Kindly confirm Asterisk 15.5 will forward call-Hold Resume reINVITE/UPDATE to upstream network. Test setup and Observations are as follows:
User A is registered with Asterisk(15.5). Asterisk is communicating to Service provider network over SIP trunk (chan_sip channel) with Media Bypass is enabled.
Users and SIP trunk is configured with CANREINVITE=YES,
And, sip trunk is enabled for media bypass using following configurations:
Settings -> Asterix SIP settings -> ‘Reinvite Behavior=yes’
Added below entries to Other SIP Settings
–> ‘directrtpsetup=yes’ and
Test scenario is PBX user calls PSTN user over SIP trunk and PBX user places call on hold.
• Voice call established between PBX endpoint(6.5.x.x:19010) and PSTN endpoint(172.22.x.x:6264)
• PBX endpoint places the call on hold
• Phone sends Hold INVITE to PBX with media attribute a= sendonly with Connection Info as Phone IP
• PBX responds with 200OK/SDP media attribute a= recvonly with Connection Info as Service Provider network IP
• Asterisk did not forwards Hold INVITE to Service Provider
• PBX(10.64.X.X) sends Music on hold to Service provider network in the already established RTP port (172.22.x.x:6264)
• Since Service Provider network did not get any notification from Asterisk on call on hold, Service Provider network rejects the RTP received from PBX (10.64.X.X)
• Could not hear MOH plays at PSTN endpoint
• PBX endpoint goes on hold, however PSTN endpoint will be in active session
• When call resumes both endpoints continues the session with 2-way audio.
Kindly confirm Asterisk 15.5 will forward call-Hold Resume INVITE/UPDATE to upstream network. And please help me if I missed any configuration.
Thanking you in advance.