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		<title><![CDATA[SipToSis Skype Gateway Bridge Forum]]></title>
		<link>http://www.mhspot.com/stsforum/index.php</link>
		<description><![CDATA[The most recent topics at SipToSis Skype Gateway Bridge Forum.]]></description>
		<lastBuildDate>Tue, 06 Dec 2011 00:39:26 +0000</lastBuildDate>
		<generator>PunBB</generator>
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			<title><![CDATA[Transfer Question]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=560&amp;action=new</link>
			<description><![CDATA[Is it possible for SipToSis or PBX to recognize if a skype user is transferring a PSTN call (that's physically attached to PBX) back into PBX and thus eliminating the extra skype calls and sip calls. My setup is as follows,
PSTN lines -> PBX -> stsProxy -> stsTrunks (Incoming and Outgoing)

In other words, when someone calls in from a PSTN line which gets answered by a skype user through SipToSis then the Skype User transfers to an incoming only siptosis channel/skypeuser which gets routed to a PBX extension. I would like to know if it's possible to eliminate the extra audio delay because of the use of skype and just rewrite the route to use a ZAP channel for the PSTN directly to the PBX extension.<br><br><a rel="nofollow" target="_blank" href="http://www.anrdoezrs.net/72103tenkem15384A55132783478?cm_mmc=CJ-_-2488992-_-3162833-_-WW-%20468x60"><img src="http://www.awltovhc.com/ic108fz2rxvGKINJPKKGIHMNIJMN" alt="AW - 468x60" border="0"/></a>]]></description>
			<author><![CDATA[dummy@example.com (playcraft222)]]></author>
			<pubDate>Tue, 06 Dec 2011 00:39:26 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=560&amp;action=new</guid>
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			<title><![CDATA[Follow me type of command]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=559&amp;action=new</link>
			<description><![CDATA[Is there any way to do a follow me command on siptosis? 

My setup is as follow

skype-user -> siptosis -> asterisk extension 

I can set up a follow me command at my asterisk extension but I want to know if I can do this at SkypetoSipauth.props file.]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Mon, 05 Dec 2011 13:34:07 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=559&amp;action=new</guid>
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			<title><![CDATA[A call to the destination user is already ongoing]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=558&amp;action=new</link>
			<description><![CDATA[I am trying to call a skype user twice using stsProxy, stsTrunkBuilder, and SipToSis through an extension in Trixbox. I can call the skype user once and accept but I cannot dial that skype user again through another phone while the call is active as I receive]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Sat, 03 Dec 2011 16:40:46 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=558&amp;action=new</guid>
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			<title><![CDATA[siptosis trunks went offline and unreachable after 30 to 90 minutes]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=549&amp;action=new</link>
			<description><![CDATA[For some reaons I am not, my siptosis trunks will become offline unreachable after one hour or so idle. I have tried various settings in siptosis_xx.cfg but still encounter the same problem.

Then I found xvfb_xx.log

FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing.
FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing.
FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing.
FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing.
FreeFontPath: FPE "unix/:7100" refcount is 2, should be 1; fixing.

I search for sometimes but can't find a fix.

Now I switch to vnc and encounter 2 problems
1. not having pid properly place. It is now at /tmp not at /root/.vnc
2. stability is not sure because it still goes away for unknown reason

I am thinking of using a cron job to stop and start all the trunks at a fixed interval but seems not a good idea.]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Fri, 02 Dec 2011 02:02:38 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=549&amp;action=new</guid>
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			<title><![CDATA[sounds cutting]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=557&amp;action=new</link>
			<description><![CDATA[The sounds was OK, nothing the configure changed, but when I start up the PC the sounds was terrible, when I call 
'A' -> 'B'
'A' listen OK but 'B' the sound is cutting

What is the problem?  is it normal this instability?

