Outbound dialing to SPA3102 behind NAT
Hi, I've currently got an Asterisk server running at home but want to switch to FS on my external server (located in a DC). Configuration: - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1) - Couple of voip phones The Asterisk server is running on my local network as are all the other devices; everything is connected to a DSL-router with NAT. The SPA3102 are configured as Trunks in Asterisk with the following configuration: allow=ulaw canreinvite=no context=from-trunk disallow=all dtmfmode=rfc2833 host=dynamic incominglimit=1 nat=never port=5061 qualify=yes secret=trunk1 username=trunk1 type=friend The SPA3102 are configured to register their PSTN-line to the Asterisk server. Outbound routes are configured to use 1 of the 2 Trunks. I want to do the same thing with my FS running on a server in the DC. I've setup all the devices and can make internal calls. Inbound calls on the PSTN lines are also forwarded to the correct voip-devices. I just cannot figure out how to configure outbound dialing to the PSTN-lines. The Wiki (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I should configure them as following: <include> <extension name="To PSTN"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/internal/${destination_number}@[IP address of SPA3102]:5061" /> </condition> </extension> </include> The problem here is that the SPA3102 are behind my router so the only possibility would be to setup some port forwarding to on the router with different ports for the 2 devices. I would rather not do that so I was wondering if there is an easier/better way to do that. Any ideas? Rgds, N.
Only way I can think of is have the spa3102 register as an fs gateway using user name and password and bridge to the gateway -- and make sure natis enabled on the 3102, so fs can detect its external ip address. Nick Rosier <nick.rosier@gmail.com> wrote:
Hi,
I've currently got an Asterisk server running at home but want to switch to FS on my external server (located in a DC).
Configuration: - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1) - Couple of voip phones
The Asterisk server is running on my local network as are all the other devices; everything is connected to a DSL-router with NAT. The SPA3102 are configured as Trunks in Asterisk with the following configuration: allow=ulaw canreinvite=no context=from-trunk disallow=all dtmfmode=rfc2833 host=dynamic incominglimit=1 nat=never port=5061 qualify=yes secret=trunk1 username=trunk1 type=friend
The SPA3102 are configured to register their PSTN-line to the Asterisk server. Outbound routes are configured to use 1 of the 2 Trunks.
I want to do the same thing with my FS running on a server in the DC. I've setup all the devices and can make internal calls. Inbound calls on the PSTN lines are also forwarded to the correct voip-devices. I just cannot figure out how to configure outbound dialing to the PSTN-lines. The Wiki (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I should configure them as following:
<include> <extension name="To PSTN"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/internal/${destination_number}@[IP address of SPA3102]:5061" /> </condition> </extension> </include>
The problem here is that the SPA3102 are behind my router so the only possibility would be to setup some port forwarding to on the router with different ports for the 2 devices. I would rather not do that so I was wondering if there is an easier/better way to do that.
Any ideas?
Rgds, N.
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici@ccs.covici.com
I think there is no need to enable nat on endpoint device, since freeswitch is able to detect correct address from SIP/RTP messages if request is received from networks in nat acl. 2011/3/20 <covici@ccs.covici.com>
Only way I can think of is have the spa3102 register as an fs gateway using user name and password and bridge to the gateway -- and make sure natis enabled on the 3102, so fs can detect its external ip address.
Nick Rosier <nick.rosier@gmail.com> wrote:
Hi,
I've currently got an Asterisk server running at home but want to switch to FS on my external server (located in a DC).
Configuration: - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1) - Couple of voip phones
The Asterisk server is running on my local network as are all the other devices; everything is connected to a DSL-router with NAT. The SPA3102 are configured as Trunks in Asterisk with the following configuration: allow=ulaw canreinvite=no context=from-trunk disallow=all dtmfmode=rfc2833 host=dynamic incominglimit=1 nat=never port=5061 qualify=yes secret=trunk1 username=trunk1 type=friend
The SPA3102 are configured to register their PSTN-line to the Asterisk server. Outbound routes are configured to use 1 of the 2 Trunks.
