44546339 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: sip:jm@10.247.255.148;expires=3600
Date: Tue, 03 May 2005 20:26:08 GMT
Content-Length: 0

to 212.169.96.218:43310
Scheduling destruction of call ‘Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq’ in 15000 ms
www2*CLI>

Sip read:
REGISTER sip:easynet.no SIP/2.0
Route: sip:217.77.32.167;lr
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKIZQ5IMmPUNWso
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no
Contact: Jm sip:jm@10.247.255.148;expires=3600
Supported: sec-agree
CSeq: 44546339 REGISTER
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
Max-Forwards: 70
Authorization: digest username=“jm”,realm=“asterisk”,nonce=“199bc224”,uri=“sip:e
asynet.no”,response="19b02096378c9e4e72b0ddec88988faf"
Content-Type: application/sdp
Content-Length: 354

v=0
o=jm 0 0 IN IP4 10.247.255.148
s=logon
c=IN IP4 10.247.255.148
t=0 0
m=audio 49154 RTP/AVP 105
a=maxptime:180
a=ptime:180
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:1500
a=poc-version:1
a=poc-dndmode:off
a=poc-callacceptance:normal
a=poc-urlvr-ind:10

13 headers, 17 lines
Using latest request as basis request
Sending to 10.247.255.148 : 5060 (NAT)
Transmitting (NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKIZQ5IMmPUNWso;received=212.16
9.96.218;rport=43310
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no;tag=as6d278e8a
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
CSeq: 44546339 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: sip:jm@217.77.32.167;expires=3600
Content-Length: 0

to 212.169.96.218:43310
Transmitting (NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.247.255.148:5060;branch=z9hG4bKIZQ5IMmPUNWso;received=212.16
9.96.218;rport=43310
From: Jm sip:jm@easynet.no;tag=Q9iUKdQBQ5
To: Jm sip:jm@easynet.no;tag=as6d278e8a
Call-ID: Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq
CSeq: 44546339 REGISTER
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Expires: 3600
Contact: sip:jm@10.247.255.148;expires=3600
Date: Tue, 03 May 2005 20:26:09 GMT
Content-Length: 0

to 212.169.96.218:43310
Scheduling destruction of call ‘Vq6UK4g2XB1uC0qkMRvvWOil1ptmKq’ in 15000 ms
www2*CLI> [root@www2 ~]#


#4

Marius, did you get anywhere with this?


#5

No, sorry…
I have not had any more time to look at this.
I am pretty certain it can be done with a minimal amount of hacking asterisk and/or apache.

Marius