On Mon, May 15, 2017 at 4:57 PM, Steve Murphy <[email protected]> wrote:
> Hmmm, according to your refs, none really apply to this situation. > <snip> > > the callerid of the target phone is set in the pjsip channel driver > config, not in my dialplan (the same as chan_sip): > > And, my dialplan doesn't care about the callerid info for the phone you > are dialing... in chan_sip, I get it via the 180 Ringing, but in pjsip, I > am given useless information instead. I don't need it to change the sip > exchanges to a re-invite, either. The phone is able to pick it up from the > 180 Ringing just fine. > > Here is the config for the endpoints, a little cut down: > > [t12] ; Yealink T49G mac=00:15:65:... > type=endpoint > auth=t12 > transport=transport-udp > aors=t12 > <snip> > callerid="Steve" <101> > > [t13] ; Yealink T48G mac=00:15:65:... > type=endpoint > auth=t13 > transport=transport-udp > aors=t13 > <snip> > callerid="s2 test" <102> > > Since the config holds the callerids for each endpoint, I don't have any > code to do lookups to get the callerid of the target. chan_sip has been > fine providing it to the phone via the 180 Ringing... > > Right now, I'm running the exact same dialplan for chan_sip and pjsip. Are > you telling me that I have to change the dialplan for pjsip? Can you give > me a solid example of what I'd need to do in the dialplan to get the same > effect? As a matter of fact, I run the two channel drivers on two > different ports, and I have phones on pjsip, and phones on chan_sip at the > same time... > There is something in your setup setting the wrong information. My test setup does send the correct information for the 180 Ringing message. I am not running the two SIP channel drivers at the same time. For a simple dialplan you should only need to do a dial when PJSIP/t12 calls 102: exten = 102,1,NoO() same = n,Dial(PJSIP/t13) same = n,Hangup() You don't need to specify the transport in the endpoint config as the transport can be determined from the defined transports. There should be a send_rpid=yes somewhere in the endpoint config. Are you dialing between chan_sip and chan_pjsip channels for your testing? You could expand upon the basic dial by adding an interception routine [1] that just shows you the values of CALERID(all) and CONNECTEDLINE(all) when the routine runs. Richard [1] https://wiki.asterisk.org/wiki/display/AST/Party+ID+Interception+Macros+and+Routines
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