Hi, your header need to be inside dial, not before pbx_exec(chan, "Dial(PJSIP/101@trunk-test,10, b(predial-vars,s,1) )");
And context: [predial-vars] exten=> s,1,Set(PJSIP_HEADER(add,X-VSPhone-Case)=${X-VSPhone-Case})) Atenciosamente, Neimar Lima de Ávila | Desenvolvimento/Telecomunicações | Virtual Sistemas Ltda. Rua Gonçalves Dias, 142 SL 704 - Funcionários - CEP:30.140-090 - Bhte/MG Tel: (31)3245-6213 - Ramal 2016 | Cel: (31)98495-2402 [ http://www.virtualsistemas.com.br/ | www.virtualsistemas.com.br ] | [ mailto:neimar.av...@virtualsistemas.com.br | neimar.av...@virtualsistemas.com.br ] Preserve o Meio Ambiente! Pense Antes de Imprimir Os dados transmitidos nesta mensagem destinam-se exclusivamente a(s) pessoa(s) mencionada(s) e contém informações confidenciais, legalmente protegidas, para conhecimento exclusivo do(s) destinatário(s).O exame, retransmissão, divulgação, leitura, cópia ou outro uso desta correspondência, por pessoas, físicas ou jurídicas, que não o(s) destinatário(s), constituirá obtenção de dados por meio ilícito, configurando ofensa ao Art. 5°, inciso XII, da CF/88. De: "Benoit Duverger" <bduver...@atwtech.com> Para: "Asterisk Developers Mailing List" <asterisk-dev@lists.digium.com> Enviadas: Quarta-feira, 21 de outubro de 2020 14:28:08 Assunto: Re: [asterisk-dev] Add SIP Header with PJSIP in C module Thanks for your quick answer. I'm not sure to understand how Pre-Dial Handlers can help my module written in C. But if I decide to rewrite this module in asterisk language, that could help me. For the moment I hope to fix my C module. A big resume of what this part of my module do is: pbx_exec(chan, "SipAddHeader(X-MyHeader:valuetest)"); pbx_exec(chan, "Dial(SIP/101@trunk-test,10)"); That works in asterisk 1.8, 11 and probably in asterisk 16 if I use chan_sip but SipAddHeader is no longer a valid application in my asterisk because I don't load chan_sip.so, just all modules related to PJSIP. So with PJSIP, I try: pbx_builtin_setvar_helper(chan, "PJSIP_HEADER(add,X-MyHeader)", "valuetest"); pbx_exec(chan, "Dial(PJSIP/101@trunk-test,10)"); I didn't have any errors but my header is not added. Thanks Le mer. 21 oct. 2020 à 12:23, Richard Mudgett < [ mailto:rmudg...@digium.com | rmudg...@digium.com ] > a écrit : You add headers in a similar way as before. It is just a matter of adding them to the right channel. You must add them to the outgoing channel for PJSIP. This can be accomplished by using pre-dial handlers [1][2]. Richard [1] [ https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers | https://wiki.asterisk.org/wiki/display/AST/Pre-Dial+Handlers ] [2] [ https://www.asterisk.org/dialplan-handler-routines-allow-customization/ | https://www.asterisk.org/dialplan-handler-routines-allow-customization/ ] On Wed, Oct 21, 2020 at 10:49 AM Benoit Duverger < [ mailto:bduver...@atwtech.com | bduver...@atwtech.com ] > wrote: BQ_BEGIN Hello, We have a module written in C which was developed initially for asterisk 1.4, modified a few years ago to run in asterisk 1.8 then 11. This module is used to verify user's limits, route calls etc... Actually, I try to adapt it to run in asterisk 16, I moved from chan_sip to PJSIP and I don't know how can I add SIP Headers into the channel. With chan_sip we used that: sprintf( cmd, "SipAddHeader(command:%s)", command ); res = astcmd( chan, cmd ); astcmd is a custom function wrapped onto pbx_exec(). I tried to use pbx_builtin_setvar_helper(), with the function PJSIP_HEADER() but I didn't see any custom headers in SIP... and no errors, res = 0. res = pbx_builtin_setvar_helper(chan, "PJSIP_HEADER(add, X-test)", "test"); How can I use PJSIP_HEADER in a C module ?, which libraries should I need to import ? Thanks -- Benoit -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by [ http://www.api-digital.com/ | http://www.api-digital.com ] -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: [ http://lists.digium.com/mailman/listinfo/asterisk-dev | http://lists.digium.com/mailman/listinfo/asterisk-dev ] -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by [ http://www.api-digital.com/ | http://www.api-digital.com ] -- asterisk-dev mailing list To UNSUBSCRIBE or update options visit: [ http://lists.digium.com/mailman/listinfo/asterisk-dev | http://lists.digium.com/mailman/listinfo/asterisk-dev ] BQ_END -- Benoit Duverger Administrateur Systèmes et Réseaux Sysadmin T : [ tel:(514)%20985-2570 | 514- ] 985-2570 #148 [ http://www.atwtech.com/ | www.atwtech.com ] 1050 de la Montagne, Suite 400 Montréal (Québec) H3G 1Y8 Avis de confidentialité Le contenu de ce message ainsi que du ou des fichiers qui y sont joints est strictement confidentiel et destiné exclusivement à son ou sa destinataire. 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