View Full Version : N900 SIP Client with Asterisk
Sofia-SIP throws "Internal Server Error"
Phone -> Accounts -> SIP
Username: 3450@192.168.2.92
Password: xxxxxxxxx
Advanced Settings:
User for telephone numbers: Checked
User name: 3450
Transport: Auto
Outbound proxy: 192.168.2.92
Port: 5060
The rest: Default
The phone registers fine with asterisk. However, when trying to make a call from another SIP phone to the N900 I get:
SIP Phone N900
------------------------
Invite with media description =>
100 Trying <=
180 Ringing <=
500 Internal Server Error <=
I have played with the settings like crazy, but nothing seems to help. Any other SIP phone works fine with my Asterisk (Cisco 9640, X-lite ....)
Did anybody else succeed in using the N900 with Asterisk?
Cheers,
// Jonas
Could be related to this:
https://bugs.maemo.org/show_bug.cgi?id=5342
https://bugs.maemo.org/show_bug.cgi?id=4259
https://sourceforge.net/tracker/?func=detail&atid=756076&aid=2412241&group_id=143636
I have built asterisk-1.6.1.11 .debs and have asterisk *running* on the phone fine. :)
Good luck,
-Jeff
Thanks Jeff,
I haven't tried to actually run Asterisk on the N900. Seems like a cool thing to do though (I use a SheevaPlug for Asterisk 1.6).
The links you have provided seems to be related to the register phase, which actually works for me. My problem is that the N900 doesn't return an OK with session description like is should after the RINGING. Instead it returns the 500 "Internal Server Error". I am thinking this has to do with some bug in the Sofia-SIP stack. However, if somebody else has successfully connected to Asterisk I guess the problem lies elsewhere.
Anyhow, thanks a lot for trying to help out!
// Jonas
Could the package be made available please? :D
bbrindle
2009-12-06, 22:40
Im using my N900 with an asterisk box. Can make and receive calls. Only difference in my config is my proxy is blank.
Dumb question but you have the phone setup in extensions.conf right? Im sure you do but had to ask.
bbrindle,
Great news that you can use it with Asterisk. Then there is some hope. To answer your question, yes the extension is well set up in extensions.conf. In fact, I have tested the same extension with a Cisco 7940 and X-Lite and they both work.
My problem is really visible in the SIP communication where you can see that the sip-sofia stack is sending an Internal Server Error. I know that this can be because of a number of reasons though.
However, with your set up working I want to focus on the asterisk side. Do you mind posting your sip.conf for your N900 working extension?
Mine non-working is
[3450]
username=3450
type=friend
secret=xxxxxxxxx
qualify=500
nat=no
host=dynamic
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=internal
canreinvite=no
callerid=Jonas HP <3450>
Also my general section is
[general]
canreinvite=no
; Include registers
#include "sip_register.conf"
dtmfmode=rfc2833
disallow=all
allow=ulaw
context=trunk ; Default context for incoming calls
allowoverlap=no
bindport=5060
bindaddr=0.0.0.0
srvlookup=yes
limitonpeers = yes
externip = xx.xxx.xxx.xxx
localnet=192.168.0.0/255.255.0.0
nat=yes
Thanks,
// Jonas
bbrindle
2009-12-06, 23:00
Here's my config:
Settings on the N900:
User name: 5005@192.168.0.82
Password: [secret]
Enabled is checked
Advanced:
Use for telephone numbers is checked
User Name: Not set
Transport: Auto
Outbound Prody: Not set
Port 5060
Discover public address disabled
Autodected STUN disabled
Asterisk configuration:
SIP.CONF
[5005]
type=friend ; This device takes and makes calls
host=dynamic ; This host is not on the same IP addr every time
username=Nokia ; Not really used
secret=*****
nat=no ; nat=yes if this phone is behind a NAT box or firewall
context=home ; My internal context for phones
insecure=very ;bypass authentication
qualify=30000
EXTENSIONS.CONF:
exten => 7005,1,Dial(SIP/7005,25) ;
exten => 7005,2,Hangup ; and then hangup.
bbrindle
2009-12-06, 23:15
Left off my general. Here it is.
