Re: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
On Sat, Sep 27, 2014 at 10:28 AM, d tbsky tbs...@gmail.com wrote: hi: when using chan_sip, I can use set SIP_CODEC in dialplan to change the codec of endpoint. this method didn't work with pjsip in asterisk 12/13. I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems can set codec, but the document didn't offer any example. i try to use something like PJSIP_MEDIA_OFFER(alaw) but didn't work. can someone give an example for the function? thanks for the help. The function should work on whatever channel it was set on. If you are going to use it on an outbound channel, then you should use a pre-dial handler to apply it to that channel. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com http://asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
2014-09-30 23:52 GMT+08:00 Matthew Jordan mjor...@digium.com: On Sat, Sep 27, 2014 at 10:28 AM, d tbsky tbs...@gmail.com wrote: I found asterisk 12/13 has a new function PJSIP_MEDIA_OFFER. according to the description, it seems can set codec, but the document didn't offer any example. i try to use something like PJSIP_MEDIA_OFFER(alaw) but didn't work. can someone give an example for the function? thanks for the help. The function should work on whatever channel it was set on. If you are going to use it on an outbound channel, then you should use a pre-dial handler to apply it to that channel. it sounds good. could you give out an one line dialplan example so I can try to use it? and the real thing I want to change is the inbound codec, can it work like the chan_sip channel variable SIP_CODEC_INBOUND? thanks a lot for your help!! Regards, tbskyd -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
