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
Re: [asterisk-users] can PJSIP_MEDIA_OFFER work like SIP_CODEC?
Am 27.09.2014 17:28, schrieb d tbsky: can someone give an example for the function? thanks for the help. Not a programmer here, just grep -r'ed through the code, but maybe try one of these: G711A G711_ALAW -- _ -- 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?
2014-09-28 14:01 GMT+08:00 Markus unive...@truemetal.org: Am 27.09.2014 17:28, schrieb d tbsky: can someone give an example for the function? thanks for the help. Not a programmer here, just grep -r'ed through the code, but maybe try one of these: G711A G711_ALAW thanks a lot for help!! I tried both but none works. maybe this function can not work like the old channel variable SIP_CODEC, which can change inbound call codec. but I do notice something different between chan_sip and chan_pjsip. I use zoiper softphone for testing: when I dialout sip trunk with chan_sip, the remote peer rings, and zoiper now shows what codec to use. if I use SIP_CODEC before dial to change the codec, zoiper will use the new CODEC, but asterisk internal won't change and still transcoding in the middle.(at least core show channel sip/x told me transcoding) when I dialout sip trunk with chan_pjsip, the remote peer rings, but zoiper didn't show what codec to use. only after the callee answer the phone, zoiper shows what codec to use. so it seems chan_pjsip have better chance to do the right thing without transcoding. it's sad that chan_pjsip won't select best codec match two peers automatically without transcoding. but I hope it at least can provide a magic function or channel variable like SIP_CODEC/SIP_CODEC_INBOUND to make correct codec selection. 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