[general]
useragent=SIP
externip=xxx.xxx.xxx.xxx
localnet=192.168.0.0/255.255.255.0
context = ********
bindport=5060
callerid=No CallID
disallow=all
allow=gsm
allow=ulaw
allow=alaw
allow=iLBC
maxexpirey=180
defaultexpirey=160
tos=reliability
nat=no ; Rely on the fixup protocols to handle the sip xlateion
qualify=yes
dtmfmode=rfc2833
canreinvite=no
realm=asterisk
Okay,
I made sure I had the same parameters in my configuration (commented out the ones that I had extra).
The following is visible in the Asterisk cli when the call from the other SIP phone comes in to the N900 (3450)
-- Called 3450
-- SIP/3450-00462198 is ringing
-- Got SIP response 500 "Internal Server Error" back from 192.168.2.94
-- SIP/3450-00462198 is circuit-busy
== Everyone is busy/congested at this time (1:0/1/0)
I still get the "Internal Server Error". I can't help to thing that it could be a SIP option in the general section that is messing it up (something that sip-sofia doesn't understand). What does your general section look like?
Also, what version of Asterisk are you running?
Cheers,
// Jonas
Still not working. Man, what is the problem....
What asterisk did you run? I am on 1.6.1.6
bbrindle
2009-12-07, 14:13
Hmm. I'm running a much older release of Asterisk than you. (1.2.18).
From the Asterisk CLI do a sip debug and see if there's any other information in the messages coming from the N900. Also try dialing your N900 from the CLI and see if that works. (Rule out any issues between sip endpoints.)
ehsjoar:
I noticed in your settings that you didn't have a realm set, did you also add that when copying bbrindle's settings? Make sure you try realm=asterisk.
i have another problem with sip. if im in wlan i can connect to my asterisk server.
if im in another wlan or umts, i installed openvpn to get a home ip adress.
but the buildin sip client seems to only try to connect over umts directly,
ssh or pinging works on both ends of openvpn
FISHBoY,
I read somewhere that realm will default to Asterisk if not set. I tried setting it now though and it didn't change anything. I did open a ticket for this, even though others have gotten it working. The ticket is on https://bugs.maemo.org/show_bug.cgi?id=6641
smurfy,
That is interesting. So you are running the openvpn server on the Asterisk server and then have openvpn client on your N900?
Also, what version of Asterisk and what firmware version on your N900 are you running?
Openvpn Server on a Linux box with bridging (to get in the same network), ssh, ping etc all works great.
Openvpn Client + Applet from extra-devel i guess. (latest)
Asterisk 1.4.17~dfsg-2ubuntu1 on another server in my network.
N900: 1.2009.42-11 (retail)
sip.conf
[XXXXXX]
type=friend
secret=XXXX
callerid="n900" <1234>
canreinvite=no ;needed this!
nat=yes ; and that
host=dynamic
disallow=all
allow=gsm
allow=ulaw
allow=alaw
extensions.conf:
exten => _X.,1,NoOp(${CALLERID(num)} ${EXTEN})
exten => _X.,2,Dial(CAPI/contr1/${EXTEN})
exten => _X.,3,Hangup
i needed canreinvite=no and nat=yes, to get it to work to talk with another sip client (fritzbox).
my problem never was that i cant connect to the other sip telephone, the problem was that i either can hear the other party or talk to the other party :)
Edit:
as i stated before, i can connect to the asterisk server if i connect to my local wlan. (without openvpn) if i connected to umts or another wlan and the only connection to my network is openvpn i can't connect to the asterisk server.
hm, its somewhat sporadic, i got it working, then disconected the openvpn connection, and reconnected and now it is broken again :)
ok, i enabled sip debugging and something about nat :)
i will try with different sip settings in asterisk.
Smurfy,
So I was never successful connecting the N900 to asterisk at all. I do use openvpn for other soft-phones though. Very handy when you are out traveling and you are on the hotel's LAN.
I had to add a general localnet parameter to the VPN subnet in sip.conf for this to work though.
Could the package be made available please? :D
(He's talking about asterisk running on N900 itself)
I just got permission to upload files to extras-devel this morning, so watch for the package in the next day or so. I'm not sure how long it takes to be available after I upload.
