Archives : March-2015

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Hey guys,tried to make tls work with pjsip on asterisk 13.2.0have compiled pjsip with ssl,added transport[tls]type=transport cert_file=/pbx/keys/server.crt ca_list_file=/pbx/keys/ca.key priv_key_file=/pbx/keys/server.key protocol=tls bind=192.168.1.4:5061local_net=192.168.1.0/24external_media_address=77.77.77.77external_signaling_address=77.77.77..

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Ive been having some issues with srtp. so I checked which version of libsrtp I built asterisk 11.6 against. Im on fedora 21, so libsrtp-1.4.4-13.20101004cvs.fc21.x86_64. From https://github.com/cisco/libsrtp it seems that latest release is 1.5.1, relea..

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, I have a question with account codes, all my outgoing calls are authenticated, but now the boss wants to monitor these calls with the codes.example: maria has an extension 110, but peter was in place and use the phone maria , maria then says that ..

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I am having a problem with my queue_log.When an agent transfers a call I am not getting the extension that was dialed for transfer, I am only getting the name of the macro we use: 1425307308|1425307242.33367|PedidosKosmos|Agente 102|TRANSFER|s|macro-stdexten|13|5..

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I am playing around with events in asterisk via asterisk manager – ive noticed it doesnt seem to be emitting events to my connected client. Is there something that I need to do to receive events?Also output from manager show eventsvoip*CLI> manager s..

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