SipToSis-Skype Gateway Tips

[drsatch]

Whoops! Didn't check back there.

 

[jrglass]

Error Messages

When Pbxiaf Org II boots I get the folowing error message in the siptosis log


2009-03-03 15:19:44,615 [main] ERROR org.zoolu.sip.provider.SipProvider_evt - error
java.net.BindException: Address already in use
at java.net.PlainDatagramSocketImpl.bind0(Native Method)
at java.net.PlainDatagramSocketImpl.bind(PlainDatagramSocketImpl.java:82)
at java.net.DatagramSocket.bind(DatagramSocket.java:368)
at java.net.DatagramSocket.<init>(DatagramSocket.java:210)
at java.net.DatagramSocket.<init>(DatagramSocket.java:261)
at java.net.DatagramSocket.<init>(DatagramSocket.java:234)
at org.zoolu.net.UdpSocket.<init>(UdpSocket.java:48)
at org.zoolu.sip.provider.UdpTransport.<init>(UdpTransport.java:54)
at org.zoolu.sip.provider.SipProvider.startTrasport(SipProvider.java:356)
at org.zoolu.sip.provider.SipProvider.<init>(SipProvider.java:277)
at local.ua.SkypeUA.initUA(SkypeUA.java:142)
at local.ua.SkypeUA.<init>(SkypeUA.java:134)
at local.ua.SkypeUA.main(SkypeUA.java:121)
2009-03-03 15:32:33,465 [main] INFO local.ua.SkypeUA - Starting SipToSis v20090214
2009-03-03 15:32:33,474 [main] INFO local.ua.SkypeUA - os=Linux arch=i386 ver=2.6.18-92.1.6.el5
2009-03-03 15:32:33,475 [main] INFO local.ua.SkypeUA - javaVer=1.6.0_12 - Sun Microsystems Inc.
2009-03-03 15:32:33,562 [main] ERROR local.ua.sscodecs.SSCodecFactory - Codec Error: GSMTRI - Codec won't instantiate - local.ua.sscodecs.SSCodec_GSMTRI
2009-03-03 15:32:33,562 [main] ERROR local.ua.SkypeUA - Codec: GSMTRI not loaded.
2009-03-03 15:32:33,793 [main] INFO local.ua.SkypeUA - Available Codecs: PCMU(0),PCMA(8),iLBC(98),speex(97)
2009-03-03 15:32:33,793 [main] INFO local.ua.SkypeUA - DTMF rfc2833(101)
2009-03-03 15:32:33,794 [main] INFO local.ua.SkypeUA - initSkype - If stuck, check Skype online & API auth

Can anyone point me in the right direction?

Thanks,

Jeff

 

[wardmundy]

Sounds like you didn't get it working manually first. To start anew, rm -r /root/.Skype

 

[jrglass]

Sounds like you didn't get it working manually first. To start anew, rm -r /root/.Skype

Ward,

I followed your instructions to start anew. When I run ./SipToSis_Linux from the siptosis directory from the xwindow. I get an error message no such file or directory. The SipToSis_Linux file is in my directory

size 354b dated 1/31/09 permission 0777

Any sugestions?

Thanks Jeff

 

[Lost Trunk]

I followed your instructions to start anew. When I run ./SipToSis_Linux from the siptosis directory from the xwindow. I get an error message no such file or directory.

I don't know if it's been fixed yet, but in at least one download the program was actually named SipToSis_linux, and since Linux filenames are case sensitive, you have to invoke it using the correct case for the letter "L": ./SipToSis_linux vs. ./SipToSis_Linux

 

[jrglass]

I don't know if it's been fixed yet, but in at least one download the program was actually named SipToSis_linux, and since Linux filenames are case sensitive, you have to invoke it using the correct case for the letter "L": ./SipToSis_linux vs. ./SipToSis_Linux

Thanks for the tip. I will try that tomorrow!

Jeff

 

[wardmundy]

It's SipToSis_linux in the build if you got it from the Nerd Vittles link.

 

[jrglass]

It's SipToSis_linux in the build if you got it from the Nerd Vittles link.