Have fun,
-Jeff
Well, I uploaded the files to be built, but it takes forever to get results. And the build failed. I see why, because asterisk (by default), during the build grabs the sound files from Digium's servers, and the build machine croaked on that (perhaps they don't allow the build machine to go out to 'net, which makes sense). Hrumph.
i think if you could attach the IP dump file that could help to analyze why the sofia-sip stack back "500" to Asterisk. in my opinion, some incorrect parameter of SDP or other reason in INVITE could cause "500" response.
funpig,
I did a tethereal and fired it off to the Maemo developers. They confirm that there is nothing strange in the SIP communication going on. Asterisk sends a REGISTER with a media description, N900 answers TRYING, RINGING and then issues the Internal Server Error.
I did debug the sofia-sip stack. It complains "nua_set_params(): failed: Error Setting SOA Parameters" before it issues the Internal Server Error. Looking in the code, this seems to be right after it has figured out all the media stuff. I have tried on Asterisk 1.4 as well, but it is the same deal there.
It could very well be that there is one media option in the SIP negotiation that i causing this. I don't know what it could be though. For sure there is never a single RTP package sent
I did a tethereal and fired it off to the Maemo developers. They confirm that there is nothing strange in the SIP communication going on. Asterisk sends a REGISTER with a media description, N900 answers TRYING, RINGING and then issues the Internal Server Error.
i assumption you had a mistake for REGISTER. normal, the REGISTER didn't with any media description. and TRYING, RINGING will be sent after N900 received the INVITE.
I did debug the sofia-sip stack. It complains "nua_set_params(): failed: Error Setting SOA Parameters" before it issues the Internal Server Error. Looking in the code, this seems to be right after it has figured out all the media stuff. I have tried on Asterisk 1.4 as well, but it is the same deal there.
i'm not familiar with sofia-sip, but as you said, "SOA" means SDP Offer/Answer Engine Module. you could check http://sofia-sip.sourceforge.net/refdocs/soa/index.html#index . then i agree with you that some parameter is incorrect.
It could very well be that there is one media option in the SIP negotiation that i causing this. I don't know what it could be though. For sure there is never a single RTP package sent
i also hope you could post the detail of INVITE. so, we're could see what's parameters in that INVITE
if im in another wlan or umts, i installed openvpn to get a home ip adress.
but the buildin sip client seems to only try to connect over umts directly,
That would be bug 1860 (https://bugs.maemo.org/show_bug.cgi?id=1860).
funpig:
Session Initiation Protocol
Request-Line: INVITE sip:3450@192.168.2.94:64550 SIP/2.0
Method: INVITE
Request-URI: sip:3450@192.168.2.94:64550
Request-URI User Part: 3450
Request-URI Host Part: 192.168.2.94
Request-URI Host Port: 64550
[Resent Packet: False]
Message Header
Via: SIP/2.0/UDP 192.168.2.51:5060;branch=z9hG4bK678ab923;rport
Transport: UDP
Sent-by Address: 192.168.2.51
Sent-by port: 5060
Branch: z9hG4bK678ab923
RPort: rport
From: "asterisk" <sip:asterisk@192.168.2.51>;tag=as0396138a
SIP Display info: "asterisk"
SIP from address: sip:asterisk@192.168.2.51
SIP from address User Part: asterisk
SIP from address Host Part: 192.168.2.51
SIP tag: as0396138a
To: <sip:3450@192.168.2.94:64550>
SIP to address: sip:3450@192.168.2.94:64550
SIP to address User Part: 3450
SIP to address Host Part: 192.168.2.94
SIP to address Host Port: 64550
Contact: <sip:asterisk@192.168.2.51>
Contact Binding: <sip:asterisk@192.168.2.51>
URI: <sip:asterisk@192.168.2.51>
SIP contact address: sip:asterisk@192.168.2.51
Call-ID: 67fe5a3f41b5fd6978a96b81452a8ec0@192.168.2.51
CSeq: 102 INVITE
Sequence Number: 102
Method: INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Mon, 07 Dec 2009 23:22:14 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Content-Type: application/sdp
Content-Length: 240
Message Body
Session Description Protocol
Session Description Protocol Version (v): 0
Owner/Creator, Session Id (o): root 18095 18095 IN IP4 192.168.2.51
