3: Allocated asterisk channel SCCP/612-0000000B DEBUG: res_rtp_asterisk.c:8440 ast_rtp_prop_set: (0x2067820) RTCP setup on RTP instance DEBUG: rtp_engine.c:543 ast_rtp_instance_new: RTP instance '0x2067820' is setup and ready to go DEBUG: res_rtp_asterisk.c:3871 rtp_allocate_transport: (0x2067820) RTP allocated port 14044 DEBUG: rtp_engine.c:526 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x2067820' SCCP: (allocPBXChannel) Create New Channel with name: SCCP/612-0000000B SCCP: (pbx_channel_allocate) try to allocate Outbound channel on line: 612 SEP002D4482438A: Set the active channel 11 on device SEP002D4482438A: No active channel on device. SEP002D4482438A: Getting the active channel on device. DEBUG: manager.c:6678 process_message: Running action 'Login' ![]() ![]() Below is when I called from a Cisco 7961 non-intercom extension 612 to a Cisco 7960 intercom extension 707: PJSIP Phone -> Cisco Intercom Extension = Cisco phone picks up and auto-answersĬisco Phone -> Another Cisco Intercom Extension = Cisco phone does not pick up and the call keeps ringing.īelow is an Asterisk console debug output if this can help anyone with this problem and I am also wondering if anyone else has this problem or if it is something I may have misconfigured. PJSIP Phone -> Paging Group = All phones work and auto-answerĬisco Phone -> Paging Group = All phones work and auto-answer I have also followed the FreePBX_Intercom Paging HowTo instructions and added the intercom extensions to the Asterisk database for autoanswer macro. I have tried this between a Cisco 7961 and a 7960, as well as between two Cisco 7961 phones with the same result. PJSIP and MicroSIP), but dialing a Cisco intercom extension from another Cisco extension (regardless if it is an intercom extension or not), the destination phone will not auto-answer and pick up the call. I have paging working with both the Cisco and non-Cisco phones together, and intercom works by dialing a Cisco intercom extension from a non-Cisco extension (e.g. I too am having an issue with the paging/intercom in FreePBX and chan-sccp. But do make sure you check out those wiki pages before. Do set core set verbose 5, core set debug 1, sccp debug core, channel, indicate, pbx before paging anyone. When reporting issues with chan-sccp, then I need to see the see the asterisk cli logging to verify this and make an educated guess about what might be going down.And you also need to make sure your switches/routers are setup to support/forward multicast traffic. And yes doing an cXML push to the phone is part of the process. You can find more about that here: Multicast-Paging. SCCP phones do support unicast and multicast and they would be perfect for paging/intercom solution. I don't know what "paging pro" is, sorry.I don't know why it would be disconnecting, if that is the case after reading the wiki page and applying the changes, then go to point 3, below. To make paging working in this simple fashion you need to follow: FreePBX_Intercom-Paging-HowTo to support sccp phones. I am assuming the simply call multiple phones at ones and put them in a large one way bridge. ![]() I don't use freepbx, so i don't know what it uses to perform intercom / paging.
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