Muhammad, 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)?
- Jeff On Tue, Sep 24, 2013 at 11:03 PM, Muhammad Shahzad Shafi < [email protected]> wrote: > ** > > 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). > > Therefore, my advice would be to change all 183 from that carrier to 180 > response. You can use change_reply_status method, > > > http://www.opensips.org/html/docs/modules/1.9.x/sipmsgops.html#change_reply_status > > Thank you. > > > > On 2013-09-25 03:05, Jeff Pyle wrote: > > No takers? :) > > 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. > > > - Jeff > > > > On Mon, Sep 23, 2013 at 10:43 AM, Jeff Pyle <[email protected]>wrote: > >> Hi Laszlo, >> >> Unfortunately the effect for the caller would be the same - ringback >> would stop. >> >> 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. >> >> 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. >> >> >> - Jeff >> >> >> >> -- >> Jeff Pyle <[email protected]> >> Director, Voice Engineering >> Fidelity Voice and Data >> 216-245-4106 >> www.fidelityvoice.com >> >> >> >> On Mon, Sep 23, 2013 at 8:57 AM, Laszlo <[email protected]> wrote: >> >>> What if you simply drop the 180 in the onreply_route? >>> >>> -Laszlo >>> >>> >>> 2013/9/23 Jeff Pyle <[email protected]> >>> >>>> Hello, >>>> >>>> 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. >>>> >>>> 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. >>>> >>>> Any recommendations are appreciated. >>>> >>>> >>>> >>>> Regards, >>>> Jeff >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> [email protected] >>>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>>> >>>> >>> >>> >>> -- >>> >>> -- >>> Kind regards, >>> Laszlo Bekesi >>> http://voipfreak.net >>> >>> _______________________________________________ >>> Users mailing list >>> [email protected] >>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users >>> >>> -- > Mit freundlichen Grüßen > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +49 176 99 83 10 85 > MSN: [email protected] > Email: [email protected] > > > _______________________________________________ > Users mailing list > [email protected] > http://lists.opensips.org/cgi-bin/mailman/listinfo/users > >
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