Owner Username: root
Session ID: 18095
Session Version: 18095
Owner Network Type: IN
Owner Address Type: IP4
Owner Address: 192.168.2.51
Session Name (s): session
Connection Information (c): IN IP4 192.168.2.51
Connection Network Type: IN
Connection Address Type: IP4
Connection Address: 192.168.2.51
Time Description, active time (t): 0 0
Session Start Time: 0
Session Stop Time: 0
Media Description, name and address (m): audio 10004 RTP/AVP 0 101
Media Type: audio
Media Port: 10004
Media Protocol: RTP/AVP
Media Format: ITU-T G.711 PCMU
Media Format: DynamicRTP-Type-101
Media Attribute (a): rtpmap:0 PCMU/8000
Media Attribute Fieldname: rtpmap
Media Format: 0
MIME Type: PCMU
Sample Rate: 8000
Media Attribute (a): rtpmap:101 telephone-event/8000
Media Attribute Fieldname: rtpmap
Media Format: 101
MIME Type: telephone-event
Sample Rate: 8000
Media Attribute (a): fmtp:101 0-16
Media Attribute Fieldname: fmtp
Media Format: 101 [telephone-event]
Media format specific parameters: 0-16
Media Attribute (a): silenceSupp:off - - - -
Media Attribute Fieldname: silenceSupp
Media Attribute Value: off - - - -
Media Attribute (a): ptime:20
Media Attribute Fieldname: ptime
Media Attribute Value: 20
Media Attribute (a): sendrecv
funpig:
http://pastebin.com/m171b9345
Also, another update on this problem. I installed sofia-sip, gstreamer and empathy on a regular desktop today. I successfully connected to my asterisk server and received a phone call. The only version difference between the N900 was a newer gstreamer. The INVITE is identical and sofia-sip responds, like it should, with an OK + Session Description
Just to make this thread complete. I got this working by adding a stun server to my internal network (thanks to the maemo guys). After I had added the stun server I went to the advanced settings (for the SIP configuration of the N900) and entered the stun server's address. Now everything is working great.
Obviously you shouldn't need a stun server on an internal network where there is no NAT. Anyhow, it is a work-around that is acceptable to me.
hi, can you adise about the stun sever name and stun server name?
because i cannot sign in in gtalk on my n900.
thx
Please, tell me what is the software that you use to connect to Asterisk from Maemo.
i have another problem with sip. if im in wlan i can connect to my asterisk server.
if im in another wlan or umts, i installed openvpn to get a home ip adress.
but the buildin sip client seems to only try to connect over umts directly,
ssh or pinging works on both ends of openvpn
I know its a long time ago, but I am trying to get this working. It sounds like you had a problem with your routing table but I had the same problem the other day I could ssh to the internal IP from a remote network over openVPN but the SIP client would not connect over the openVPN link. I didnt get round to a full debug, I thought I would be lazy and ask first...
To complete: This was bug 1860. I applied the workarounds contain therein and I get a VPN connection with a SIP session running over the N900 SIP stack.
I added the libmissioncontrol-utils with this command
#apt-get install libmissioncontrol-utils
this allows the use of mc-tool commands
then added the lines
script security 2
up /etc/openvpn/nokia.up
down /etc/openvpn/nokia.down
to client.ovpn config file
then created the startup scripts nokia.up
#!/bin/sh
run-standalone.sh /usr/bin/mc-tool update sofiasip/sip/_xxxx string:local-ip-address=$4
run-standalone.sh /usr/bin/mc-tool enable sofiasip/sip/_xxxx
route add -net 192.168.2.0 netmask 255.255.255.0 gw 10.8.0.1 dev tun0
and nokia.down
#!/bin/sh
run-standalone.sh /usr/bin/mc-tool disable sofiasip/sip/_xxxx
then these two scripts must be made executable
#chmod 744 nokia*
thanks to KWEK and others who have been through this before. By the way for completeness the command
$mc-tool list
will give you the sofiasip files that you created when you made the SIP client on the phone, these must be used where I have put _xxxx in the above scripts...
mrfishball
2010-03-12, 16:41
hey has anyone try to integrate asterisk with google voice?? i did some research online and found that it actually works... but i m not sure if it ll work on n900. cant anyone give a pointer???
Off-topic, sorry!