Thanks,
That did the trick. Now I have an outgoing problem. Here is the log
9-03-05 10:46:31,189 [main] INFO local.ua.SkypeUA - Starting SipToSis v20090214
2009-03-05 10:46:31,191 [main] INFO local.ua.SkypeUA - os=Linux arch=i386 ver=2.6.18-92.1.6.el5
2009-03-05 10:46:31,192 [main] INFO local.ua.SkypeUA - javaVer=1.6.0_12 - Sun Microsystems Inc.
2009-03-05 10:46:31,286 [main] ERROR local.ua.sscodecs.SSCodecFactory - Codec Error: GSMTRI - Codec won't instantiate - local.ua.sscodecs.SSCodec_GSMTRI
2009-03-05 10:46:31,288 [main] ERROR local.ua.SkypeUA - Codec: GSMTRI not loaded.
2009-03-05 10:46:32,013 [main] INFO local.ua.SkypeUA - Available Codecs: PCMU(0),PCMA(8),iLBC(98),speex(97)
2009-03-05 10:46:32,014 [main] INFO local.ua.SkypeUA - DTMF rfc2833(101)
2009-03-05 10:46:32,015 [main] INFO local.ua.SkypeUA - initSkype - If stuck, check Skype online & API auth
2009-03-05 10:46:32,155 [main] INFO local.ua.SkypeUA - Skype Not Connected - retrying every 5 seconds
2009-03-05 10:46:37,214 [main] INFO local.ua.SkypeUA - SkypeVer:2.0.0.72
2009-03-05 10:46:37,268 [main] INFO local.ua.SkypeUA - SkypeUserId:voipohio
2009-03-05 10:46:37,303 [main] INFO local.ua.SkypeUA - Config - skypeClientSupportsMultiCalls:false concurrentCallLimit:1
2009-03-05 10:46:37,303 [main] INFO local.ua.SkypeUA - SipToSis contact_url=<sip:[email protected]:5070>
2009-03-05 10:46:37,304 [main] INFO local.ua.SkypeUA - RTP Ports: 63200-63200 Local Skype Ports: 64432-64433
2009-03-05 10:46:37,584 [local.ua.SSCallChannel.#C0] INFO local.ua.SkypeUserAgent.#C0 - WAITING FOR INCOMING CALL
2009-03-05 10:47:22,296 [org.zoolu.net.UdpProvider.T0] INFO local.ua.SSCallChannel.#C0 - incoming sip call from "Office desk Ext-705" <sip:[email protected]> callee=<sip:[email protected]:5070>
2009-03-05 10:47:22,319 [org.zoolu.net.UdpProvider.T0] INFO local.ua.SSCallChannel.#C0 - handleSipCall - rejected call
2009-03-05 10:47:25,144 [org.zoolu.net.UdpProvider.T0] INFO local.ua.SkypeUserAgent.#C0 - WAITING FOR INCOMING CALL
2009-03-05 10:47:35,555 [org.zoolu.net.UdpProvider.T0] INFO local.ua.SSCallChannel.#C0 - incoming sip call from "6142213800" <sip:[email protected]> callee=<sip:[email protected]:5070>
2009-03-05 10:47:35,556 [org.zoolu.net.UdpProvider.T0] INFO local.ua.SSCallChannel.#C0 - handleSipCall - rejected call
2009-03-05 10:47:38,367 [org.zoolu.net.UdpProvider.T0] INFO local.ua.SkypeUserAgent.#C0 - WAITING FOR INCOMING CALL

Incoming works perfect. Outgoing the call gets rejected?

Setup Trunk, Outgoing Route and Speed dial oer Nerd Vittles artical.

Jeff

 

[rossiv]

Re-Write!

1. How do you un-minimize skype once its minimized?

2. Can you change the mothership URI to something different when someone calls you from Skype?
Ex. When someone calls you, it calls sip://[email protected]:5060 or sip://[email protected]:5060

3. Could you base the routing on Skype name?

4. After rebooting and having Skype auto-launch, how do you get back to it to sign out and use a different user name??

 

[wardmundy]

1. Reboot. If it's been committed to the background, you'll have to delete the setup and start over.

2. Feel free. But don't expect a tutorial. Why does it matter?

3. You could but, again, why? You can only have one Skype account logged in with this tutorial. If you need more, visit the SipToSis site and rework for multiple Skype accounts to meet your needs.

