Re: [SR-Users] Issue with Asterisk interconnection for VoiceMail

Hello Daniel, I tried and it works! Thank you. So, as suggested, I write $du=$null before call RELAY route. Regards, Igor. -----Message d'origine----- De : Igor Potjevlesch [mailto:igor.potjevlesch@gmail.com] Envoyé : lundi 18 mai 2015 16:42 À : 'Kamailio (SER) - Users Mailing List' Objet : RE: [SR-Users] Issue with Asterisk interconnection for VoiceMail Hello Daniel, I can try this. But there are cases where lookup is called and the redirection to VoiceMail is working fine. Could it be an issue with a missing "append_branch()" instruction? Regards, Igor. -----Message d'origine----- De : sr-users [mailto:sr-users-bounces@lists.sip-router.org] De la part de Daniel Tryba Envoyé : vendredi 15 mai 2015 10:58 À : sr-users(a)lists.sip-router.org Objet : Re: [SR-Users] Issue with Asterisk interconnection for VoiceMail On Friday 15 May 2015 10:30:45 Igor Potjevlesch wrote: Unset $du ($du=$null) when routing to voicemail. It is set after a lookup(): http://www.kamailio.org/wiki/cookbooks/4.1.x/pseudovariables#du_-_destinatio n_uri _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users