Thanks<br><br><a rel="nofollow" target="_blank" href="http://www.kqzyfj.com/aa73qgpmgo375A6C77354854CA4"><img src="http://www.awltovhc.com/hb66p59y31NRPUQWRRNPOSPOWUO" alt="Internet Phone Service" border="0"/></a>]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Fri, 25 Nov 2011 21:03:24 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=557&amp;action=new</guid>
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			<title><![CDATA[xvfb problem]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=552&amp;action=new</link>
			<description><![CDATA[I am new to this forum, I did have problem with the ststrunk_Control start, boot, option. it seem that the xvfb having a problem. when take look on the log file  xvfb_01.log, i found that the xvfb having some problem. So when it will be unsuccessful when running the ststrunk_Control start command.

mh@poise-desktop:~/siptosis/log$  more xvfb_01.log 

Backtrace:
0: Xvfb (xorg_backtrace+0x37) [0x81b5b27]
1: Xvfb (0x8048000+0x17128a) [0x81b928a]
2: (vdso) (__kernel_rt_sigreturn+0x0) [0x30540c]
3: Xvfb (fbValidateGC+0x3ce) [0x806588e]
4: Xvfb (0x8048000+0xad8af) [0x80f58af]
5: Xvfb (ValidateGC+0x26) [0x81712f6]
6: Xvfb (0x8048000+0x1122bd) [0x815a2bd]
7: Xvfb (0x8048000+0x115b57) [0x815db57]
8: Xvfb (0x8048000+0x1654c) [0x805e54c]
9: /lib/i386-linux-gnu/libc.so.6 (__libc_start_main+0xf3) [0x14b113]
10: Xvfb (0x8048000+0x16861) [0x805e861]
Floating point exception at address 0x806588e

Caught signal 8 (Floating point exception). Server aborting


on this system,  ststrunk_Control config command is working fine, SipToSis_linux is also working fine.
my system is ubantu 11.10, all of the software are uptodate. I did try to search on this site and find more information regarding with the xvfb without success, so please help me on the issue.]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Mon, 14 Nov 2011 13:33:46 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=552&amp;action=new</guid>
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		<item>
			<title><![CDATA[How to set on the status Skype]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=553&amp;action=new</link>
			<description><![CDATA[How to set on status Skype Online always ?

My siptosis.cfg:

#interval in minutes to set Skype Online Status (0=disabled)
setSkypeOnlineStatusInterval=5
#status to set to when interval reached (Options:ONLINE,DND,AWAY,INVISIBLE,OFFLINE)
skypeOnlineStatus=ONLINE

My Skype version: 4.xxx.xxx.206
Skype configuration: Show me as away when I've been inactive for 0 minutes.
(Skype tells 0 minutes for always Online)

But the status Skype goes to Away after 5 minutes, any idea ?]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Mon, 14 Nov 2011 13:31:49 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=553&amp;action=new</guid>
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			<title><![CDATA[Using button tones]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=551&amp;action=new</link>
			<description><![CDATA[Something I've noticed.  If I call somewhere that has an option to press a button for a service, I can't use it.  For some reasone the button tones aren't working....

I searched here for a possible solution and didn't find any.  Anyone else have this problem?

Scott]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Sat, 05 Nov 2011 14:58:57 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=551&amp;action=new</guid>
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			<title><![CDATA[Sharing my experience....]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=550&amp;action=new</link>
			<description><![CDATA[I had siptosis running on a (vmware) XP machine with a virtual sound card and then piped to my trixbox server.  I had an annoying pause with my broadcast voice that was barely a second but would trip you up when actually talking to someone.

I decided to try siptosis on a standalone machine with a sound card (and less processing power) and the difference was like night and day.  The voice was dead on , split second and functions much better.  I was thinking of putting a sound card in the vmware server but this is working to well to mess with...

now on to trunking....RAH!


2 cents<br><br><a rel="nofollow" href="http://www.kqzyfj.com/3481efolfn26495B662438B84AC" target="_blank"><img src="http://www.awltovhc.com/qe65g04tzxIMKPLRMMIKJOROKQS" alt="Award-Winning Business Class Phone Service" border="0"/></a>]]></description>
			<author><![CDATA[dummy@example.com (solusmib)]]></author>
			<pubDate>Fri, 04 Nov 2011 02:33:18 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=550&amp;action=new</guid>
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		<item>
			<title><![CDATA[No more paid version]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=546&amp;action=new</link>
			<description><![CDATA[Are there something wrong because the paid versions disappear?]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Thu, 03 Nov 2011 12:27:43 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=546&amp;action=new</guid>
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		<item>
			<title><![CDATA[Voice delay problem]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=548&amp;action=new</link>
			<description><![CDATA[I wanted to relay a problem that I've been having with my skype trunk using siptosis.  When someone calls me, there is no delay with incoming or outgoing speech.  If I call someone, my speech is delayed about 4 seconds but the incoming speech is not.

