Asterisk Dahdi Creating Sip Trunk
I have made a trunk to make calls within my sip group (i.e) sip.antisip.comNow I want to make calls to another sip network (i.e) sip.fairytel.at.
Once a Trunk has been created, you should next create an Inbound Route in order to handle calls coming from Digium's SIP Trunking service to your FreePBX system. To make these configuration changes, visit the Connectivity - Inbound Routes page. To configure a Digium SIP Trunking account, make modifications to the following options. Short for “Digium Asterisk Hardware Device Interface”. (this will generate /etc/dahdi/system.conf and /etc/asterisk/dahdi-channels.conf) asterisk:#. An SLA trunk is a mapping between a virtual trunk and a real Asterisk device. This device may be an analog FXO line, or something like a SIP trunk. A trunk must be configured in two places. First, configure the device itself in the channel specific configuration file such as dahdi.conf or sip.conf.
I know we need to make dedicate trunks for these, but I am not sure of the configurations that I should make.
For making calls with antisip I tried the context of the trunk as default or from-pstn. When I call from an extension to external antisip number, the call is connected but gets cut immediately. I am not sure what I am doing wrong here ? I have given the server name, port, username, secret and selected the codecs as ulaw and alaw. I have connected this to a outbound route with dialplans as 4XXXXXXX as the number which I am testing now is that. I gave a password to check if the outbound route works correctly and it works.
For making calls to fairytel, I am using a trunk with the same credentials but the context is from-sip-external. I have connected this to another new outbound route with a dialplan of 4NNNXNNNN1X which exactly matches my fairytel number. I am not sure, if I am going completely wrong somewhere or not ?
1 Answer
Freepbx Sip Trunk Configuration
I did some changes to make this work finally,
I setup a trunk with antisip credentials and another trunk with fairytel credentials The context in both cases were from-pstn and codecs were ulaw and alaw only. Later, I created two outbound routes.One was called antisip-outbound and it has the dial pattern of fairytel in it (i.e) 479XXXXX and the trunks were in the order antisip and then fairytel. But, I think we don't need antisip here. Then. I created another outbound route call fairytel-outbound which has the dial pattern of anti-sip (i.e) 431XXXXXXX. The trunks for this were fairytel first, then antisip. Now, if I make a call from any extension, it knows if it should connect to the fairytel trunk or antisip trunk and makes the call correctly. This way, I got two different providers connected inside my freePBX.
Freepbx Dahdi
