Hi, Brad: This is Dan Lan. Thanks I got it working. Yes, I have to use "bridge" to connect to Sofia external trunk. I will put my configuration here for mail list archive purpose, so other people can refer to this in the future. 1. First, make an extension profile in the dialplan\public folder as following. 2. This dialplan will recognize the incoming IP trunk and bridge the call to second IP trunk 3. The reason I disable DTMF, is that I hear 2 DTMF tone (maybe one in-band, one RFC2833) so I disable one, and it works for me. <include> <extension name="trunkin_to_trunkout"> <condition field="network_addr" expression="^xxx\.xxx\.xxx\.xxx$"/> <-- The first Trunk's IP address --> <condition field="destination_number" expression="^(\d+)$"> <action application = "set" data="dtmf_type=none"/> <action application="bridge" data="sofia/external/$1@xxx.xxx.xxx.xxx"/> <-- The second Trunk's IP address --> </condition> </extension> </include> On Tue, Aug 30, 2011 at 4:10 PM, Brad Mina <brad@tech21.com> wrote:
1. Username and password are manditory in the XML - you don't have to put anything that makes sense, just fill it in with your DID or something and keep register=false and those details will never do anything.
2. bridge would be the proper tool, you might have to mess around with proxy media or make sure proxy_media is off to ensure the data is coming from you directly and not negotiated between providers.
On Tue, Aug 30, 2011 at 3:57 PM, Dan Lan <danlanweb@gmail.com> wrote:
Hi,
I want to use FS to accept call from SIP_TrunkA and terminate to SIP_TrunkB Both SIP trunks are using IP authentication, no need for username and password.
for incoming call (SIP_TrunkA), I have add the IP address of SIP_TrunkA in to acl.conf.xml <node type="allow" cidr="xxx.xxx.xxx.xxx/32"/>
I understand the incoming call will go to the public context, so I think I need to do something here.
I dont know what to do next. 1. I try to establish a gateway for SIP_TrunkB for my outgoing call, but sofia require me to have the username and password for the trunk. I dont know where to add the SIP_TrunkB in freeswitch, since the provider of SIP_TrunkB only need to recoginize my FS IP address. 2. After I establish SIP_TrunkB, how should I do on public dialplan to route the call from SIP_TrunkA to SIP_TrunkB? should I use "transfer" or "bridge", could I make a dialplan that can route all the call from IP address of A to IP address of B?
Sorry for the newbie question, I try to look up on wiki but only got partial information for me.
Any help and any directions or hints are appreciated
Dan Lan
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