Hi, i have some problems to make a integration between my asterisk and a freeswitch pabx, both sides can “see” each other on sip show peers, but we can’t make calls for the numbers who configurated, someone has maded a integration like this sometime?
Thanks
SIP is SIP, it’s all in how you configure it. You’ll need to be specific about what you are trying that isn’t working and the output you get.
Thks for answer jcolp, my problem is: i trying to integrate a freeswitch with a asterisk using a trunk.I can register a trunk in my network, i created the extensions in sip.conf and extensions.conf, but when i try to connect with the freeswitch(they aren’t in the same network), i receive errors like: CHAINUNNAVAILABLE
When debug mode is active on asterisk cli i get this message:
Dialplan: sofia/external/5555@XXX.XXX.XX.XXX parsing [public->XXXXXXXXXX] continue=false
Dialplan: sofia/external/5555@XXX.XXX.XX.XXX Regex (FAIL) [XXXXXXXXXX] destination_number(5000) =~ /^(XXXXXXXXX)$/ break=on-false
Dialplan: sofia/external/5555@XXX.XXX.XXX.XXX parsing [public->XXXXXXXXXX] continue=false
Anyone can see this sometime?
I hide IPs and telephone numbers for security reasons.
Thanks
I don’t work on Freeswitch and have no experience configuring it, so I can’t comment or provide help on that.
Ok jcolp, thanks for the answer.