4. Delete the existing setup and start over.

 

[rossiv]

What I meant by route based on skype name was that different caller's skype names, not my skype names.
I agree about the mothership thing. Not too much of a big deal.

 

[kwest]

I have a google voice number pointed to gizmo5 then pointed at my PIAF box, then I do a CID lookup so I get the caller information, is there a way to pass the CID over to skype so the receiving skype software can see who is really calling?
Thanks

 

[nojstevens]

I have all this working properly, can receive and make calls via skype. However, if I leave the config as *.*.mydynamicdns,calleeid then after about 2 hours I can't make outbound calls, but I can still receive them (I get an 'all circuits are busy' error). If i change this and leave it as *.*.*,calleeid then it works permanently.

I've read this thread and can't determine if I am comfortable with the risk of doing this.My Piaf is behind a dd-wrt firewall, so is it ok to leave *.*.* in or should i tackle the root cause. If so, what might this be?

I also tried localhost and 127.0.0.1 but that didn't work either

Thanks for any advice

Jon

 

[jpuddy]

I've been working through this thread and various others trying to get Skype and Asterisk working together inside an ESXi VM. The problem was the lack of real sound devices. To get Skype working inside a virtual machine, all you need to do is enable the dummy driver. This can easily be done by editing the skype-start script, and adding this line ahead of the other commands:

modprobe snd-dummy enable=1

Hope it helps someone else.

 

[dandy_don]

No outbound skype audio

Firstly -- Thanks for the excellent PBX documentation, etc.
I'm a PBX noob. I followed the incredible PBX install and have a working PBX with two GXP-2000 phones with an Artigo A1100 server.

I have it working with Sipgate & Google Voice, as well as with Vitelity. That all works great!

I followed the instructions to add Skype. I can make Skype test calls, but can't send any audio. I have searched quite bit but can't find the solution. When I plug a working microphone into the server, and make a Skype test call from the server I still can't get any audio sent. I can receive audio but can't send. I can make a Skype test call from the GXP-2000s but there is no audio being sent to Skype.

I'm just a noob here and would really appreciate some help.

Thanks,
Don

 

[wardmundy]

Depends on the sound card in your machine. Dell's generally are incompatible. Acer Aspire's of all flavors seem to work great. YMMV with other brands.

Not much you can do if it doesn't work other than trying another type of machine. Sorry.

 

[dandy_don]

Thanks!

Ward -- Thanks for doing such a great job with this! I had wanted to setup an asterisk system for several years but the entire PIAF and Increbible PBX scripts made it so much easier!

I'm disapointed that my server sound card won't support outbound audio to Skype, even when initiated from the working SIP hardware phones. I don't understand why this would matter when the audio originates from a hardware SIP phone elsewhere in the PBX system.

Would you please weigh in on this idea... I have a USB audio adapter that I would like to try -- it is a small USB dongle that is the size of a flashdrive and has a mic input and headphone output. This works with my other linux boxes when I need more audio ports -- Do you think this would be compatible with the incredible PBX setup and with Skype?

I really like the Artigo A1100 so will try to stick with it.
Either way, I'll post my results.

Thanks again!
Don

 

[blanchae]

USB sound card - nobody will know until you try it.. So go ahead and see if it works. Don't forget to report back what the results are - good or bad.

 

[dandy_don]

Well -- I gave it a good try but no good! I was finally able to get audio back and forth when using the built-in sound card, but the mic audio was consistently shifted downward in frequency. The tempo was correct, but the entire audio was shifted lower in frequency.
However, I still get no transmit audio when I call *echo123...

Why is the sound card needed when placing or receiving a skype call when the audio is digital from the internet and should be digital when coming from the hardware sip phone?

Thanks,
Don

 

[wardmundy]

Because SIPtoSIS intercepts the incoming digital audio and redirects it via the analog sound card and vice versa. Think of it as using a tape recorder to capture a song on the radio and then playing it back on the phone via the tape recorder. Only difference is SIPtoSIS does it in both directions transparently.