:: Matthew Simpson :: :: December 05, 2005 11:34 AM :: (6055) TxLink Termination Customers -- Please read this global ticket in its entirety! We have added a new feature to VC that allows you to provision your own SIP termination accounts. You can add and remove IP addresses using VC that will be allowed to send SIP termination traffic. To access this feature click "My Tools", then "SIP IP Addresses". You can put in a ip address or a hostname. Currently this system updates every hour at 47 minutes after the hour so any IPs you add won't work for termination until then. If you are an IAX account, please convert to SIP as soon as possible. IAX does not work well between asterisk versions [we currently run 1.2.0 stable] and does not scale well causing call quality issues under heavy loads. All you need to do is add your SIP ip address in VC. Once it is active, change your Dial string to Dial(SIP/${EXTEN}@sip.txlink.net). Make sure that ${EXTEN} is 10 digits, don't add the "1" in front of it. In sip.conf you can create a peer entry for us if you want, but that is not required. You are not required and cannot sip register with our servers! We authenticate you by IP address. If all your endpoints are in public IP space you can use canreinvite=yes in sip.conf. This will allow asterisk to reinvite the endpoint directly to our endpoint. SIP signalling will still go through your asterisk server so your CDRs will be correct, but your server load and call quality will significantly improve because asterisk no longer has to p! roxy the RTP voice media. If you have endpoints behind nat or have issues with one way audio or dropped calls after the other end picks up, try canreinvite=no in your sip.conf instead. If you have IAX DIDs, after you have added your SIP ip addresses into VC, click "Routing" under DIDs and Ordering then choose "sip", input your DIDs, and choose the SIP ip address you wish to route the DIDs to. Then click "Route DIDS'. Unlike termination, re-routing DIDs takes effect IMMEDIATELY. If you ever have a sip server go down you can re-route your DIDs on the fly using VC. To configure SIP DIDs you will want to go into your sip.conf file and look at the [general] tab. This is the default tab that all unauthenticated SIP calls will go to. DO NOT CREATE an entry for TxLink in sip.conf! Instead, look at the context= field under [general]. In extensions.conf, find that context and create a new context in extensions.conf and include => that context in the context referenced in sip.conf's [general] field. Example: sip.conf --------- [general] disallow=all allow=ulaw context=noauth rtptimeout=60 rtpholdtimeout=300 tos=reliability port=5060 qualify=no dtmfmode=rfc2833 trustrpid=yes canreinvite=yes nat=no In this example "noauth" is the default context. So, if my DID is 214-555-1212 in extensions.conf I would: [noauth] include => mydids [mydids] exten => 2145551212,1,Answer exten => 2145551212,2,Playback(welcome) exten => 2145551212,3,Hangup We have redundant systems in place in case of any host failure on this end. sip.txlink.net is the default and will return the "best" available IP. if you have problems with that hostname you can use sip-dfw1.txlink.net or sip-dfw2.txlink.net as well. If you have questions about the VC interface, please email Chris Sigler at chris.sigler@txlink.net. If you have other questions please email me Thanks, Matthew :: Matthew Simpson :: :: December 05, 2005 12:05 PM :: (6057) I should add that if you are currently sending traffic to a commpartners Acme trunk group directly you do not need to change anything. But, if you are sending us traffic via IAX, or you are currently sending traffic to us using a 6 digit account code, please follow the above instructions. :: Matthew Simpson :: :: December 14, 2005 3:31 PM :: (6159) IAX customers, we are considering December 31, 2005 at 5:00pm to be the turn-off date for IAX. If this is a problem for anyone please contact me directly at matthew@txlink.net. Please switch to SIP, if you need help, email me and I will assist. :: Matthew Simpson :: :: December 27, 2005 10:27 AM :: (6307) Closing this ticket. If you have any problems please email voip email or open a trouble ticket with Network Operations. Thank you!