Can you give me an invitation for Google Voice? ;-D
hey has anyone try to integrate asterisk with google voice?? i did some research online and found that it actually works... but i m not sure if it ll work on n900. cant anyone give a pointer???
Guys,
I need some help in setting up sip on openvpn. Some details of my network are:
1. Home router: 192.168.1.1
2. Asterisk server and Openvpn server: 192.168.1.152
The server.conf looks like this
port 1194
proto udp
dev tun
ca privnet/ca.crt
cert privnet/server.crt
key privnet/server.key
dh privnet/dh1024.pem
server 10.8.0.0 255.255.255.0
ifconfig-pool-persist ipp.txt
push "redirect-gateway def1 bypass-dhcp"
keepalive 10 120
comp-lzo
user nobody
group nobody
persist-key
persist-tun
status openvpn-status.log
verb 3
The client.conf on N900 is like this:
client
script-security 2
up /etc/openvpn/nokia.up
down /etc/openvpn/nokia.down
dev tun
proto udp
remote asterisk.dyndns.org 1194
resolv-retry infinite
nobind
persist-key
persist-tun
ca ca.crt
cert client.crt
key client.key
ns-cert-type server
comp-lzo
verb 3
The nokia.up and nokia.down scripts are as follows:
nokia.up
#!/bin/sh
run-standalone.sh /usr/bin/mc-tool update sofiasip/sip/_3101_40asterisk_2edyndns_2eorg0 string:local-ip-address=$4
run-standalone.sh /usr/bin/mc-tool enable sofiasip/sip/_3101_40asterisk_2edyndns_2eor
nokia.down
run-standalone.sh /usr/bin/mc-tool disable sofiasip/sip/_3101_40asterisk_2edyndns_2eorg0
When i do a test through the applet the applet i get the following response:
Apr 9 20:26:05 2010 OpenVPN 2.1_rc20 arm-unknown-linux-gnueabi [SSL] [LZO2] [EPOLL] [MH] [PF_INET6] built on Nov 29 2009
Fri Apr 9 20:26:05 2010 NOTE: the current --script-security setting may allow this configuration to call user-defined scripts
Fri Apr 9 20:26:05 2010 /usr/bin/openssl-vulnkey -q -b 1024 -m <modulus omitted>
Fri Apr 9 20:26:05 2010 ******* WARNING *******: 'client.key' cannot be verified as a non-vulnerable key. See 'man openssl-vulnkey' for details.
Fri Apr 9 20:26:05 2010 LZO compression initialized
Fri Apr 9 20:26:05 2010 Control Channel MTU parms [ L:1542 D:138 EF:38 EB:0 ET:0 EL:0 ]
Fri Apr 9 20:26:05 2010 Data Channel MTU parms [ L:1542 D:1450 EF:42 EB:135 ET:0 EL:0 AF:3/1 ]
Fri Apr 9 20:26:05 2010 Local Options hash (VER=V4): '41690919'
Fri Apr 9 20:26:05 2010 Expected Remote Options hash (VER=V4): '530fdded'
Fri Apr 9 20:26:05 2010 Socket Buffers: R=[65536->131072] S=[16384->131072]
Fri Apr 9 20:26:05 2010 UDPv4 link local: [undef]
Fri Apr 9 20:26:05 2010 UDPv4 link remote: [AF_INET]86.9.87.233:1194
Fri Apr 9 20:26:05 2010 TLS: Initial packet from [AF_INET]86.9.87.233:1194, sid=a7692b5f 7a0dab40
Fri Apr 9 20:26:09 2010 VERIFY OK: depth=1, /C=US/ST=CA/L=SanFrancisco/O=Fort-Funston/CN=Fort-Funston_CA/emailAddress=me@myhost.mydomain
Fri Apr 9 20:26:09 2010 VERIFY OK: nsCertType=SERVER
Fri Apr 9 20:26:09 2010 VERIFY OK: depth=0, /C=US/ST=CA/L=SanFrancisco/O=Fort-Funston/CN=server/emailAddress=me@myhost.mydomain
