Utility: sip2yac

[KJake]

Originally Posted by SDeGonge

This sounds like a very cool utility. Will it display incomming calls on the HD200 clients?

yup!

see: http://forums.sagetv.com/forums/showthread.php?t=37783

[KJake]

toricred and seaverd - what version of Windows are you running? I'll attempt to replicate the issues!

[rochurch]

Ok, I poured through the other thread at http://forums.sagetv.com/forums/show...ghlight=vonage and it looks like there is a good deal of outdated and unrelated (for me) information there.
Does anybody have a setup or could do a quick step by step to get this working with Vonage and a Linksys router running Tomato? I would love to do this without tracking down a CID modem.
Much appreciated..... If I got this working the WAF would go up!

[seaverd]

KJake,

I am running Win XP SP3.

Thank you for looking into this!

Dan

[seaverd]

Another quick question...Are people installing the yac listener as a service or do they just add it to their startup group....just curious.

Thanks,
Dan

[toricred]

I'm also running XP SP-3.

I'm currently using Vonage and DD-WRT. I used iptables in the startup script to tee all packets to my server. I'm currently mirroring all packets. I've also recently purchased a managed switch that does port mirroring so I'll be moving to that out of convenience, but it's been working perfectly with my current setup as long as I run the service with a username that has a password. It also happens to be the account I'm running the Sage service under if that makes any difference.

[seaverd]

To be honest - I am not sure what I did but everything is running perfect. I gave the service to interact with the desktop, and changed the path for the program and config file to c:\sip2yac\ and everything now works great.

Thanks,

Dan

[toricred]

I've been using this for several months now and it has worked perfectly. Last week I switched to ooma. I had this working for the first two days, but it suddenly stopped working. I also upgraded several plugins at the same time so maybe that is related. Any ideas?

[KJake]

Ooma does not use standard SIP ports, make sure that you have it configured correctly.

From their site: http://ooma.custhelp.com/app/answers/detail/a_id/104 - I would try these ports to see if it starts working again. If you suspect it is randomly assigned within that 10000-20000 range, you should be able to use "udp portrange 10000-20000" as your filter string.

Ooma uses the following application ports for outbound data and voice traffic:
UDP 53, UDP 123, UDP 514, UDP 1194, UDP 3386, UDP 3480, UDP 10000-20000, TCP 53 and TCP 443.

[toricred]

According to their support everything is encrypted so I guess I'll have to go back to using a modem.

[war10ck]

toricred,

I currently have vonage and want to implement what you had success with. I'm not very technical and I'm struggling with some of the steps.

You said you used Vonage and DD-WRT then used iptables in the startup script to tee all packets to your server. How would the managed switch with port mirroring make your setup easier? Would that allow you to get rid of the startup script, DD-WRT or both?

Sorry for the noob questions, all of this is completely new to me.

[toricred]

I only did the stuff with iptables because at the time I didn't have a managed switch at the time. With a switch that supports mirroring it is completely unecessary.

[Zogg]

I just switched from Vonage to Voipo, and sip2yac stopped working for me. I changed the port number after using Wireshark to confirm what was going on, but I guess the packet data is different and the call isn't detected.

I don't see CNAM or CIDNAME but instead info like:

CSeq: 27308 INVITE
Max-Forwards: 51
m: <sip:+19727451234@66.218.197.19:5060;nat=yes>
Contact-URI: sip:+19727451234@66.218.197.19:5060;nat=yes
Contact parameter: nat=yes>
Content-Length: 283
c: application/sdp
P-Asserted-Identity: "WIRELESS CALLER" <sip:9727451234@pstn>
SIP Display info: "WIRELESS CALLER"
SIP PAI Address: sip:9727451234@pstn
SIP PAI User Part: 9727451234
SIP PAI Host Part: pstn

I can provide more info if needed. Would appreciate assistance to make it work again. Thanks!

[KJake]

Originally Posted by Zogg

I just switched from Vonage to Voipo, and sip2yac stopped working for me.

Hi Zogg - can you give this a shot? I think what you provided was enough to make changes.

http://dl.dropbox.com/u/15596865/sip2yac.zip

If it works for you, then if someone else in the thread will confirm that it still works with Vonage, etc, I'll commit the new version to SVN as the next release.

[Zogg]

Originally Posted by KJake

Hi Zogg - can you give this a shot? I think what you provided was enough to make changes.

http://dl.dropbox.com/u/15596865/sip2yac.zip

If it works for you, then if someone else in the thread will confirm that it still works with Vonage, etc, I'll commit the new version to SVN as the next release.

Hi KJake, thanks for taking a look at this. It isn't working quite right. First, it seems to be parsing other packets as well as just incoming/INVITE so I see YAC pop up on the TV sometimes when it shouldn't.

The incoming name is basically okay but I think too long as it doesn't all show in the pop up window. Also, getting this error on the server: "Use of uninitialized value in concatenation <.> or string at sip2yac.pl line 352.".

I have attached two complete INVITE packets captured with Wireshark from different external phone lines, so hopefully these will help.

Thanks!

[FlyingDoc]

Can anyone tell if if this will work on Fios Digital Voice? and if so how do i set it up?

apologies if this is obvious to everyone else but i am new to the IP phone game and its all magic to me

Thanks

[KJake]

Zogg - please try that same link. I've forced on some extra debugging - you should be able to see the actual INVITE packet as I'm able to parse it. Can you send that to me?

Thanks.

[KJake]

@FlyingDoc - I don't personally have experience with it, nor do I recall anyone here using this with it, but the Internets seem to suggest that the FIOS VOIP uses SIP. Now, as far as ports and finding out if they encrypt the traffic...no idea...

[Zogg]

Originally Posted by KJake

Zogg - please try that same link. I've forced on some extra debugging - you should be able to see the actual INVITE packet as I'm able to parse it. Can you send that to me?

Thanks.

Hi KJake, thanks for the continued assistance. I ran the new version and here are two logs from two different calls. This version of the program dies after it receives an incoming call, but you are probably aware of that.

The concatenation error was repeated twice and then the program closed, but I only included the error message once.

Thanks!

[FlyingDoc]

Originally Posted by KJake

@FlyingDoc - I don't personally have experience with it, nor do I recall anyone here using this with it, but the Internets seem to suggest that the FIOS VOIP uses SIP. Now, as far as ports and finding out if they encrypt the traffic...no idea...

Thanks for the info ...looks like this will have to take a back seat for now but i may get back to it later in the year. If i ever make any progress i'll post here