Furikup and Asterisk (VOIP Server)

[GoDE]

Hi there,
here is my status.

I tested furikup with my internal (same lan) Asterisk Server and the connection will be established.
But directly after this i see on the console from Asterisk that the furikup client is "...unreachable".

Making calls works with the permanent interrupted sound and the simultanously blinking lan led on the psp.
Icoming calls don't work, Asterisk says about like this "creating sip chanel to peer is not possible".
But when you enable sip debugging on asterisk you see that there is traffic between the client
and the server. You see it also on the client, i will take a look this evening.

No Stun entries in the config.
Using the disablenat=yes option in the config (found in the src).  

-snip-
server=192.168.0.5
username=xxxx
password=yyyy
stunserver=
stunport=
input=file
disablenat=yes
-snip-

Using an Fritz!Box 7170 and WEB encryption and also tested with
an Linksys AP WAP54G without encryption in 11MBit Mode.
Same results.

I am on German language PSP1004 3.71M33#4.

Good Job  

Cheers
GoDE

[samuraix]

Ok....   Not sure but is your Asterisk's being used as a proxy or User Agent?  If Proxy, I think Furikup is not handling probably record-route headers properly.  Could you post a trace so we can look at it deeper.

Or post your logfile.

[GoDE]

Hi,
thanks for the reply.

My Asterisk is running on eisfair http://www.eisfair.org which is an small linux distribution.
The server has two isdn cards. One for the connection to my fritz!box (which handles
the external sip accounts and isdn trunks), and one for internal S0 ISDN Buss (for ISDN phones).
I am connected with a few sip clients (X-Lite/NDS Sipclient/furikup) and an ISDN Phone to the asterisk server.

I am not sure what you mean with proxy/user agent.
When you look against an external Sip Provider than i will say Asterisk runs as an proxy.

IP's
Eisfair/Asterisk: 192.168.0.5
PSP: 192.168.0.55 (DHCP)


I enable the verbose mode on the psp and get continously the messages
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"Registration failed!"
"Users is successfully regsitered!"
...and again !


Here is the debug from the asterisk deug console direkt after the logon of furikup.
furikup has the extension 1000 on my Asterisk.

-snip-
eisfair*CLI>
eisfair*CLI> Connected to Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r currently running on eisfair (pid = 356)
eisfair*CLI> Verbosity was 0 and is now 5
eisfair*CLI>     -- Registered SIP '1000' at 192.168.0.55 port 5060 expires 120
eisfair*CLI>     -- Saved useragent "eXosip/3.0.3" for peer 1000
eisfair*CLI> Jan 27 12:53:21 NOTICE[389]: chan_sip.c:11787 sip_poke_noanswer: Peer '1000' is now UNREACHABLE!  Last qualify: 0
eisfair*CLI> eisfair*CLI>
-snip-

And a part of the debug/log during the connection when the PSP says
"New request ouside call received!"

-snip-
eisfair*CLI> sip debug
SIP Debugging enabled
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" <sip:asterisk@192.168.0.5>;tag=as1898fa3d
To: <sip:1000@192.168.0.55:5060;line=0d3eaaf29456571>
Contact: <sip:asterisk@192.168.0.5>
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #1 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" <sip:asterisk@192.168.0.5>;tag=as1898fa3d
To: <sip:1000@192.168.0.55:5060;line=0d3eaaf29456571>
Contact: <sip:asterisk@192.168.0.5>
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #2 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" <sip:asterisk@192.168.0.5>;tag=as1898fa3d
To: <sip:1000@192.168.0.55:5060;line=0d3eaaf29456571>
Contact: <sip:asterisk@192.168.0.5>
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #3 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" <sip:asterisk@192.168.0.5>;tag=as1898fa3d
To: <sip:1000@192.168.0.55:5060;line=0d3eaaf29456571>
Contact: <sip:asterisk@192.168.0.5>
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #4 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" <sip:asterisk@192.168.0.5>;tag=as1898fa3d
To: <sip:1000@192.168.0.55:5060;line=0d3eaaf29456571>
Contact: <sip:asterisk@192.168.0.5>
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5'
eisfair*CLI>
-snip-

I will test the new client now....
GoDE
Posted on: January 27, 2008, 12:28:43


...and the log from the new furikup v0.1.12.

