I'm trying to set up a dial group using an IP phone (Grandstream GXP 2020) and a phone via Handy-Tone 386, but there seems to be an issue with the Handy-tone. Routing a DID directly to either extension works without a hitch, and both phones ring using the following dial group: <extension name="group_dial_test"> <condition field="destination_number" expression="^2000$"> <action application="set" data="hangup_after_bridge=true"/> <action application="bridge" data="group/test@${domain_name}"/> </condition> </extension> Incidentally, I've also seen the following syntax, is there a difference? <action application="bridge" data="${group_call(test@${domain_name}+A)}"/> The only problem is that when the call is answered by the phone connected to the Handy-Tone FXS port, neither end has audio. The GXP 2020 works in all cases.
Adding the following seems to have fixed my issue: <action application="answer"/> But I'm still a bit foggy on the details. Specifically I'd like to know more about "answer" and if there is anything in the log which indicates a call being picked up in early media. (it seems counter intuitive) Presumably it's not safe to always insert the above, otherwise would it not be the default behavior? -gh
If this "fixed" your issue, it sounds like some issue getting the real answer indication from the answering phone. I would enable sip trace and see if we are actually getting the 200 from the phone or not. Mike On Apr 14, 2010, at 1:40 AM, Garrison Hoffman wrote:
Adding the following seems to have fixed my issue:
<action application="answer"/>
But I'm still a bit foggy on the details. Specifically I'd like to know more about "answer" and if there is anything in the log which indicates a call being picked up in early media. (it seems counter intuitive)
Presumably it's not safe to always insert the above, otherwise would it not be the default behavior?
-gh
Michael Jerris wrote:
If this "fixed" your issue, it sounds like some issue getting the real answer indication from the answering phone. I would enable sip trace and see if we are actually getting the 200 from the phone or not.
Actually it seems worse, but I've not had any time to look carefully. This weekend I'll read through all the debugging info on the wiki, looks like SIP trace is just the beginning of available diagnostics. Thanks for the tips. -gh
participants (2)
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Garrison Hoffman
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Michael Jerris