I tried everything I could to run down who or what was causing the problem. 

After reading a few posts here, I reinstalled skype 5.3 and the problem VANISHED.....  way to go skype.

Just 2 cents I guess...]]></description>
			<author><![CDATA[dummy@example.com (solusmib)]]></author>
			<pubDate>Tue, 01 Nov 2011 02:50:31 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=548&amp;action=new</guid>
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			<title><![CDATA[Kernel Panic - snd_pcm]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=511&amp;action=new</link>
			<description><![CDATA[Hello Forum,

I just installed SipToSis and Trunkbuilder a month ago on folowing environments:

- Dell T110, Xeon X3430/4C, 2.4 GHz
- 4 GB RAM
- Dell T110 has no sound crad, So I use snd-dummy (put modprobe snd-dummy in /etc/rc.local)
==================================================================
- CentOS 5.5 64-bit (Elastix 2.0.3)
- I created 8 skype trunks

It works fine. However, I got kernel panic (please refer to attached picture) 3 times already in 
one month running time. I suspect snd-dummy is root cause but don't know how to fix it. Please advise. Thanks a lot for helping.

Regards,
Vitaya

[url=http://voipinvent.com/voip/images/kernel-panic.jpg]Kernel Panic Picture[/url]]]></description>
			<author><![CDATA[dummy@example.com (vitaya)]]></author>
			<pubDate>Mon, 24 Oct 2011 08:09:12 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=511&amp;action=new</guid>
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			<title><![CDATA[Robotic Voice]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=545&amp;action=new</link>
			<description><![CDATA[Hi, I have an SPA3102 with 5.1.10(GW) version, and Skype 5.3..0.111. My SPA is correctly configured as the guide in this site, and my SipToSIs configuration is very simple, infact I made 1 or 2 changes accordly with installation guide.

My only issue is that when I do a Skype call after some minutes I hear a robotic voice while my listener hear me perfectly. I don't find nothing like this in the forum, anyone knows how can I solve this issue?

Thank you very much<br><br><a rel="nofollow" target="_blank" href="http://www.dpbolvw.net/18103dlurlt8CAFBHCC8A9E9DAED"><img src="http://www.ftjcfx.com/g181iw-ousDHFKGMHHDFEJEIFJI" alt="" border="0"/></a>]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Thu, 20 Oct 2011 12:49:40 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=545&amp;action=new</guid>
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			<title><![CDATA[Trouble with SipToSis and Asterisk - bad sound in chan_spy]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=544&amp;action=new</link>
			<description><![CDATA[Hello!

I have a problem with SipToSis channel in Asterisk.
Sound is very good when I make an outgoing call, bu if I use chan_spy, sound get bad.
Can you help me please?

P.S. And sorry for my English]]></description>
			<author><![CDATA[dummy@example.com (MH)]]></author>
			<pubDate>Tue, 18 Oct 2011 01:09:11 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=544&amp;action=new</guid>
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			<title><![CDATA[Siptosis not picking up call from Skype]]></title>
			<link>http://www.mhspot.com/stsforum/viewtopic.php?id=543&amp;action=new</link>
			<description><![CDATA[Hello,  I am testing siptosis on Linux (CentOS 5.6) with PABX in the Flash setup. It is mostly working except one problem: after restart of computer if I try to call Skype user, set up to work with siptosis, it does not answer in most cases. I can clearly hear through headphones that Skype is ringing. If I call from other SIP extension to Skype's test service echo1234 through siptosis, call always goes through. And after that siptosis is accepting incoming calls for some time too. But next day its the same story: siptosis will not pickup call from Skype unless I make SIP to Skype call.]]></description>
			<author><![CDATA[dummy@example.com (vcoolas)]]></author>
			<pubDate>Mon, 17 Oct 2011 03:56:42 +0000</pubDate>
			<guid>http://www.mhspot.com/stsforum/viewtopic.php?id=543&amp;action=new</guid>
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