We have an Asterisk server which receives a call from a SIP telephone, and forwards it on to another destination using Dial(). So are two SIP calls bridged together by Asterisk. At the moment it appears that Asterisk doesn’t forward early RTP media from the caller while the second call is ringing - is there any way to change that? We’re using Asterisk 13 and chan_sip. Thanks in advance!
Even an obsolete version like 13 will forward early media, providing there is only one outgoing leg being attempted and the caller isn’t being “entertained” (m or r options on Dial). You may need to explicitly call Progress(), but that might just be a rumour.
It generally won’t work for calls arriving from public networks, as it allows services to be provided without someone to bill.
It won’t work with late offer SDP, as used by CUCM.
Thank you David. I’ll check those Dial options, and if they’re not used then look at using Progress().
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