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<p class=MsoNormal><span class=style131><font size=2 color="#003366"
face="Times New Roman"><span style='font-size:10.0pt;color:#003366'>Anyone have
any idea on how to force marker bits on in RTP ?<o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=style131><font size=2 color="#003366"
face="Times New Roman"><span style='font-size:10.0pt;color:#003366'><o:p> </o:p></span></font></span></p>
<div class=MsoNormal align=center style='text-align:center'><font size=3
face="Times New Roman"><span style='font-size:12.0pt'>
<hr size=2 width="100%" align=center tabindex=-1>
</span></font></div>
<p class=MsoNormal><b><font size=2 face=Tahoma><span style='font-size:10.0pt;
font-family:Tahoma;font-weight:bold'>From:</span></font></b><font size=2
face=Tahoma><span style='font-size:10.0pt;font-family:Tahoma'>
asterisk-users-bounces@lists.digium.com
[mailto:asterisk-users-bounces@lists.digium.com] <b><span style='font-weight:
bold'>On Behalf Of </span></b><st1:PersonName w:st="on">Adrian Marsh</st1:PersonName><br>
<b><span style='font-weight:bold'>Sent:</span></b> 10 June 2009 14:50<br>
<b><span style='font-weight:bold'>To:</span></b> <st1:PersonName w:st="on">Asterisk
Users Mailing List - Non-Commercial Discussion</st1:PersonName><br>
<b><span style='font-weight:bold'>Subject:</span></b> [asterisk-users]
Resetting Marker Bits</span></font><o:p></o:p></p>
</div>
<p class=MsoNormal><font size=3 face="Times New Roman"><span style='font-size:
12.0pt'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>(resend as apparently I was blocked)<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=2 face=Arial><span lang=EN-GB style='font-size:
10.0pt;font-family:Arial'>Hi</span></font><span class=style131><font
color="#003366"><span lang=EN-GB style='color:#003366'> All,</span><o:p></o:p></font></span></p>
<p class=MsoNormal><span class=style131><font size=3 color="#003366"
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=style131><font size=3 color="#003366"
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;color:#003366'>I’m
looking for how to enable SIP Markers, or specifically, how to have the TIME
reset when a call route changes.<o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=style131><font size=3 color="#003366"
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=style131><font size=3 color="#003366"
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;color:#003366'>I’m
debugging an issue, where a sip client we have switching to one-way-audio, when
an asterisk server fruther down the call path dials out to the PSTN. Scenario
is:<o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=style131><font size=3 color="#003366"
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal><span class=style131><font size=3 color="#003366"
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;color:#003366'>SIP
Client -> A*k1 -> A*k2 -> PSTN
Provider/Gradwell -> O2 -> <st1:place w:st="on"><st1:City
w:st="on">Mobile</st1:City></st1:place><o:p></o:p></span></font></span></p>
<p class=MsoNormal><span class=style131><font size=3 color="#003366"
face="Times New Roman"><span lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></span></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><font
size=3 face="Times New Roman"><span style='font-size:12.0pt'><span
style='mso-list:Ignore'>-<font size=1 face="Times New Roman"><span
style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><span class=style131><font
color="#003366"><span lang=EN-GB style='color:#003366'>the SIP client dials on
O2 mobile, call goes out</span></font></span><font color="#003366"><span
lang=EN-GB style='color:#003366'> to A*1.</span></font><o:p></o:p></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><font
size=3 color="#003366" face="Times New Roman"><span lang=EN-GB
style='font-size:12.0pt;color:#003366'><span style='mso-list:Ignore'>-<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font color="#003366"><span
lang=EN-GB style='color:#003366'>A*1 Dials out to A*k2 as A*k2 is the gateway
to PSTN providers and normal office phones.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><font
size=3 color="#003366" face="Times New Roman"><span lang=EN-GB
style='font-size:12.0pt;color:#003366'><span style='mso-list:Ignore'>-<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font color="#003366"><span
lang=EN-GB style='color:#003366'>A*k2 dials some local Cisco phones, then on no
answer plays an audio file, so call is ANSWERED.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><font
size=3 color="#003366" face="Times New Roman"><span lang=EN-GB
style='font-size:12.0pt;color:#003366'><span style='mso-list:Ignore'>-<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font color="#003366"><span
lang=EN-GB style='color:#003366'>A*k2 then Dials out to gradwell, to a mobile
phone number.<o:p></o:p></span></font></p>
<p class=MsoNormal style='margin-left:36.0pt;text-indent:-18.0pt;mso-list:l0 level1 lfo2'><![if !supportLists]><font
size=3 color="#003366" face="Times New Roman"><span lang=EN-GB
style='font-size:12.0pt;color:#003366'><span style='mso-list:Ignore'>-<font
size=1 face="Times New Roman"><span style='font:7.0pt "Times New Roman"'>
</span></font></span></span></font><![endif]><font color="#003366"><span
lang=EN-GB style='color:#003366'>Gradwell takes the call, routes it via PSTN.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'>My problem, is that at the
point where the O2 mobile accepts the call, I get one-way audio. (SIP Client
outbound, nothing inbound).<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'>Tracing the RTP stream all
the way back, I can see that audio makes it all the way to the SIP Client.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'>However, we notice that
at the point where the O2 mobile answers, the TIME= value of the packet jumps
significantly, say from 119248 to <st2:dial w:st="on">1518324408</st2:dial>.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'>Talking to the sip client
developer, they say that I need to enable SIP Markers on the server (I guess
A*k2), so that if the stream source changes then the timers are reset.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'>Does this sound right, and if
so, how do I do that ?<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'>I am running an older load on
A*<st1:place w:st="on">K2</st1:place> of 1.4.18, and 1.4.15svn (privately
compiled to add an extra codec) on A*k1. I can look into upgrading
these, but the developer thinks it’s just a missing config on Asterisk.<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'>Thanks,<o:p></o:p></span></font></p>
<p class=MsoNormal><font size=3 color="#003366" face="Times New Roman"><span
lang=EN-GB style='font-size:12.0pt;color:#003366'><o:p> </o:p></span></font></p>
<p class=MsoNormal><st1:place w:st="on"><st1:City w:st="on"><font size=3
color="#003366" face="Times New Roman"><span lang=EN-GB style='font-size:
12.0pt;color:#003366'>Adrian</span></font></st1:City></st1:place><font
color="#003366"><span lang=EN-GB style='color:#003366'><o:p></o:p></span></font></p>
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