-snip-
Uploaded to
http://uploaded.to/?id=57e2kf
because of to many text lines.
-snip-

Thanks
GoDe

[cswindle]

That is caused by Furikup not sending a response to the SIP OPTIONS message, which Asterisk is using as a NAT traversal keep-alive. At some point I will sort out replying to that message, however for now I expect that there is an option in Asterisk to disable it, as you do not need it.


Chris

[GoDE]

..not sending a response to the SIP OPTIONS message...
I expect that there is an option in Asterisk to disable it...

Found here http://www.asteriskguru.c...r_is_now_unreachable.html
and change in sip.conf the option "qualify=1000" to "qualify=no" for the furikup extension.
Incoming calls now be received by the psp   .

Only the stuttering audio now....
Thx

GoDE

[cswindle]

Glad the adition of extra diagnostics was able to help out.


Chris

[GoDE]

Glad the adition of extra diagnostics was able to help out.

..yeah that's really helpfull  

GoDE

I like that APP  

[babyboi828]

wha?? how do you even manage to call the PSP??? i can call landlines and cell phones but my number always comes in as "restricted"  so i have no idea what to even dial to attempt this  Embarrassed thanks for anyhelp i can get ..

[klofton11]

Hi,
thanks for the reply.

My Asterisk is running on eisfair http://www.eisfair.org which is an small linux distribution.
The server has two isdn cards. One for the connection to my fritz!box (which handles
the external sip accounts and isdn trunks), and one for internal S0 ISDN Buss (for ISDN phones).
I am connected with a few sip clients (X-Lite/NDS Sipclient/furikup) and an ISDN Phone to the asterisk server.

I am not sure what you mean with proxy/user agent.
When you look against an external Sip Provider than i will say Asterisk runs as an proxy.

IP's
Eisfair/Asterisk: 192.168.0.5
PSP: 192.168.0.55 (DHCP)


I enable the verbose mode on the psp and get continously the messages
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"Registration failed!"
"Users is successfully regsitered!"
...and again !


Here is the debug from the asterisk deug console direkt after the logon of furikup.
furikup has the extension 1000 on my Asterisk.

-snip-
eisfair*CLI>
eisfair*CLI> Connected to Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r currently running on eisfair (pid = 356)
eisfair*CLI> Verbosity was 0 and is now 5
eisfair*CLI>     -- Registered SIP '1000' at 192.168.0.55 port 5060 expires 120
eisfair*CLI>     -- Saved useragent "eXosip/3.0.3" for peer 1000
eisfair*CLI> Jan 27 12:53:21 NOTICE[389]: chan_sip.c:11787 sip_poke_noanswer: Peer '1000' is now UNREACHABLE!  Last qualify: 0
eisfair*CLI> eisfair*CLI>
-snip-

And a part of the debug/log during the connection when the PSP says
"New request ouside call received!"

-snip-
eisfair*CLI> sip debug
SIP Debugging enabled
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #1 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Voip Phone System
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #2 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #3 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent:
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #4 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5'
eisfair*CLI>
-snip-

I will test the new client now....
GoDE
Posted on: January 27, 2008, 12:28:43


...and the log from the new furikup v0.1.12.

-snip-
Uploaded to
http://uploaded.to/?id=57e2kf
because of to many text lines.
-snip-

Thanks
GoDe

Cool. I checked out your debug and its' perfect. I was having the same problem. Thanks,

Jeff
Posted on: July 01, 2009, 07:52:16 PM

That is caused by Furikup not sending a response to the voip OPTIONS message, which Asterisk is using as a NAT traversal keep-alive. At some point I will sort out replying to that message, however for now I expect that there is an option in Asterisk to disable it, as you do not need it.


Chris

Yeah I noticed that too. I think there is an option in Asterisk but I couldn't find it either. I k now for a fact you don't need it though.

[donmo]

Glad the adition of extra diagnostics was able to help out.


Chris
International Calling Voip Phone System

I just tried this app out for the first time and dug it.
Thanks for posting!

