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[Freeswitch-users] VoicePulse Setup Help


 
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jflowers at ezo.net
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PostPosted: Tue Sep 23, 2008 8:19 pm    Post subject: [Freeswitch-users] VoicePulse Setup Help Reply with quote

I've gotten very confused in trying to set VoicePulse up as an outgoing
gateway to the PSTN, after reading all the various wiki sections about
gateways and VoicePulse:

1. Added voicepulse.xml to conf/sip_profiles/default/external..
2. edited conf/dialplan/default.xml to route 10/11 digit calls to VoicePulse.

I get registered with VoicePulse OK and the gateways show up in '>sofia
status' but the routing isn't working.

Obviously, I'm doing something wrong but I'm at a loss at what that is. Can
someone point me in the right direction?

Thank you.

freeswitch.log
--------------
2008-09-22 21:05:41 [DEBUG] mod_enum.c:582 enum_dialplan_hunt() ENUM Lookup
on 13304535344
2008-09-22 21:05:41 [DEBUG] mod_enum.c:208 dnserror() unable to lookup NAPTR
record for 4.4.3.5.3.5.4.0.3.3.1.e164.org: valid domain but no data of
requested type
2008-09-22 21:05:41 [INFO] switch_core_state_machine.c:114
switch_core_standard_on_routing() No Route, Aborting
2008-09-22 21:05:41 [NOTICE] switch_core_state_machine.c:115
switch_core_standard_on_routing() Hangup sofia/internal/1000@192.168.2.2
[CS_ROUTING] [NO_ROUTE_DESTINATION]

voicepulse.xml
--------------
<include>
<gateway name="voicepulse">
<param name="username" value="Qrq22VWD13"/>
<param name="realm" value="jfk-primary.voicepulse.com"/>
<param name="password" value="FQA32wvw77"/>
<param name="proxy" value="nyc-primary.voicepulse.com"/>
<param name="expire-seconds" value="5000"/>
<param name="register" value="true"/>
</gateway>
<gateway name="voicepulse-backup">
<param name="username" value="Qrq24VWC12"/>
<param name="realm" value="jfk-backup.voicepulse.com"/>
<param name="password" value="FQU41wvx77"/>
<param name="proxy" value="nyc-primary.voicepulse.com"/>
<param name="extension" value="1000"/> <!-- inbound calls go to -->
<param name="expire-seconds" value="5000"/>
<param name="register" value="true"/>
</gateway>
</include>

default.xml
-----------
<!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here
-->
<extension name="Long Distance - voicepulse">
<condition field="destination_number" expression="^(1{0,1}\d{10})$">
<action application="set"
data="effective_caller_id_number=13304535344"/>
<!-- If your provider does not provide ringback (180 or 183) you may
simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" -->
<action application="bridge" data="sofia/gateway/voicepulse/$1"/>
</condition>

--
Jim Flowers <jflowers@ezo.net>



--
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msc at freeswitch.org
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PostPosted: Tue Sep 23, 2008 9:31 pm    Post subject: [Freeswitch-users] VoicePulse Setup Help Reply with quote

Be sure that your extension definition is *not* located in default.xml
after the enum extension. It looks like your dial string might not be
matching what you want.

Also, if those are real usernames and passwords for your voicepulse
account then you might want to change them otherwise I will now have
my own VP account! Wink

-MC

Sent from my iPhone

On Sep 23, 2008, at 6:25 PM, "Jim Flowers" <jflowers@ezo.net> wrote:

Quote:
I've gotten very confused in trying to set VoicePulse up as an
outgoing
gateway to the PSTN, after reading all the various wiki sections about
gateways and VoicePulse:

1. Added voicepulse.xml to conf/sip_profiles/default/external..
2. edited conf/dialplan/default.xml to route 10/11 digit calls to
VoicePulse.

I get registered with VoicePulse OK and the gateways show up in
'>sofia
status' but the routing isn't working.

Obviously, I'm doing something wrong but I'm at a loss at what that
is. Can
someone point me in the right direction?

Thank you.

freeswitch.log
--------------
2008-09-22 21:05:41 [DEBUG] mod_enum.c:582 enum_dialplan_hunt() ENUM
Lookup
on 13304535344
2008-09-22 21:05:41 [DEBUG] mod_enum.c:208 dnserror() unable to
lookup NAPTR
record for 4.4.3.5.3.5.4.0.3.3.1.e164.org: valid domain but no data of
requested type
2008-09-22 21:05:41 [INFO] switch_core_state_machine.c:114
switch_core_standard_on_routing() No Route, Aborting
2008-09-22 21:05:41 [NOTICE] switch_core_state_machine.c:115
switch_core_standard_on_routing() Hangup sofia/internal/1000@192.168.2.2
[CS_ROUTING] [NO_ROUTE_DESTINATION]