Fri Apr 9 20:26:16 2010 Data Channel Encrypt: Cipher 'BF-CBC' initialized with 128 bit key
Fri Apr 9 20:26:16 2010 Data Channel Encrypt: Using 160 bit message hash 'SHA1' for HMAC authentication
Fri Apr 9 20:26:16 2010 Data Channel Decrypt: Cipher 'BF-CBC' initialized with 128 bit key
Fri Apr 9 20:26:16 2010 Data Channel Decrypt: Using 160 bit message hash 'SHA1' for HMAC authentication
Fri Apr 9 20:26:16 2010 Control Channel: TLSv1, cipher TLSv1/SSLv3 DHE-RSA-AES256-SHA, 1024 bit RSA
Fri Apr 9 20:26:16 2010 [server] Peer Connection Initiated with [AF_INET]86.9.87.233:1194
Fri Apr 9 20:26:18 2010 SENT CONTROL [server]: 'PUSH_REQUEST' (status=1)
Fri Apr 9 20:26:19 2010 PUSH: Received control message: 'PUSH_REPLY,redirect-gateway def1 bypass-dhcp,route 10.8.0.1,topology net30,ping 10,ping-restart 120,ifconfig 10.8.0.6 1
Fri Apr 9 20:26:19 2010 OPTIONS IMPORT: timers and/or timeouts modified
Fri Apr 9 20:26:19 2010 OPTIONS IMPORT: --ifconfig/up options modified
Fri Apr 9 20:26:19 2010 OPTIONS IMPORT: route options modified
Fri Apr 9 20:26:19 2010 ROUTE default_gateway=192.168.1.254
Fri Apr 9 20:26:19 2010 TUN/TAP device tun0 opened
Fri Apr 9 20:26:19 2010 TUN/TAP TX queue length set to 100
Fri Apr 9 20:26:19 2010 /sbin/ifconfig tun0 10.8.0.6 pointopoint 10.8.0.5 mtu 1500
Fri Apr 9 20:26:19 2010 /etc/openvpn/nokia.up tun0 1500 1542 10.8.0.6 10.8.0.5 init
Fri Apr 9 20:26:20 2010 /sbin/route add -net 86.9.87.233 netmask 255.255.255.255 gw 192.168.1.254
Fri Apr 9 20:26:20 2010 /sbin/route add -net 0.0.0.0 netmask 128.0.0.0 gw 10.8.0.5
Fri Apr 9 20:26:20 2010 /sbin/route add -net 128.0.0.0 netmask 128.0.0.0 gw 10.8.0.5
Fri Apr 9 20:26:20 2010 /sbin/route add -net 10.8.0.1 netmask 255.255.255.255 gw 10.8.0.5
Fri Apr 9 20:26:20 2010 Initialization Sequence Completed
I also did a
echo 1 > /proc/sys/net/ipv4/ip_forward
on the asterisk/openvpn box.
I am able to ping to 10.8.0.1 from the N900. But the sip connection doesnt go online and i get a network error.
In the sip settings i have defined the server as asterisk.dyndns.org, do i need to change this to 10.8.0.1?
Anybody knows what might be the problem?
See bug 1860 (https://bugs.maemo.org/show_bug.cgi?id=1860).
See bug 1860 (https://bugs.maemo.org/show_bug.cgi?id=1860).
Actually i am using the nokia.up and nokia.down scripts from the bugreport but still stuck with sip network error. In the sip settings i have 'discover public address' and 'stun'. i am able to ping to the asterisk server (10.8.0.1) but cannot ping to anything else like yahoo.com from n900. The dns resolving works fine though.
any further ideas?
Does 10.8.0.5 NAT traffic from the N900? It won't go far with a 10.8.0.6 source address.
Does 10.8.0.5 NAT traffic from the N900? It won't go far with a 10.8.0.6 source address.
I havent added any iptables on the server side, what should i add for NAT?