[krahman]

Hi, Can you let me know what option it is to disable Asterisk Sip Retransmission of INVITE ACK. I have tried qualify=no, but not working. I am using Trixbox with Asterisk 1.4.22-3. Due to this my call gets disconnected after every 10 to 20 seconds. And I have even put the machine on DMZ still same issue, so it seems that firewall is not an issue. Looking forward for a solution soon...... Thanks...

[Acecare]

In regards problem, I've been seeing a lot online about opening TCP ports when using Asterisk. I am currently using UDP ports open with sip signalling and RTP. However, I get random audio problems with it. So as of now, I am searching for the possible solution of this problem.

[David Morgan]

Hi,
thanks for the reply.

My Asterisk is running on eisfair http://www.eisfair.org which is an small linux distribution.
The server has two isdn cards. One for the connection to my fritz!box (which handles
the external sip accounts and isdn trunks), and one for internal S0 ISDN Buss (for ISDN phones).
I am connected with a few sip clients (X-Lite/NDS Sipclient/furikup) and an ISDN Phone to the asterisk server.

I am not sure what you mean with proxy/user agent.
When you look against an external Sip Provider than i will say Asterisk runs as an proxy.

IP's
Eisfair/Asterisk: 192.168.0.5
PSP: 192.168.0.55 (DHCP)


I enable the verbose mode on the psp and get continously the messages
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"New request ouside call received!"
"Registration failed!"
"Users is successfully regsitered!"
...and again !


Here is the debug from the asterisk deug console direkt after the logon of furikup.
furikup has the extension 1000 on my Asterisk.

-snip-
eisfair*CLI>
eisfair*CLI> Connected to Asterisk 1.2.9.1-BRIstuffed-0.3.0-PRE-1r currently running on eisfair (pid = 356)
eisfair*CLI> Verbosity was 0 and is now 5
eisfair*CLI>     -- Registered SIP '1000' at 192.168.0.55 port 5060 expires 120
eisfair*CLI>     -- Saved useragent "eXosip/3.0.3" for peer 1000
eisfair*CLI> Jan 27 12:53:21 NOTICE[389]: chan_sip.c:11787 sip_poke_noanswer: Peer '1000' is now UNREACHABLE!  Last qualify: 0
eisfair*CLI> eisfair*CLI>
-snip-

And a part of the debug/log during the connection when the PSP says
"New request ouside call received!"

-snip-
eisfair*CLI> sip debug
SIP Debugging enabled
12 headers, 0 lines
Reliably Transmitting (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #1 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Phone Service
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #2 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #3 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Voip Provider List
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Retransmitting #4 (NAT) to 192.168.0.55:5060:
OPTIONS sip:1000@192.168.0.55:5060;line=0d3eaaf29456571 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.5:5060;branch=z9hG4bK0188378b;rport
From: "asterisk" ;tag=as1898fa3d
To:
Contact:
Call-ID: 2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Sun, 27 Jan 2008 12:23:25 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Length: 0


---
Destroying call '2b471d5472faa4c5602fe8ea4cd8edfc@192.168.0.5'
eisfair*CLI>
-snip-

I will test the new client now....
GoDE
Posted on: January 27, 2008, 12:28:43


...and the log from the new furikup v0.1.12.

-snip-
Uploaded to
http://uploaded.to/?id=57e2kf
because of to many text lines.
-snip-

Thanks
GoDe

Been looking for a solution for the same exact problem for ages. I don't know how I never found this place before.  

Thanks man.

[donmo]

Hi,
thanks for the reply.
My Asterisk is running on eisfair Business Telephone System which is an small linux distribution.
The server has two isdn cards. One for the connection to my fritz!box (which handles
the external sip accounts and isdn trunks), and one for internal S0 ISDN Buss (for ISDN phones).
I am connected with a few sip clients (X-Lite/NDS Sipclient/furikup) and an ISDN Phone to the asterisk server.
I am not sure what you mean with proxy/user agent.
When you look against an external Sip Provider than i will say Asterisk runs as an proxy.
IP's
Thanks
GoDe

Been looking for a solution for the same exact problem for ages. I don't know how I never found this place before.  
Thanks man.

_________
Great site and a plethora of info!
Thanks!