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Vladimir Toncar (Kerio)

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Hi,

This thread will contain several mini-howtos about connecting Operator to SIP providers.

If you have some experience (good or bad) with SIP providers, feel free to share it with the group in this thread as well. Why is the particular provider's service good? Was the configuration in Operator straightforward? Which provider should Kerio test against?

Vladimir

[Updated on: Fri, 04 November 2011 10:41]

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Vladimir Toncar (Kerio)

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This how-to explains how to configure a Voicepulse.com account in Operator

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Vladimir Toncar (Kerio)

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This is a how-to for sipgate.co.uk.

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Vladimir Toncar (Kerio)

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A mini how-to for sipgate.com

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Vladimir Toncar (Kerio)

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A how-to for 802.cz

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steinham

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What You need:
- your phone number registered with ha-vel.cz
- your password
- SIP proxy address (curently ustredna.ha-vel.cz)

All informations you can find at https://ha-loo.ha-vel.eu/cz/index.php after login (section "Information / Informace" and ("SIP/IAX Settings / Nastaveni SIP/IAX")

1) login to Kerio Operator admin interface, go to "Call Routing" section and press "Add a SIP Interface" button

2) name your new interface and enter your assigned phone number at 1-st screen; press "Next" button

3) select desired internal extension (queue, script, conference or group) and optionally fill some outbound preffix at 2-nd screen; press "Next" button

4) enter "ustredna.ha-vel.cz" into "Hostname or IP address" field, eave default port (5060), assigned phone number into "Username" field and password into "Password" field; press "Finish" button

5) open route again, go to "Codecs" tab and remove unsupported codecs (remove SpeeX, G.722 and G.726; press "OK" button

[Updated on: Tue, 09 November 2010 11:36]


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Martin Steinhauser
tester
Kerio Technologies
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steinham

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What You need:
- your phone number registered with fayn.cz
- your password
- SIP proxy address (curently sip.fayn.cz)

All informations you can find at https://iz.fayn.cz/ after login (section "Přehled MSN" and "MSN" button)

1) login to Kerio Operator admin interface, go to "Call Routing" section and press "Add a SIP Interface" button

2) name your new interface and enter your assigned phone number at 1-st screen; press "Next" button

3) select desired internal extension (queue, script, conference or group) and optionally fill some outbound preffix at 2-nd screen; press "Next" button

4) enter "sip.fayn.cz" into "Hostname or IP address" field, leave default port (5060), assigned phone number (MSN) into "Username" field and password into "Password" field; press "Finish" button

5) open route again, go to "Codecs" tab and correct supported codecs (see http://www.fayn.cz/pece-a-podpora/navody-a-nastaveni/); press "OK" button


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Martin Steinhauser
tester
Kerio Technologies
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steinham

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What You need:
- your phone number registered with netphone.cz
- your password
- SIP proxy address (curently sip1.netphone.cz)

All informations you can find at https://admin.netphone.cz/prihlaseni after login (section "Přehled tel. cisel" and "MSN" button)

1) login to Kerio Operator admin interface, go to "Call Routing" section and press "Add a SIP Interface" button

2) name your new interface and enter your assigned phone number at 1-st screen; press "Next" button

3) select desired internal extension (queue, script, conference or group) and optionally fill some outbound preffix at 2-nd screen; press "Next" button

4) enter "sip1.netphone.cz" into "Hostname or IP address" field, leave default port (5060), assigned phone number (MSN) into "Username" field and password into "Password" field; press "Finish" button

5) open route again, go to "Codecs" tab and correct supported codecs (see your netphone.cz account, "Prehled tel. cisel" section, "Upravit" button and enable "Expert mod" checkbox); press "OK" button


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Martin Steinhauser
tester
Kerio Technologies
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steinham

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What You need:
- your phone number registered with xphone.cz
- your password
- SIP proxy address (curently sip2.xphone.cz)

All informations you can find at http://www.xphone.cz after login ("Nastaveni / Settings", section "Prihlasovaci udaje / login credentials")

1) login to Kerio Operator admin interface, go to "Call Routing" section and press "Add a SIP Interface" button

2) name your new interface and enter your assigned phone number at 1-st screen; press "Next" button

3) select desired internal extension (queue, script, conference or group) and optionally fill some outbound preffix at 2-nd screen; press "Next" button

4) enter "sip2.xphone.cz" into "Hostname or IP address" field, leave default port (5060), enter your USERNAME *not* phone number into username field and password into "Password" field

5) enable "User ID differs..." checkbox and enter your USERNAME into its field; press "Finish" button

6) open route again, go to "Codecs" tab and correct supported codecs (see your xphone.cz account: https://www.xphone.cz/zone_user_settings_codecs.php); press "OK" button


______________________________
Martin Steinhauser
tester
Kerio Technologies
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pechan

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Any info for broadvoxDOTcom
thanks!
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Vladimir Toncar (Kerio)

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Hi,

No, we have not tried broadvox so far. If there are instructions for a generic Asterisk, it should be easy to figure it out though.

Aside from knowing the right SIP server name, authentication ID and password, the two most important questions are: (1) What format of telephone number do I need to use? (2) Is the SIP user ID the same as the phone number or is it something else?

Vladimir



Vladimir
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zuki

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We currently use numbergroup.com on our operator setup, works perfectly and was easy to setup.
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gahchoung

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been testing net2phone for a few weeks on operator , and having trouble with some codecs , seems net 2 phone requires g729.

guys at net2phone trying to find a workaround, maybe kerio people can work together with them.
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joyroony

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Hello friend

Thanks for information.
zuki

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We have been using numbergroup for several months now, we have had no problems and it is easy to set up.
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