Asterisk, calls are torn

Question

Asterisk

There is a server with an e1 card, which has an e1 stream from the beeline. There is a problem in the form of periodically torn calls to cell phones during outgoing / incoming communications.

A long ups and downs with the beeline led to the fact that the tracing from both the central switch and the switch to which we are directly connected, according to them, says that we are tearing the session.

Logs and settings are posted here , as there is a limit on the text in questions.

Additionally, there is another sip provider on the server - everything works fine through it, the same phones, the same numbers through the same dialplan.

Answer 1

I can not get used to the interface, in general, here:

[Nov 13 15:10:24] DEBUG[31363] channel.c: Got a FRAME_CONTROL (8) frame on channel DAHDI/i1/989*********-1010
[Nov 13 15:10:24] DEBUG[31363] res_rtp_asterisk.c: Setting the marker bit due to a source update
[Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Requested indication 20 on channel DAHDI/i1/989*********-1010
[Nov 13 15:10:24] DEBUG[31363] channel.c: Bridge stops bridging channels SIP/117-00004715 and DAHDI/i1/989*********-1010
[Nov 13 15:10:24] DEBUG[31363] channel.c: Soft-Hanging up channel 'SIP/117-00004715'
[Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Disabled echo cancellation on channel 1
[Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Set option TDD MODE, value: OFF(0) on DAHDI/i1/989*********-1010
[Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Updated conferencing on 1, with 0 conference users
[Nov 13 15:10:24] DEBUG[31363] chan_dahdi.c: Set option AUDIO MODE, value: OFF(0) on DAHDI/i1/989*********-1010
[Nov 13 15:10:24] VERBOSE[31363] chan_dahdi.c:     -- Hungup 'DAHDI/i1/989*********-1010'
[Nov 13 15:10:24] DEBUG[31363] app_dial.c: Exiting with DIALSTATUS=ANSWER.
[Nov 13 15:10:24] DEBUG[31363] pbx.c: Spawn extension (office,9989*********,3) exited non-zero on 'SIP/117-00004715'
[Nov 13 15:10:24] VERBOSE[31363] pbx.c:   == Spawn extension (office, 9989*********, 3) exited non-zero on 'SIP/117-00004715'
[Nov 13 15:10:24] DEBUG[31363] channel.c: Soft-Hanging up channel 'SIP/117-00004715'
[Nov 13 15:10:24] DEBUG[31363] channel.c: Hanging up channel 'SIP/117-00004715'
[Nov 13 15:10:24] DEBUG[31363] chan_sip.c: Hanging up zombie call. Be scared.
[Nov 13 15:10:24] DEBUG[31363] chan_sip.c: Updating call counter for incoming call
[Nov 13 15:10:24] DEBUG[31363] res_rtp_asterisk.c: Setting RTCP address on RTP instance '0x7f27dc155f38'
[Nov 13 15:10:24] DEBUG[31363] netsock2.c: Splitting '192.168.5.132:5060' into...
[Nov 13 15:10:24] DEBUG[31363] netsock2.c: ...host '192.168.5.132' and port '5060'.
[Nov 13 15:10:24] DEBUG[31363] chan_sip.c: Trying to put 'BYE sip:117' onto UDP socket destined for 192.168.5.132:5060

Answer 2

There is little data, the only thing that can be seen from the log is that the call is bounced in your direction with a normal exit code.
grep "\[31363" /var/log/asterisk/full