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I have put canreinvite=no for all my internal SIP-clients in sip.conf because I want Asterisk to be in the middle of the RTP-stream so he can provide MusiconHold and so...<BR>
<BR>
Now, what the Asterisk CLI tells me when I make a call from my one internal SIP-phone to another internal SIP-phone is :<BR>
<BR>
Verbosity is at least 25<BR>
== Spawn extension (intern, 51, 1) exited non-zero on 'SIP/BT201-088f93e0'<BR>
-- Executing [52@intern:1] Dial("SIP/GXP1200-088f93e0", "SIP/BT201|30") in new stack<BR>
-- Called BT201<BR>
-- SIP/BT201-088faa00 is ringing<BR>
-- SIP/BT201-088faa00 answered SIP/GXP1200-088f93e0<BR>
<B> -- Packet2Packet bridging SIP/GXP1200-088f93e0 and SIP/BT201-088faa00</B><BR>
== Spawn extension (intern, 52, 1) exited non-zero on 'SIP/GXP1200-088f93e0'<BR>
<BR>
Why is there this native bridging ? Does this mean that Asterisk is no longer in the middle of it ?<BR>
<BR>
Also : there is no audio at all ! Just when I put down the phone there's the DTMF-signal that the line is cancelled...<BR>
<BR>
Everything worked well before I edited musiconhold.conf and features.conf (to create a park extension).<BR>
<BR>
<BR>
My sip.conf :<BR>
<BR>
<I><FONT SIZE="2">[root@asterisk asterisk]# cat sip.conf</FONT></I><BR>
<I><FONT SIZE="2">[general]</FONT></I><BR>
<I><FONT SIZE="2">context=default</FONT></I><BR>
<I><FONT SIZE="2">port=5060</FONT></I><BR>
<I><FONT SIZE="2">bindaddr=192.168.4.248</FONT></I><BR>
<I><FONT SIZE="2">srvlookup=yes</FONT></I><BR>
<I><FONT SIZE="2">disallow=all</FONT></I><BR>
<I><FONT SIZE="2">allow=alaw</FONT></I><BR>
<I><FONT SIZE="2">allow=gsm</FONT></I><BR>
<I><FONT SIZE="2">allow=ulaw</FONT></I><BR>
<I><FONT SIZE="2">language=be</FONT></I><BR>
<BR>
<I><FONT SIZE="2">[BT201]</FONT></I><BR>
<I><FONT SIZE="2">type=friend</FONT></I><BR>
<I><FONT SIZE="2">context=intern</FONT></I><BR>
<I><FONT SIZE="2">host=dynamic</FONT></I><BR>
<I><FONT SIZE="2">username=BT201</FONT></I><BR>
<I><FONT SIZE="2">secret=testpaswoord</FONT></I><BR>
<I><FONT SIZE="2">canreinvite=no</FONT></I><BR>
<I><FONT SIZE="2">callerid=Jonas Kellens <52></FONT></I><BR>
<I><FONT SIZE="2">qualify=yes</FONT></I><BR>
<BR>
<I><FONT SIZE="2">[GXP1200]</FONT></I><BR>
<I><FONT SIZE="2">type=friend</FONT></I><BR>
<I><FONT SIZE="2">context=intern</FONT></I><BR>
<I><FONT SIZE="2">host=dynamic</FONT></I><BR>
<I><FONT SIZE="2">username=GXP1200</FONT></I><BR>
<I><FONT SIZE="2">secret=testpaswoord</FONT></I><BR>
<I><FONT SIZE="2">canreinvite=no</FONT></I><BR>
<I><FONT SIZE="2">callerid=callerid? <51></FONT></I><BR>
<I><FONT SIZE="2">qualify=yes</FONT></I><BR>
<BR>
<BR>
<I><FONT SIZE="2">[GXP2020]</FONT></I><BR>
<I><FONT SIZE="2">type=friend</FONT></I><BR>
<I><FONT SIZE="2">context=intern</FONT></I><BR>
<I><FONT SIZE="2">host=dynamic</FONT></I><BR>
<I><FONT SIZE="2">username=GXP2020</FONT></I><BR>
<I><FONT SIZE="2">secret=testpaswoord</FONT></I><BR>
<I><FONT SIZE="2">canreinvite=no</FONT></I><BR>
<I><FONT SIZE="2">callerid=Kristof Teirlinck <50></FONT></I><BR>
<I><FONT SIZE="2">qualify=yes</FONT></I><BR>
<BR>
Musiconhold.conf :<BR>
<BR>
<I><FONT SIZE="2">[default]</FONT></I><BR>
<I><FONT SIZE="2">mode=files</FONT></I><BR>
<I><FONT SIZE="2">directory=/var/lib/asterisk/moh</FONT></I><BR>
<I><FONT SIZE="2">random=yes</FONT></I><BR>
<BR>
Features.conf :<BR>
<BR>
; Sample Call Features (parking, transfer, etc) configuration<BR>
;<BR>
<BR>
<I><FONT SIZE="2">[general]</FONT></I><BR>
<I><FONT SIZE="2">parkext => 90 What extension to dial to park</FONT></I><BR>
