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<p>Yes of course, you need to remove sdp as well while changing reply from 183 to 180,</p>
<p><a href="http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#id292832">http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#id292832</a></p>
<p>Thank you.</p>
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<p>On 2013-09-26 18:21, Jeff Pyle wrote:</p>
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<div dir="ltr">Muhammad,
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<div>That makes sense. I think in my case I would have to strip the SDP as well? Any thoughts on the media sent from the b-leg back to the a-leg when it's not being expected (because there is no SDP)?</div>
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<div>- Jeff</div>
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<div class="gmail_quote">On Tue, Sep 24, 2013 at 11:03 PM, Muhammad Shahzad Shafi <span><<a href="mailto:shahzad@voip-demos.com">shahzad@voip-demos.com</a>></span> wrote:<br />
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<p>Well, you have to sacrifice 183 Early Media, since converting 183 to 180 is far more easy and convenient then converting 180 to 183 (since then you have to involve a media server, which is not going to be so easy).</p>
<p>Therefore, my advice would be to change all 183 from that carrier to 180 response. You can use change_reply_status method,</p>
<p><a href="http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status">http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status</a></p>
<p>Thank you.</p>
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<p>On 2013-09-25 03:05, Jeff Pyle wrote:</p>
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<div dir="ltr">No takers? :)
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<div>I wonder if it's possible to script this in a B2BUA scenario? I'm not sure how one would do detection of 180 without SDP versus 180/183 with SDP in B2B-land. Or, what to do from there once it knew.</div>
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<div>- Jeff</div>
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<div class="gmail_quote">On Mon, Sep 23, 2013 at 10:43 AM, Jeff Pyle <span><<a href="mailto:jpyle@fidelityvoice.com">jpyle@fidelityvoice.com</a>></span> wrote:<br />
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<div dir="ltr">Hi Laszlo,
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<div>Unfortunately the effect for the caller would be the same - ringback would stop.</div>
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<div>Here's the whole flow. My terminating gateway is SIP to ISDN PRI. Call terminates through the gateway to a particular mobile switching office. I receive an ISDN PROGRESS message with inband audio. This translates to the 183 with SDP. Then I receive an ALERTING message with no inband audio. This translates to the 180. When the MSO sends the ALERTING, it has stopped sending the inband audio from the previous PROGRESS message. </div>
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<div>I'm thinking I need to do something else in the onreply_route to connect to the media server for a new 183. Since I've executed t_relay to route the INVITE to the gateway, it seems my options are limited.</div>
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<div>- Jeff</div>
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<div>Jeff Pyle <<a href="mailto:jpyle@fidelityvoice.com">jpyle@fidelityvoice.com</a>><br /> Director, Voice Engineering<br /> Fidelity Voice and Data<br /> 216-245-4106<br /><a href="http://www.fidelityvoice.com">www.fidelityvoice.com</a><br /><br /></div>
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<div class="gmail_quote">On Mon, Sep 23, 2013 at 8:57 AM, Laszlo <span><<a href="mailto:laszlo@voipfreak.net">laszlo@voipfreak.net</a>></span> wrote:<br />
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<div>What if you simply drop the 180 in the onreply_route?<br /><br /></div>
-Laszlo</div>
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<div class="gmail_quote">2013/9/23 Jeff Pyle <span><<a href="mailto:jpyle@fidelityvoice.com">jpyle@fidelityvoice.com</a>></span><br />
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<div dir="ltr">Hello,
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<div>I have one particular PSTN call flow that causes a 183 with SDP, then a 180 without SDP prior to 200 OK. Some of my customer endpoints don't handle the 180 properly after a 183 and they cease to hear ringback.</div>
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<div>I'm thinking through how intercept the 180 and convert it to a 183 with SDP. I have a media server available to generate the 183 and the media. I'm struggling with how to relay the INVITE to the media server when the 180 arrives in the middle of the call setup.</div>
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<div>Any recommendations are appreciated.</div>
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<div>Regards,</div>
<div>Jeff</div>
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Kind regards,</div>
Laszlo Bekesi</div>
<a href="http://voipfreak.net">http://voipfreak.net</a></div>
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<pre>--
Mit freundlichen Grüßen
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: <a href="mailto:shari_786pk@hotmail.com">shari_786pk@hotmail.com</a>
Email: <a href="mailto:shaheryarkh@googlemail.com">shaheryarkh@googlemail.com</a></pre>
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<pre>--
Mit freundlichen Grüßen
Muhammad Shahzad
-----------------------------------
CISCO Rich Media Communication Specialist (CRMCS)
CISCO Certified Network Associate (CCNA)
Cell: +49 176 99 83 10 85
MSN: shari_786pk@hotmail.com
Email: shaheryarkh@googlemail.com</pre>
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