–9DBBeVdKdClu1OoU3sEkJLxXEY5S86MAv Content-Type: text/plain; charset=utf-8Content-Transfer-Encoding: quoted-printable Content-Language: en-USall,on my old Asterisk 14.x box i use queue for some offices. For example, in this scenario phone 5710 is ring..
Author : Michele Pinassi
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–C0QFpPkb1GnU2Jkez22QMuv3Jxua41OsHContent-Type: text/plain; charset=utf-8Content-Transfer-Encoding: quoted-printable Content-Language: en-USall, im getting this error:[Feb 21 09:29:09] ERROR[1250]: pjproject:0 : sip_transport.c Er..
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–jVb5J9GsQalSmvgEsSWrloCFTLBemLxou Content-Type: text/plain; charset=utf-8Content-Transfer-Encoding: quoted-printableall,after upgrading from 13.7 to 14.2, asterisk cli (asterisk -r) dont show whats happens. Ive trying setting debug and verbose to ..
all,im experiencing a really frustrating issue with my Asterisk 13.7.2 with realtime configuration on MySQL and Voicemail.Heres res_config_mysql.conf:/[default]////dbhost = 192.168.1.1////dbname = asterisk////dbuser = asterisk////dbpass = [xxxxx]////dbp..
all,im writing because going crazy on this issue im unable to solve. My VoIP system is based on OpenSIPS router that forward calls to an Asterisk BOX to have IVR and Queue services.If a call was directed to a queue and operator answer, on transfer..
all,im trying to setup a function like secretary/director: when an user call director number (eg. 5000), the call were firstly diverted to secretary (5001). At this point, when secretary answer, can decide to transfer back the call to director (5000).Beca..
all,im build and using a voip pbx system using OpenSIPS as a router (i need to serve thousand of users…) and an Asterisk server as media box, for IVR, queues and so on.Ive a PATTON PSTN GW (172.20.1.4), the VoIP OpenSIPS ROUTER(172.20.1.2) andnIn queu..