Nokia N91 SIP Client Setup

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[edit] Origin

After Purchasing a Nokia N91 I wanted to setup the SIP client through my home Asterisk Server. I read some online tutorials abut none either covered the N91 specifically[1]

or they were not very descriptive[2]

.

NOTE: This does not end with a working setup, As it stands there is no way for the Nokia N91 to work with a SIP server yet. The SIP support appears to be broken in it's handling of the "content-type" of the SIP packets. I believe this is because the Push to Talk system uses SIP

[edit] Configuring Asterisk

This assumes you know a little bit about asterisk and that you have a already running asterisk box (basically how I started)

[edit] sip.conf

[general]
context=bogon-calls
realm=magi

[201]
type=friend
callerid=("Kosh Mobile <201>")
username=201
host=dynamic
secret=secretpass
dtmfmode=rfc2833
insecure=very
canreinvite=yes
nat=yes
qualify=no
context=mobile-outbound
mailbox=200
disallow=all
allow=alaw

These are the important lines in the sip.conf, I will explain the major ones to you.

[edit] [General]

context 
bogon-calls This is the context for calls that attempt to arrive via sip without authenticating.
realm 
magi This is the realm for authenticating, IMPORTANT: This is required for configuring the phone

[edit] [Extension] (201)

username 
201 The username for the SIP connection
secret 
secretpass The password for the SIP connection
context 
mobile-outbound The context for calls made from the mobile.
disallow 
all Codecs usable by the phone, we are disabling all of them then enabling a specific codec
allow 
alaw The codec we are setting for the phone to use

[edit] extensions.conf

[bogon-calls]
exten => _.,1,Congestion
exten => s,1,Congestion

[mobile-except]
include => ext-sip
exten => _X.,1,Dial(IAX2/122220/${EXTEN})

[ext-sip]
exten => 200,1,Macro(stdexten,200,SIP/200)
exten => 201,1,Macro(stdexten,201,SIP/201)

This is a VERY VERY cutdown version of my extensions.conf, down to the bare essentials for this tutorial.

[edit] [bogon-calls]

This is where incoming calls with no authentication information get thrown.

exten => _.,1,Congestion 
Any incoming number gets Congestion
exten => s,1,Congestion 
Any incoming call gets Congestion.

[edit] [mobile-except]

include => ext-sip 
Include my extension list, so handsets can call each other
exten => _X.,1,Dial(IAX2/122220/${EXTEN}) 
Any other number that isn't picked up by the extension include is dialed via IAX to my phone provider.

[edit] [ext-sip] (Include)

exten => 200,1,Macro(stdexten,200,SIP/200) 
IF you search google:macro-stdexten you can find an example of this macro.

[edit] Nokia N91 Setup

On the Handset you will need to setup access to your Wireless Access Point (Assuming you want to use WiFi, In Australia, you'd be crazy to want to do VoIP over carrier internet)

[edit] Wireless Setup

On the phone

  1. Menu
  2. Connect.
  3. Conn. mgr.
  4. Select Availab WLAN
  5. Move the selection to the WLAN of your choice, then press Options
  6. Select Define access point
  7. Press Yes
  8. Select Options
  9. Select Exit

Congratulations, your WLAN is now setup !

[edit] SIP Setup

Back to the Phone again, These settings are paraphrased from newlc.com[2]

  1. Menu
  2. Tools
  3. Settings
  4. Connection
  5. SIP settings

[edit] SIP settings

Profile mame : Home Asterisk Just a simple name to describe the connection
Service profile : IETF what SIP you are using, asterisk uses IETF
Default access point : HomeWiFi The default access point to use, whatever you setup in the last step.
Public user name : sip:201@10.11.81.21 the address other phones would use to call you, basically your extension @ your SIP server
Use compression : No Compression, Select No
Registration : Always on This option tells the phone to stay registered whenever it can
Use security : No disable security

[edit] Proxy Server

Proxy Server Address : sip:10.11.81.21 The hostname or IP address of your SIP server
Realm : magi IMPORTANT: this must be the same as the realm specified in your sip.conf or your authentication WILL fail
User name : 201 The user name required to connect to your SIP server
Password : **** The password required to connect, this is the "secret" listed in the sip.conf
Allow loose routing : Yes
Transport type : UDP
Port : 5060

[edit] Registrar Server

Registrar serv. addr. : sip:10.11.81.21 The hostname or IP address of your SIP server
Realm : magi IMPORTANT: this must be the same as the realm specified in your sip.conf or your authentication WILL fail
User name : 201 The user name required to connect to your SIP server
Password : **** The password required to connect, this is the "secret" listed in the sip.conf
Transport type : UDP
Port : 5060

[edit] The End

This is as far as I could get to, after this I was stuck at #SIP Problem

[edit] Troubleshooting

Errors and their notes

[edit] Settings keeps crashing when I try to edit my SIP Config

I hated this error, you might need to go play a fighting game on a console to get your speed up to do this.

Now you could probably do this if you just turned off your WLAN first, but cmon, where is the challenge.

This needs to be done really quick, what I assume is happening is when the registration is failing some option you have set bones the parsing, no idea how, but here is the fix.

On the Phone again

  1. Go into the Menu
  2. Select Tools

Here is where you need the speed, from here to Yes needs to be done before it connects

  1. Select Settings
  2. Select Connection
  3. Select SIP settings
  4. Select the SIP config
  5. Press C (clear on the keypad)
  6. Select Yes

Tada, now create a new connection from scratch, but it shouldn't crash out when you try to edit it.

[edit] SIP Problem

Couldn't get past this error, it seems pretty common[3] [4]

Sip read:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.11.81.21:5060;branch=z9hG4bK0ecefbf9;rport
To: <sip:201@192.168.14.205>
From: "Handset" <sip:200@10.11.81.21>;tag=as42488490
Call-ID: 6df150b328b6d2316594128c10220f55@10.11.81.21
CSeq: 102 INVITE
Content-Length: 0


7 headers, 0 linesebug

Sip read:
SIP/2.0 415 Unsupported Media Type
Via: SIP/2.0/UDP 10.11.81.21:5060;branch=z9hG4bK0ecefbf9;rport
To: <sip:201@192.168.14.205>;tag=iod3oqnlehhc6smi9n5l
From: "Handset" <sip:200@10.11.81.21>;tag=as42488490
Call-ID: 6df150b328b6d2316594128c10220f55@10.11.81.21
CSeq: 102 INVITE
Accept: application/poc+xml
Content-Length: 0

8 headers, 0 linesebug
    -- Got SIP response 415 "Unsupported Media Type" back from 192.168.14.205

[edit] References

  1. . How to connect your Nokia E61 to your Asterisk phone server SIP VoIP
  2. . 2.0 2.1 Using SIP with Nokia Series60 and Asterisk
  3. . 415 Error - N91 and VOIP through Asterisk
  4. . 415 Error - Nokia SIP plugin 3.0 - 415 Unsupported Media Type

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