I'm having all sorts of grief trying to get outgoing calls routed to this SIP provider Does anyone have a working trunk configuration that I can use with Freepbx (Trixbox etc.)? After three days we have got incoming calls working but outgoing proves to be elusive. Hint: Outgoing: Peer Details:
context=from-pstn canreinvite=no fromdomain=(domain name) host=(ipaddress) insecure=invite,port type=friend disallow=all allow=ulaw Incoming: User Details: context=from-pstn canreinvite=no fromdomain=(domain name) host=(ipaddress) insecure=invite,port nat=yes secret=***password*** type=user username=647NXXXXXX disallow=all allow=ulaw (there is no registration string required) sip_nat.conf is normal for a natted server but their documentation adds srvlookup=yes Thx H ================================= Henry L.Coleman [www.VoIP-PBX.ca] Tel: 647-723-5160 Ext.203 ================================= --------------------------------------------------------------------- To unsubscribe, e-mail: [email protected] For additional commands, e-mail: [email protected]