<I><FONT SIZE="2">parkpos => 91-95 What extensions to park calls on. These needs to be</FONT></I><BR>
<I><FONT SIZE="2"> numeric, as Asterisk starts from the start position</FONT></I><BR>
<I><FONT SIZE="2"> and increments with one for the next parked call.</FONT></I><BR>
<I><FONT SIZE="2">context => parkedcalls Which context parked calls are in</FONT></I><BR>
<I><FONT SIZE="2">parkingtime => 60 Number of seconds a call can be parked for </FONT></I><BR>
<I><FONT SIZE="2"> (default is 45 seconds)</FONT></I><BR>
<I><FONT SIZE="2">;courtesytone = beep Sound file to play to the parked caller </FONT></I><BR>
<I><FONT SIZE="2"> when someone dials a parked call</FONT></I><BR>
<I><FONT SIZE="2"> or the Touch Monitor is activated/deactivated.</FONT></I><BR>
<I><FONT SIZE="2">;parkedplay = caller Who to play the courtesy tone to when picking up a parked call</FONT></I><BR>
<I><FONT SIZE="2"> one of: parked, caller, both (default is caller)</FONT></I><BR>
<I><FONT SIZE="2">;parkedcalltransfers = caller ; Enables or disables DTMF based transfers when picking up a parked call.</FONT></I><BR>
<I><FONT SIZE="2"> ; one of: callee, caller, both, no (default is both)</FONT></I><BR>
<I><FONT SIZE="2">;parkedcallreparking = caller ; Enables or disables DTMF based one-touch parking when picking up a parked call.</FONT></I><BR>
<I><FONT SIZE="2"> ; one of: callee, caller, both, no (default is no)</FONT></I><BR>
<I><FONT SIZE="2">;parkedcallhangup = caller ; Enables or disables DTMF based hangups when picking up a parked call.</FONT></I><BR>
<I><FONT SIZE="2"> ; one of: callee, caller, both, no (default is no)</FONT></I><BR>
<I><FONT SIZE="2">;parkedcallrecording = caller ; Enables or disables DTMF based one-touch recording when picking up a parked call.</FONT></I><BR>
<I><FONT SIZE="2"> ; one of: callee, caller, both, no (default is no)</FONT></I><BR>
<I><FONT SIZE="2">;adsipark = yes if you want ADSI parking announcements</FONT></I><BR>
<I><FONT SIZE="2">;findslot => next Continue to the 'next' free parking space. </FONT></I><BR>
<I><FONT SIZE="2"> Defaults to 'first' available</FONT></I><BR>
<I><FONT SIZE="2">parkedmusicclass=default This is the MOH class to use for the parked channel</FONT></I><BR>
<I><FONT SIZE="2"> as long as the class is not set on the channel directly</FONT></I><BR>
<I><FONT SIZE="2"> using Set(CHANNEL(musicclass)=whatever) in the dialplan</FONT></I><BR>
<BR>
<I><FONT SIZE="2">;transferdigittimeout => 3 Number of seconds to wait between digits when transferring a call</FONT></I><BR>
<I><FONT SIZE="2"> (default is 3 seconds)</FONT></I><BR>
<I><FONT SIZE="2">;xfersound = beep to indicate an attended transfer is complete</FONT></I><BR>
<I><FONT SIZE="2">;xferfailsound = beeperr to indicate a failed transfer</FONT></I><BR>
<I><FONT SIZE="2">pickupexten = *8 Configure the pickup extension. (default is *8)</FONT></I><BR>
<I><FONT SIZE="2">;featuredigittimeout = 1000 ; Max time (ms) between digits for </FONT></I><BR>
<I><FONT SIZE="2"> ; feature activation (default is 1000 ms)</FONT></I><BR>
<I><FONT SIZE="2">;atxfernoanswertimeout = 15 ; Timeout for answer on attended transfer default is 15 seconds.</FONT></I><BR>
<BR>
<BR>
Do you need extra info ??<BR>
What setting can I have set in musiconhold.conf or features.conf to affect the audiostream between my clients ???<BR>
<BR>
Before I could call all my clients, I had musiconhold when putting 'on hold' and I was just figuring out how parked calls worked...<BR>
<BR>
Thanks for the help !<BR>
<BR>
Jonas Kellens.
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