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Hello list !<BR>
<BR>
SETUP :<BR>
Grandstream --sip--> Local Asterisk (NSLU) --iax--> Hosted Asterisk (VirtualDedicatedServer) --sip--> SIPprovider --> my CellPhone<BR>
<BR>
PROBLEM :<BR>
I've noticed that when I put down the horn of my Grandstream it still takes a while for my GSM/CellPhone to stop ringing.<BR>
<BR>
INFORMATION :<BR>
This is the output on the CLI of the local Asterisk-server :<BR>
<BR>
<FONT SIZE="2">[Oct 3 17:40:33] -- Executing [0473775006@intern:1] NoOp("SIP/nslu-00181d90", "Call to gsm-number") in new stack</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:33] -- Executing [0473775006@intern:2] Dial("SIP/nslu-00181d90", "IAX2/hostedasterisk/0473775006") in new stack</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:33] -- Called hostedasterisk/0473775006</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:33] -- Call accepted by XX.31.XX.XX (format alaw)</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:33] -- Format for call is alaw</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:39] -- IAX2/hostedasterisk-9480 is making progress passing it to SIP/nslu-00181d90</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:47] -- Hungup 'IAX2/hostedasterisk-9480'</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:47] == Spawn extension (intern, 0473775006, 2) exited non-zero on 'SIP/nslu-00181d90'</FONT><BR>
<BR>
This is the output on the CLI of the Asterisk-server in the datacenter :<BR>
<BR>
<FONT SIZE="2">[Oct 3 17:40:55] -- Accepting AUTHENTICATED call from XX.22.XX.XX:</FONT><BR>
<FONT SIZE="2"> > requested format = alaw,</FONT><BR>
<FONT SIZE="2"> > requested prefs = disabled,</FONT><BR>
<FONT SIZE="2"> > actual format = alaw,</FONT><BR>
<FONT SIZE="2"> > host prefs = disabled,</FONT><BR>
<FONT SIZE="2"> > priority = reqonly</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:55] -- Executing [0473775006@from-BOX-YOCAN:1] Set("IAX2/BOX-YOCAN-10022", "SIPOUT=YOCAN-3STARSNET") in new stack</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:55] -- Executing [0473775006@from-BOX-YOCAN:2] NoOp("IAX2/BOX-YOCAN-10022", "call from BOX-YOCAN") in new stack</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:55] -- Executing [0473775006@from-BOX-YOCAN:3] Dial("IAX2/BOX-YOCAN-10022", "SIP/YOCAN-3STARSNET/0473775006") in new stack</FONT><BR>
<FONT SIZE="2">[Oct 3 17:40:55] -- Called YOCAN-3STARSNET/0473775006</FONT><BR>
<FONT SIZE="2">[Oct 3 17:41:01] -- SIP/YOCAN-3STARSNET-076fa990 is making progress passing it to IAX2/BOX-YOCAN-10022</FONT><BR>
<B><FONT SIZE="2">[Oct 3 17:41:06] ERROR[1499]: rtp.c:2482 ast_rtcp_write_sr: RTCP SR transmission error to 8</FONT></B><B><FONT SIZE="2">X</FONT></B><B><FONT SIZE="2">.</FONT></B><B><FONT SIZE="2">1X</FONT></B><B><FONT SIZE="2">.</FONT></B><B><FONT SIZE="2">XX</FONT></B><B><FONT SIZE="2">.</FONT></B><B><FONT SIZE="2">XX</FONT></B><B><FONT SIZE="2">:14129, rtcp halted Operation not permitted</FONT></B><BR>
<FONT SIZE="2">[Oct 3 17:41:15] == Spawn extension (from-BOX-YOCAN, 0473775006, 3) exited non-zero on 'IAX2/BOX-YOCAN-10022'</FONT><BR>
<FONT SIZE="2">[Oct 3 17:41:15] -- Hungup 'IAX2/BOX-YOCAN-10022'</FONT><BR>
<BR>
<BR>
QUESTIONS :<BR>
<BR>
1. what is this RTCP SR transmission error ???<BR>
<BR>
2. You can notice that the local server detects the 'hungup' @ <FONT SIZE="2">17:40:47</FONT> and the Asterisk-server in the datacenter @ <FONT SIZE="2">17:41:15</FONT>. There's a lot of time between the moment that I put down the horn of the Grandstream and the moment that my CellPhone stops ringing.<BR>
Is the <U>delay</U> due to the RTCP SR transmission error ???<BR>
<BR>
<BR>
Greetingz,<BR>
Jonas.
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