I want to do the same thing with my FS running on a server in the DC. I've setup all the devices and can make internal calls. Inbound calls on the PSTN lines are also forwarded to the correct voip-devices. I just cannot figure out how to configure outbound dialing to the PSTN-lines. The Wiki (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I should configure them as following:
<include> <extension name="To PSTN"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/internal/${destination_number}@[IP address of SPA3102]:5061" /> </condition> </extension> </include>
The problem here is that the SPA3102 are behind my router so the only possibility would be to setup some port forwarding to on the router with different ports for the 2 devices. I would rather not do that so I was wondering if there is an easier/better way to do that.
Any ideas?
Rgds, N.
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-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.com
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-- Best regards, Dmitry Sytchev, IT Engineer
I've using Bluebox to configure the FS. I've defined a gateway as following: <gateways> <gateway name="trunk1"> <param name="realm" value=""/> <param name="username" value="trunk1"/> <param name="password" value="trunk1"/> <param name="register" value="false"/> </gateway> </gateways> sofia status show the SPA3102: Call-ID: abb17a79-760cd2dc@192.168.1.153 User: trunk1@pbx.domain Contact: Trunk1 <sip:trunk1@<internal-ip>:5078;transport=tcp;fs_nat=yes;fs_path=sip%3Atrunk1%40<external-ip>%3A5078%3Btransport%3Dtcp> Agent: Linksys/SPA3102-5.1.10(GW) Status: Registered(TCP-NAT)(unknown) EXP(2011-03-20 23:04:09) EXPSECS(2795) Host: pbx.domain IP: <external-ip> Port: 5078 Auth-User: unknown Auth-Realm: pbx.domain MWI-Account: trunk1@pbx.domain I can see the dialplan matching but nothing happens; from the log it looks like the dialing fails instantly: 2011-03-20 22:14:47.713322 [DEBUG] sofia.c:4581 Channel sofia/sipinterface_1/ entering state [calling][0] 2011-03-20 22:14:47.713322 [DEBUG] sofia.c:4581 Channel sofia/sipinterface_1/ entering state [terminated][503] Any ideas what else I need to do? Rgds, N. On 20 March 2011 10:38, <covici@ccs.covici.com> wrote:
Only way I can think of is have the spa3102 register as an fs gateway using user name and password and bridge to the gateway -- and make sure natis enabled on the 3102, so fs can detect its external ip address.
Nick Rosier <nick.rosier@gmail.com> wrote:
Hi,
I've currently got an Asterisk server running at home but want to switch to FS on my external server (located in a DC).
Configuration: - 2 SPA3102 connected to 2 PSTN lines and 2 phones (on Line1) - Couple of voip phones
The Asterisk server is running on my local network as are all the other devices; everything is connected to a DSL-router with NAT. The SPA3102 are configured as Trunks in Asterisk with the following configuration: allow=ulaw canreinvite=no context=from-trunk disallow=all dtmfmode=rfc2833 host=dynamic incominglimit=1 nat=never port=5061 qualify=yes secret=trunk1 username=trunk1 type=friend
The SPA3102 are configured to register their PSTN-line to the Asterisk server. Outbound routes are configured to use 1 of the 2 Trunks.
I want to do the same thing with my FS running on a server in the DC. I've setup all the devices and can make internal calls. Inbound calls on the PSTN lines are also forwarded to the correct voip-devices. I just cannot figure out how to configure outbound dialing to the PSTN-lines. The Wiki (http://wiki.freeswitch.org/wiki/SPA3102_FreeSwitch_HowTo) shows I should configure them as following:
<include> <extension name="To PSTN"> <condition field="destination_number" expression="(.*)"> <action application="bridge" data="sofia/internal/${destination_number}@[IP address of SPA3102]:5061" /> </condition> </extension> </include>
The problem here is that the SPA3102 are behind my router so the only possibility would be to setup some port forwarding to on the router with different ports for the 2 devices. I would rather not do that so I was wondering if there is an easier/better way to do that.
Any ideas?
Rgds, N.
_______________________________________________ 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
-- Your life is like a penny. You're going to lose it. The question is: How do you spend it?
John Covici covici@ccs.covici.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
participants (3)
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covici＠ccs.covici.com
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Dmitry Sytchev
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Nick Rosier