My pjsip.conf is the auto_generated file from freepbx and it should not be modified. I really cannot find where to set the messge_context in freepbx UI at all. could you please show me where?
On Tue, Sep 22, 2015 at 10:22 PM, Thyda ENG <[email protected]> wrote: > how if I use the auto generate once from freepbx ? > > On Tue, Sep 22, 2015 at 10:12 PM, Ishfaq Malik <[email protected]> wrote: > >> >> >> On 22 September 2015 at 16:04, Thyda ENG <[email protected]> wrote: >> >>> I have many endpoints and each endpoint has some parameter in common so >>> i wonder is there any way to config one for all endpoints? Like in my >>> example I have two endpoints and I repeat the same thing, >>> >>> [100] >>> >>> type=endpoint >>> >>> aors=100 >>> >>> auth=100-auth >>> >>> allow=ulaw,alaw,gsm,g726 >>> >>> context=from-internal >>> >>> callerid=device <100> >>> >>> dtmf_mode=rfc4733 >>> >>> use_avpf=no >>> >>> ice_support=no >>> >>> media_use_received_transport=no >>> >>> trust_id_inbound=yes >>> >>> send_pai=yes >>> >>> rtp_symmetric=yes >>> >>> rewrite_contact=yes >>> >>> message_context=astsms >>> >>> >>> [200] >>> >>> type=endpoint >>> >>> aors=200 >>> >>> auth=200-auth >>> >>> allow=ulaw,alaw,gsm,g726 >>> >>> context=from-internal >>> >>> callerid=device <200> >>> >>> dtmf_mode=rfc4733 >>> >>> use_avpf=no >>> >>> ice_support=no >>> >>> media_use_received_transport=no >>> >>> trust_id_inbound=yes >>> >>> send_pai=yes >>> >>> rtp_symmetric=yes >>> >>> rewrite_contact=yes >>> >>> message_context=astsms >>> >>> >>> how could I avoid duplicate thing like this ? >>> >>> -- >>> >>> >> From my brief look at pjsip.conf it uses the same template concept as the >> sip.conf. >> >> Here's the relevant instructions from the sip.conf in asteris13 >> >> ; >> ; Because you might have a large number of similar sections, it is >> generally >> ; convenient to use templates for the common parameters, and add them >> ; the the various sections. Examples are below, and we can even leave >> ; the templates uncommented as they will not harm: >> >> [basic-options](!) ; a template >> dtmfmode=rfc2833 >> context=from-office >> type=friend >> >> [natted-phone](!,basic-options) ; another template inheriting >> basic-options >> directmedia=no >> host=dynamic >> >> [public-phone](!,basic-options) ; another template inheriting >> basic-options >> directmedia=yes >> >> [my-codecs](!) ; a template for my preferred codecs >> disallow=all >> allow=ilbc >> allow=g729 >> allow=gsm >> allow=g723 >> allow=ulaw >> ; Or, more simply: >> ;allow=!all,ilbc,g729,gsm,g723,ulaw >> >> [ulaw-phone](!) ; and another one for ulaw-only >> disallow=all >> allow=ulaw >> ; Again, more simply: >> ;allow=!all,ulaw >> >> ; and finally instantiate a few phones >> ; >> ; [2133](natted-phone,my-codecs) >> ; secret = peekaboo >> ; [2134](natted-phone,ulaw-phone) >> ; secret = not_very_secret >> ; [2136](public-phone,ulaw-phone) >> ; secret = not_very_secret_either >> ; ... >> ; >> >> Regards >> >> Ish >> -- >> >> Ishfaq Malik >> Department: VOIP Support >> Company: Packnet Limited >> t: +44 (0)161 660 2350 >> f: +44 (0)161 660 9825 >> e: [email protected] >> w: http://www.pack-net.co.uk >> >> Registered Address: PACKNET LIMITED, Duplex 2, Ducie House >> 37 Ducie Street >> Manchester, M1 2JW >> COMPANY REG NO. 04920552 >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > >
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