Search found 18 matches

by jj
Tue Jan 15, 2008 8:56 pm
Forum: Technical Support
Topic: How to Use MySipSwitch
Replies: 42
Views: 109797

my experience is that the call trace does not work with the switchcall function but only for calls from a registered device. Thanks for the info! I tried this debugging function previously with my incoming rule which forwards the call to a provider. My phone is registered to this provider. And ther...
by jj
Tue Jan 15, 2008 8:49 pm
Forum: Getting Started
Topic: P-src-ip (for Betamax users)
Replies: 57
Views: 32134

You can see if you have a device currently registered by checking the Monitoring screen. You mean the "status" info box, right? I guess it is this box because I can see the "Not logged in" message there when my phone is registered to my provider directly. But when my phone is registered to mysipswi...
by jj
Tue Jan 15, 2008 2:37 pm
Forum: Technical Support
Topic: How to Use MySipSwitch
Replies: 42
Views: 109797

By setting the last parameter as True a trace of the call will be recorded and emailed to the sipswitch owner of the call after it's complete. Providing a trace of a problematic call mkaes it a lot easier to troubleshoot :). It seems that this debug option does not work for me. I configured my e-ma...
by jj
Fri Jan 04, 2008 10:15 pm
Forum: Getting Started
Topic: Caller ID lost for incoming calls
Replies: 5
Views: 2766

Thanks! I did not read the sticky carefully. But the rule needed a small change in my case. I had to remove the ${fromname} part, i.e. the final rule looks as follows: exten => <USERNAME1>,1,SwitchCall(,,USERNAME2@sip.voipdiscount.com,<sip:${fromuriuser}@sip.mysipswitch.com>) With the above modifica...
by jj
Wed Jan 02, 2008 9:28 am
Forum: Getting Started
Topic: Caller ID lost for incoming calls
Replies: 5
Views: 2766

Has it been always that way or is this something new ? Always - since December 09, 2007 when I created this rule because of "FUP exceeded problems" on voipdiscount (see http://www.mysipswitch.com/forum/viewtopic.php?p=985#985) - and thank you for your previous help in that thread but, unfortunately...
by jj
Tue Jan 01, 2008 11:37 pm
Forum: Getting Started
Topic: Caller ID lost for incoming calls
Replies: 5
Views: 2766

Caller ID lost for incoming calls

It seems that caller identification is lost for incoming calls forwarded to another provider. More precisely, a rule in my dialplan of mysipswitch is as follows: exten => <USERNAME1>,1,SwitchCall(,,USERNAME2@sip.voipdiscount.com) This rule forwards all incoming calls to my account at voipdiscount. M...
by jj
Tue Jan 01, 2008 9:11 pm
Forum: Getting Started
Topic: P-src-ip (for Betamax users)
Replies: 57
Views: 32134

It means if Betamax are filtering on the P-src-ip header then your direct traffic and your sipswitch traffic will be added together! Aaron, Please explain it in a bit more detail. I have been unable to use mysipswitch for the last few weeks due to "FUP Exceeded" charges on sip.voipdiscount.com. I g...
by jj
Wed Dec 12, 2007 9:33 pm
Forum: Getting Started
Topic: P-src-ip (for Betamax users)
Replies: 57
Views: 32134

Hi Aaron,

Can you add voipdiscount server to the list? It seems I faced the same problem with this server - see my description: http://www.mysipswitch.com/forum/viewto ... ?p=985#985

Thanks!
by jj
Mon Dec 10, 2007 3:50 pm
Forum: News
Topic: VoipBuster (and other Betamax services) Petition
Replies: 14
Views: 14891

My problem with Betamax and mysipswitch is that my recent outgoing calls directed via mysipswitch and voipdiscount.com are marked on voipdiscount's billing as "FUP (Fair Use Policy) exceeded" and are charged with the normal rate, instead of being free. When I register to voipdiscount directly (witho...
by jj
Mon Nov 12, 2007 8:30 pm
Forum: Technical Support
Topic: No audio for incoming calls
Replies: 15
Views: 7523

Oops! MYSIPSWITCH incoming audio started working after I enable stun server stun.xten.com in my ATA. I am 100% sure. I did not use any STUN server before NOV 2 2007. Later I tried to use other public stun servers like fwd etc., It did not work. Now it works only with xten stun server. I am not sure...