Bit more information. After the openvpn connection has been made, the client N900 route looks like this:
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
10.8.0.5 0.0.0.0 255.255.255.255 UH 0 0 0 tun0
10.8.0.1 10.8.0.5 255.255.255.255 UGH 0 0 0 tun0
86.9.87.233 192.168.1.254 255.255.255.255 UGH 0 0 0 wlan0
192.168.1.0 10.8.0.5 255.255.255.0 UG 0 0 0 tun0
192.168.1.0 0.0.0.0 255.255.255.0 U 0 0 0 wlan0
0.0.0.0 10.8.0.5 128.0.0.0 UG 0 0 0 tun0
128.0.0.0 10.8.0.5 128.0.0.0 UG 0 0 0 tun0
0.0.0.0 192.168.1.254 0.0.0.0 UG 0 0 0 wlan0
and the asterisk/openvpn server looks like
#route -n
Kernel IP routing table
Destination Gateway Genmask Flags Metric Ref Use Iface
10.8.0.2 0.0.0.0 255.255.255.255 UH 0 0 0 tun0
10.8.0.0 10.8.0.2 255.255.255.0 UG 0 0 0 tun0
192.168.1.0 0.0.0.0 255.255.255.0 U 0 0 0 eth0
127.0.0.0 0.0.0.0 255.0.0.0 U 0 0 0 lo
0.0.0.0 192.168.1.1 0.0.0.0 UG 0 0 0 eth0
So 192.168.1.1 is your Internet-facing gateway, and presumably does the NAT. Does it have a route for 10.8.0.6 via 192.168.1.152? What are 192.168.1.254 and 10.8.0.2?
I changed the vpn network from 192.168.1.1 to 172.16.1.1. Also specified nat on the vpn server using
iptables -t nat -A POSTROUTING -o eth0 -j MASQUERADE
The vpn connection now works for email/browser but doesnt work for sip. Please follow-up on this post (http://talk.maemo.org/showthread.php?t=49765&highlight=sip).
Hi All,
A noob N900 user here...
I'm getting a different problem with my N900 on Asterisk.
Mine connects to the server fine, I can dial extensions, outside numbers, all OK. I can call the N900 over the sip network.
The problem - no audio either transmitted or received.
I've looked at the settings, can't see anything different to those posted on the forum as working.
I've used an E70, an E65, a N95, and an E71 on the box with no problems.
Any ideas where to start looking?
I've tried runnig with Stun enabled and disabled, which made no difference.
Hi All,
A noob N900 user here...
I'm getting a different problem with my N900 on Asterisk.
Mine connects to the server fine, I can dial extensions, outside numbers, all OK. I can call the N900 over the sip network.
The problem - no audio either transmitted or received.
I've looked at the settings, can't see anything different to those posted on the forum as working.
I've used an E70, an E65, a N95, and an E71 on the box with no problems.
Any ideas where to start looking?
I've tried runnig with Stun enabled and disabled, which made no difference.
Iam getting the exact same thing, i can register and make calls to extensions but i cant send or recive and voices.
Works for all my other computers and ata boxes but not with my n900, any suggestions ?
Valdo_23
2010-12-04, 22:30
Hello Ufoeke
I have the same problem too. Did you find any answer for this problem and could you please help me if you solved your problem?
Thank you.
Hi.
I've been reading around to try to find answers but I couldn't, so I'll expose my question here which is where I think fits better:
I would like to connect the N900's SIP client to my home network, so that my fixed line is available when I am at work, but the corporate network of the company I work for only allows access via http/https proxy to the outside world.
I have managed to configure my corporate laptop to use a http tunnel to access a host in my home network from work.
My home network's host listens on port 443 and switches incoming connections to port 22 (via apaches's proxy option/directive), where I can get a command line via ssh server.
On the other hand, my SIP server is in the ISP's network and only works when accessed from my home network.
My doubt is wether it is possible to use this http tunnel (not VPN) to access my home network and then, connect to the SIP server.
Has anyone dealed with this environment before?
Regards.
Offhand, as a quick guess, you may be able to do it, but set SIP to use TCP instead of UDP.
Good luck,
-Jeff Moe
O.K.
Thank you. I'll keep that in mind.
By the way, I found a package called proxytunnel (the one I use in my laptop at work) in the extras-devel repository, but hadn't been able to install the binaries (seems it has never been compiled successfully).
Or maybe the only option is to try to compile the sources?
Regards.
O.K.
Thank you. I'll keep that in mind.
By the way, I found a package called proxytunnel (the one I use in my laptop at work) in the extras-devel repository, but hadn't been able to install the binaries (seems it has never been compiled successfully).
Or maybe the only option is to try to compile the sources?
Regards.
Followup.
I finally found the 'armel' .deb somewhere and installed it flawlessly with dpkg -i XXX
This morning tested it and worked from my N900....
Nice :) Thanks for letting us know.
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