For this ITSP, registration is ok and inbound calls work. But when making an outbound call the call is sent to the itsp but then gets a fast busy. From what I see in the trace the itsp uses asterisk. I sent a trace in a previous email but it shows nothing I could see missing or in error. The itsp sent the config info below but I have been using them without a STUN server fine on my old PBX. Is there anything that looks like it is out of sync with defaults (like port?). Also, the SIP and DP are below but since the call is sent the DP must be ok. Thanks for any help/suggestions!
From the ITSP:> Sip Proxy - sip.voipyourphone.com STN Server - stun.voipyourphone.com Registration Port - any one port between 5060 and 5069 DTMF port 5004 RTP ports 16384 to 16482 Audio Codecs G711u (account default), G711a, G729a and iLbc DTMF rfc2833 (default), inband, SIP-INFO
SIP: <gateway name="uid1"> <param name="proxy" value="sip.voipyourphone.com"/> <param name="username" value="uid1"/> <param name="from-user" value="uid1"/> <param name="password" value="pwd1"/> <param name="expire-seconds" value="600"/> <param name="register" value="true"/> <param name="retry-seconds" value="30"/> DP <extension name="local.uid1"> <condition field="${toll_allow}" expression="local"/> <condition field="destination_number" expression="^9(\d{7})$"> <action application="set" data="effective_caller_id_number=${outbound_caller_id_number}"/> <action application="set" data="effective_caller_id_name=${outbound_caller_id_name}"/> <action application="bridge" data="sofia/gateway/uid1/1${default_areacode}$1@sip.voipyourlife.com"/> </condition> </extension>