voicepulse.xml
--------------
<include>
<gateway name="voicepulse">
<param name="username" value="Qrq22VWD13"/>
<param name="realm" value="jfk-primary.voicepulse.com"/>
<param name="password" value="FQA32wvw77"/>
<param name="proxy" value="nyc-primary.voicepulse.com"/>
<param name="expire-seconds" value="5000"/>
<param name="register" value="true"/>
</gateway>
<gateway name="voicepulse-backup">
<param name="username" value="Qrq24VWC12"/>
<param name="realm" value="jfk-backup.voicepulse.com"/>
<param name="password" value="FQU41wvx77"/>
<param name="proxy" value="nyc-primary.voicepulse.com"/>
<param name="extension" value="1000"/> <!-- inbound calls go to -->
<param name="expire-seconds" value="5000"/>
<param name="register" value="true"/>
</gateway>
</include>

default.xml
-----------
<!-- Dial any 10 digit number (2223334444) or 1+10 number
(12223334444) here
-->
<extension name="Long Distance - voicepulse">
<condition field="destination_number" expression="^(1{0,1}\d{10})
$">
<action application="set"
data="effective_caller_id_number=13304535344"/>
<!-- If your provider does not provide ringback (180 or 183)
you may
simulate
ringback by uncommenting the following line. -->
<!-- action application="ringback" -->
<action application="bridge" data="sofia/gateway/voicepulse/$1"/>
</condition>

--
Jim Flowers <jflowers@ezo.net>



--
This message has been scanned for viruses and
dangerous content by MailScanner, and is
believed to be clean.


_______________________________________________
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Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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jflowers at ezo.net
Guest





PostPosted: Tue Sep 23, 2008 11:03 pm    Post subject: [Freeswitch-users] VoicePulse Setup Help Reply with quote

Michael,

Thanks for the reply. That wasn't it but it got me in the right ballpark.
I just found the last extension definition that showed up in the log and
moved my voicepulse extension to follow it. Works great now. I'll figure
out what's broken in the lower part of default.xml (probably a bad comment
placement) some other day but it looks like it is there from loading the
sample data.

If you can figure out which 6 characters I changed in the username and
password, you are welcome to use part of my free 2$ worth of minutes ;<}

I see that you responded from your iPhone (better thumbs than mine). Have
you found a way to use it as a sip-phone yet?


Michael S Collins-2 wrote:
Quote:

Be sure that your extension definition is *not* located in default.xml
after the enum extension. It looks like your dial string might not be
matching what you want.

Also, if those are real usernames and passwords for your voicepulse
account then you might want to change them otherwise I will now have
my own VP account! Wink

-MC

Sent from my iPhone

On Sep 23, 2008, at 6:25 PM, "Jim Flowers" <jflowers@ezo.net> wrote:

Quote:
I've gotten very confused in trying to set VoicePulse up as an
outgoing
gateway to the PSTN, after reading all the various wiki sections about
gateways and VoicePulse:
...

--
View this message in context: http://www.nabble.com/VoicePulse-Setup-Help-tp19640262p19641631.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org
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msc at freeswitch.org
Guest





PostPosted: Tue Sep 23, 2008 11:54 pm    Post subject: [Freeswitch-users] VoicePulse Setup Help Reply with quote

I haven't come across anything that isn't a scam of some sort. Sad

-MC

Sent from my iPhone

On Sep 23, 2008, at 9:01 PM, jflowers <jflowers@ezo.net> wrote:

Quote:

Michael,

Thanks for the reply. That wasn't it but it got me in the right
ballpark.
I just found the last extension definition that showed up in the log
and
moved my voicepulse extension to follow it. Works great now. I'll
figure
out what's broken in the lower part of default.xml (probably a bad
comment
placement) some other day but it looks like it is there from loading
the
sample data.

If you can figure out which 6 characters I changed in the username and
password, you are welcome to use part of my free 2$ worth of
minutes ;<}

I see that you responded from your iPhone (better thumbs than
mine). Have
you found a way to use it as a sip-phone yet?


Michael S Collins-2 wrote:
Quote:

Be sure that your extension definition is *not* located in
default.xml
after the enum extension. It looks like your dial string might not be
matching what you want.

Also, if those are real usernames and passwords for your voicepulse
account then you might want to change them otherwise I will now have
my own VP account! Wink

-MC

Sent from my iPhone

On Sep 23, 2008, at 6:25 PM, "Jim Flowers" <jflowers@ezo.net> wrote:

Quote:
I've gotten very confused in trying to set VoicePulse up as an
outgoing
gateway to the PSTN, after reading all the various wiki sections
about
gateways and VoicePulse:
...

--
View this message in context: http://www.nabble.com/VoicePulse-Setup-Help-tp19640262p19641631.html
Sent from the Freeswitch-users mailing list archive at Nabble.com.


_______________________________________________
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
http://www.freeswitch.org

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http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
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