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List:       vocal
Subject:    [Vocal] Call from ATA-186 failed
From:       hspark () etri ! re ! kr
Date:       2003-08-25 23:46:56
Message-ID: B4B7A29DE8BBD740BF47F84B8D516FCB81D425 () cms3 ! cms ! etri ! re ! kr
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Hello, 
I am using an ATA-186 with firmware v2.15 for SIP... 
Analog Phone --- ATA186 --- SIP Proxy1 --- SIP Proxy2 --- SIP UA 
(Analog Phone - sip:1001@home1.net 
 SIP Proxy1    - sip:home1.net 
 SIP Proxy2    - sip:home2.net 
 SIP UA         - sip:2001@home2.net) 
In above configuration, 
The call from SIP UA to Analog phone has succeeded... 
But the call from Analog phone to SIP UA has failed... 
When I try to make a call by dialing "2001", 
ATA-186 sets Request-URI "sip:2001@home1.net" in INVITE message... 
Then SIP Proxy 1 responds with error response 
because "2001" is not a "home1.net" user... 
In Administrator¡¯s Guide Appendix D ¡°Cisco ATA-to-Cisco ATA-Basic SIP to
SIP Call Without Authentication¡±, 
SIP Proxy 1 changes Request-URI (sip:2001@home1.net) into
(sip:2001@home2.net)... 
But it is inconsistent with SIP standard... 
If the call succeeds with conformance to SIP standard, 
ATA-186 should set Request-URI "sip:2001@home2.net" in INVITE message... 
Is there any solution to set Request-URI with other than user@proxy's host
in ATA-186 ? 
In Administrator¡¯s Guide Chapter4 ¡°Placing an IP Call¡±, 
ATA-186 supports IP call by IP dialing <phone number>**<ipaddress># ... 
But when it is used with SIP Proxy, IP dialing is not supported... 
I think URL dialing is the best solution for inter-domain calls in ATA-
186... 
But there is a line states "URL dialing is not Supported" ... 
Any comments appreciated... 
Thanks in advance... 
Brad Park 
 

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<p><font size=2 face=±¼¸²><span lang=EN-US \
style='font-size:10.0pt'>Hello,</span></font><span lang=EN-US> <o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>I am using an
ATA-186 with firmware v2.15 for SIP...</span></font><span lang=EN-US> \
<o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>Analog Phone
--- ATA186 --- SIP Proxy1 --- SIP Proxy2 --- SIP <span class=GramE>UA<font
size=3><span style='font-size:12.0pt'> <br>
</span></font>(</span>Analog Phone&nbsp;- sip:1001@home1.net</span></font><span
lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>&nbsp;SIP
Proxy1&nbsp;&nbsp; <span style='mso-spacerun:yes'>&nbsp;</span>- \
sip:home1.net</span></font><span lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>&nbsp;SIP
Proxy2&nbsp;&nbsp; <span style='mso-spacerun:yes'>&nbsp;</span>- \
sip:home2.net</span></font><span lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>&nbsp;SIP
UA&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <span
style='mso-spacerun:yes'>&nbsp;</span>- sip:2001@home2.net) </span></font><span
lang=EN-US><o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>In above
configuration<span class=GramE>,<font size=3><span style='font-size:12.0pt'> <br>
</span></font>The</span> call from SIP UA to Analog phone has \
succeeded...</span></font><span lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>But the call from
Analog phone to SIP UA has failed...</span></font><span lang=EN-US> \
<o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>When I try to
make a call by dialing &quot;2001&quot;<span class=GramE>, <font size=3><span
style='font-size:12.0pt'><br>
</span></font>ATA</span>-186 sets Request-URI &quot;sip:2001@home1.net&quot; in
INVITE message...</span></font><span lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>Then SIP Proxy 1
responds with error <span class=GramE>response <font size=3><span
style='font-size:12.0pt'><br>
</span></font>because</span> &quot;2001&quot; is not a &quot;home1.net&quot;
user...</span></font><span lang=EN-US> <o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>In
Administrator¡¯s Guide Appendix D ¡°Cisco ATA-to-Cisco ATA-Basic SIP to SIP Call
Without Authentication¡±<span class=GramE>, <font size=3><span style='font-size:
12.0pt'><br>
</span></font>SIP</span> Proxy 1 changes Request-URI (sip:2001@home1.net) into
(sip:2001@home2.net)...</span></font><span lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>But it is
inconsistent with SIP standard...</span></font><span lang=EN-US> \
<o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>If the call
succeeds with conformance to SIP standard<span class=GramE>, <font size=3><span
style='font-size:12.0pt'><br>
</span></font>ATA</span>-186 should set Request-URI
&quot;sip:2001@home2.net&quot; in INVITE message...</span></font><span
lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>Is there any
solution to set Request-URI with other than <span class=SpellE>user@proxy's</span>
host in ATA-<span class=GramE>186 ?</span></span></font><span lang=EN-US> \
<o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>In
Administrator¡¯s Guide Chapter4 ¡°Placing an IP Call¡±<span class=GramE>,<font
size=3><span style='font-size:12.0pt'> <br>
</span></font>ATA</span>-186 supports IP call by IP dialing &lt;phone
number&gt;**&lt;<span class=SpellE>ipaddress</span>&gt;# ...</span></font><span
lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>But when it is
used with SIP Proxy, IP dialing is not supported...</span></font><span
lang=EN-US> <o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>I think URL
dialing is the best solution for inter-domain calls in ATA-186... </span></font><span
lang=EN-US><br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>But there is a
line states &quot;URL dialing is not Supported<span class=GramE>&quot; \
...</span></span></font><span lang=EN-US> <o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>Any comments
appreciated...</span></font><span lang=EN-US> <br>
</span><font size=2><span lang=EN-US style='font-size:10.0pt'>Thanks in
advance...</span></font><span lang=EN-US> <o:p></o:p></span></p>

<p><font size=2 face=±¼¸²><span lang=EN-US style='font-size:10.0pt'>Brad \
Park</span></font><span lang=EN-US> <o:p></o:p></span></p>

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mso-bidi-font-family:Arial'><o:p>&nbsp;</o:p></span></font></p>

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