How to Use MySipSwitch

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Aaron
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Post by Aaron » Wed Dec 12, 2007 2:04 am

emoci wrote:Made some changes to the original post, especially in regards to optional features of the Dial Plan expression...

If Aaron or gbonnet has some time, do read over it to make sure I didn't mess anything up
Hi emoci,

Thanks a lot for that you've distilled it nicely!

One very minor point is that "Natural Expressions" should be "Regular Expressions".

Also it would be woth mentioning the Trace option as this can help quickly isolate issues if a problem is encountered on a call.

exten => _X.,1,SwitchCall(user,pass,${dst}@sip.blueface.ie,From User, Server, True)

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 :).

Thanks again,

Aaron

gbonnet
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Post by gbonnet » Wed Dec 12, 2007 8:33 am

Hi Emoci,

Thanks a lot for helping like that. That surely is useful to everyone.

;)
Blueface [url=http://www.blueface.ie/]Phone[/url] Service

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TheFug
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Post by TheFug » Wed Dec 12, 2007 6:21 pm

Yeah...... wow, i've got something to do, to translate that for the Dutch version :shock:
Thanks, The Fug.

gear: my ISP's Zyxel Modem/Router in bridge, Sitecom WL309 Router, Siemens Gigaset 301D

jj
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Post by jj » Tue Jan 15, 2008 2:37 pm

Aaron wrote: 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-mail address (and received a confirmation e-mail to this address) but I do not get any e-mails with debug info despite the last parameter is set to True.

Any idea what I can do to make it work?

gbonnet
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Post by gbonnet » Tue Jan 15, 2008 3:21 pm

Hi JJ

If you are not using the optional parameters, you need to make sure you have the right number of comas, for instance: ${EXTEN}@sip.blueface.ie,,,True) is working fine for me.
Let me know if you are still not receiving anything.
Blueface [url=http://www.blueface.ie/]Phone[/url] Service

satphoneguy
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Post by satphoneguy » Tue Jan 15, 2008 5:16 pm

gbonnet wrote:Hi JJ

If you are not using the optional parameters, you need to make sure you have the right number of comas, for instance: ${EXTEN}@sip.blueface.ie,,,True) is working fine for me.
Let me know if you are still not receiving anything.
my experience is that the call trace does not work with the switchcall function but only for calls from a registered device. it would be nice if switchcall could be added.

spg

jj
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Post by jj » Tue Jan 15, 2008 8:56 pm

satphoneguy wrote: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 there was no e-mail.

I have just reconfigured my device and tried this with an outgoing rule and it works! Thanks a lot!

usf
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do not ring

Post by usf » Sun Jan 20, 2008 3:37 am

I follow the set up to my pap2 ( contact sip:myuserid@mysipswitch.com), I use gizmoproject call my mysipswitch, but it (ATA) do not ring.
mysipswitch call out function, it works.
Have any debug method to find out what is wrong?

emoci
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Re: do not ring

Post by emoci » Wed Jan 30, 2008 6:15 pm

usf wrote:I follow the set up to my pap2 ( contact sip:myuserid@mysipswitch.com), I use gizmoproject call my mysipswitch, but it (ATA) do not ring.
mysipswitch call out function, it works.
Have any debug method to find out what is wrong?
Just make sure of the following:

On your ATA:
Username: Your MySipSwitchUsername
Password: Your Password
Host/Proxy: sip.mysipswitch.com
Make sure to have the ATA register with the server

Second, inside MySipSwitch (if I get this correctly you want Gizmo to ring MySipSwitch):
Perform a registration for Gizmo and point it to username@sip.mysipswitch.com for the contact address

satphoneguy
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Joined: Tue Oct 23, 2007 12:54 pm

Re: do not ring

Post by satphoneguy » Wed Jan 30, 2008 7:10 pm

usf wrote:I follow the set up to my pap2 ( contact sip:myuserid@mysipswitch.com), I use gizmoproject call my mysipswitch, but it (ATA) do not ring.
mysipswitch call out function, it works.
Have any debug method to find out what is wrong?
there appears to be a recent change at gizmo and you have to use your gizmo number(1747xxxxxxx) instead of your username on registrations. all my gizmo accounts failed until i made the change; now they are working again.

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