0

I know that Asterisk works like that. 1. When sip has incoming call, then it can not do outgoing call. 2. When the during conversation sip do call outgoing the current call going to the hold.

But in the my log i see some incorrect cases. How it can be? Sip 185 has incoming call on 16:06:12, but it can be do call outgoing on 16:06:13

[2019-09-20 16:06:12] VERBOSE[10838][C-00000bb3] netsock2.c: Using SIP RTP CoS mark 5
[2019-09-20 16:06:12] VERBOSE[10838][C-00000bb3] app_queue.c: Called SIP/185
[2019-09-20 16:06:12] VERBOSE[10838][C-00000bb3] app_queue.c: SIP/185-0000146d is ringing
[2019-09-20 16:06:17] VERBOSE[10838][C-00000bb3] app_queue.c: SIP/185-0000146d answered SIP/195-0000146a
[2019-09-20 16:06:17] VERBOSE[10838][C-00000bb3] res_musiconhold.c: Stopped music on hold on SIP/195-0000146a
[2019-09-20 16:06:17] VERBOSE[10916][C-00000bb3] bridge_channel.c: Channel SIP/185-0000146d joined 'simple_bridge' basic-bridge <53a37729-4e10-4abf-984e-64fea2f74694>
[2019-09-20 16:06:17] VERBOSE[10838][C-00000bb3] bridge_channel.c: Channel SIP/195-0000146a joined 'simple_bridge' basic-bridge <53a37729-4e10-4abf-984e-64fea2f74694>

[2019-09-20 16:06:13] VERBOSE[10888][C-00000bb5] pbx.c: Executing [900556611015@phones:9] Dial("SIP/185-0000146e", "SIP/195/00556611015") in new stack
[2019-09-20 16:06:13] VERBOSE[10888][C-00000bb5] netsock2.c: Using SIP RTP CoS mark 5
[2019-09-20 16:06:13] VERBOSE[10888][C-00000bb5] app_dial.c: Called SIP/195/00556611015
[2019-09-20 16:06:13] VERBOSE[10888][C-00000bb5] app_dial.c: SIP/195-0000146f is ringing
[2019-09-20 16:06:16] VERBOSE[10888][C-00000bb5] app_dial.c: SIP/195-0000146f is ringing
[2019-09-20 16:06:20] VERBOSE[10888][C-00000bb5] app_dial.c: SIP/195-0000146f is ringing
[2019-09-20 16:06:35] VERBOSE[10888][C-00000bb5] app_dial.c: SIP/195-0000146f answered SIP/185-0000146e
[2019-09-20 16:06:35] VERBOSE[11027][C-00000bb5] bridge_channel.c: Channel SIP/195-0000146f joined 'simple_bridge' basic-bridge <53aee598-22a7-4ad4-a7ef-2153fa863889>
[2019-09-20 16:06:35] VERBOSE[10888][C-00000bb5] bridge_channel.c: Channel SIP/185-0000146e joined 'simple_bridge' basic-bridge <53aee598-22a7-4ad4-a7ef-2153fa863889>
[2019-09-20 16:07:04] VERBOSE[11027][C-00000bb5] bridge_channel.c: Channel SIP/195-0000146f left 'simple_bridge' basic-bridge <53aee598-22a7-4ad4-a7ef-2153fa863889>
[2019-09-20 16:07:04] VERBOSE[10888][C-00000bb5] bridge_channel.c: Channel SIP/185-0000146e left 'simple_bridge' basic-bridge <53aee598-22a7-4ad4-a7ef-2153fa863889>
Ramin Darvishov
  • 1,043
  • 1
  • 15
  • 30

1 Answers1

1

Asterisk itself can do thousand of calls at same moment via same SIP trunk.

If you want limit number of calls per extension, you have write special dialplan which check number of current calls. You can do that using function GROUP/GROUP_COUNT.

arheops
  • 15,544
  • 1
  • 21
  • 27
  • Thanks for answer. It 's not possible that at the same time there would be these two things. Incoming Ringing and execute outgoing call. – Ramin Darvishov Oct 02 '19 at 07:56
  • Asterisk is digital PBX. Forget about your experience with legacy PBX. There are no lines inside, incoming and outgoing calls are fully independed. – arheops Oct 02 '19 at 12:18
  • I do not agree with you. With some parameters may controls incoming and outgoing calls belonging some sip (example: call-limit). – Ramin Darvishov Oct 02 '19 at 12:49
  • calllimit have bug. it limit only one direction. Sure you can not agree with expert, no problem. good luck. – arheops Oct 02 '19 at 16:52
  • "It limit only one direction". it is influence of parameter to Asterisk. – Ramin Darvishov Oct 02 '19 at 18:23
  • Common. I give you solution. You not agree with me. Not a problem, wait for other answers. – arheops Oct 03 '19 at 00:52