chan_mobile и pjsip, тишина

[yandrey]

16.1.1, если использовать sip, все нормально, а pjsip нет звука, в чем может быть причина?

Код: Выделить всё

[Jan 22 14:39:18] DEBUG[15032]: res_pjsip_session.c:762 handle_incoming_sdp: Negotiating incoming SDP media stream 'audio' using audio SDP handler
[Jan 22 14:39:18] DEBUG[15032]: res_pjsip_sdp_rtp.c:236 create_rtp: Transport transport-udp bound to 0.0.0.0: Using it for RTP media.
[Jan 22 14:39:18] DEBUG[15032]: rtp_engine.c:511 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x7ff65401a250'
[Jan 22 14:39:18] DEBUG[15032]: res_rtp_asterisk.c:3509 rtp_allocate_transport: Allocated port 13812 for RTP instance '0x7ff65401a250'
[Jan 22 14:39:18] DEBUG[15032]: rtp_engine.c:528 ast_rtp_instance_new: RTP instance '0x7ff65401a250' is setup and ready to go
[Jan 22 14:39:18] DEBUG[15032]: res_rtp_asterisk.c:7298 ast_rtp_prop_set: Setup RTCP on RTP instance '0x7ff65401a250'
[Jan 22 14:39:18] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 9 based on m type on 0x7ff628a4e310
[Jan 22 14:39:18] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 0 based on m type on 0x7ff628a4e310
[Jan 22 14:39:18] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 8 based on m type on 0x7ff628a4e310
[Jan 22 14:39:18] DEBUG[15032]: rtp_engine.c:1299 ast_rtp_codecs_payloads_set_m_type: Don't have a default tx payload type 2 format for m type on 0x7ff628a4e310
[Jan 22 14:39:18] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 18 based on m type on 0x7ff628a4e310
[Jan 22 14:39:18] DEBUG[15032]: res_pjsip_session.c:775 handle_incoming_sdp: Media stream 'audio' handled by audio
[Jan 22 14:39:18] DEBUG[15032]: res_rtp_asterisk.c:7197 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7ff65401a250'
[Jan 22 14:39:18] DEBUG[15032]: channel.c:989 __ast_channel_alloc_ap: Channel 0x7ff6540235c0 'PJSIP/200-00000004' allocated
[Jan 22 14:39:18] DEBUG[15038][C-00000005]: pbx.c:2917 pbx_extension_helper: Launching 'Dial'
[Jan 22 14:39:18] DEBUG[15038][C-00000005]: channel.c:989 __ast_channel_alloc_ap: Channel 0x7ff66c0035e0 'Mobile/N9005-de3c' allocated
[Jan 22 14:39:18] DEBUG[15038][C-00000005]: chan_mobile.c:992 mbl_call: Calling N9005/+79000000000 on Mobile/N9005-de3c
]Jan 22 14:39:18] DEBUG[15038][C-00000005]: chan_mobile.c:1453 rfcomm_write_full: rfcomm_write() (23) [ATD+79000000000;
[Jan 22 14:39:18] DEBUG[15038][C-00000005]: channel.c:5635 set_format: Channel Mobile/N9005-de3c setting read format path: slin -> slin
[Jan 22 14:39:18] DEBUG[15038][C-00000005]: channel.c:5635 set_format: Channel PJSIP/200-00000004 setting write format path: slin -> ulaw
[Jan 22 14:39:18] DEBUG[15038][C-00000005]: channel.c:5635 set_format: Channel PJSIP/200-00000004 setting read format path: ulaw -> slin
[Jan 22 14:39:18] DEBUG[15038][C-00000005]: channel.c:5635 set_format: Channel Mobile/N9005-de3c setting write format path: slin -> slin
[Jan 22 14:39:20] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read OK
[Jan 22 14:39:20] DEBUG[14536]: chan_mobile.c:3393 handle_response_ok: [N9005] dial sent successfully
[Jan 22 14:39:20] DEBUG[15032]: res_pjsip_session.c:831 handle_negotiated_sdp_session_media: Applying negotiated SDP media stream 'audio' using audio SDP handler
[Jan 22 14:39:20] DEBUG[15032]: res_rtp_asterisk.c:7197 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x7ff65401a250'
[Jan 22 14:39:20] DEBUG[15032]: res_rtp_asterisk.c:7369 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x7ff65401a250'
[Jan 22 14:39:20] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 9 based on m type on 0x7ff628a4dfb0
[Jan 22 14:39:20] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 0 based on m type on 0x7ff628a4dfb0
[Jan 22 14:39:20] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 8 based on m type on 0x7ff628a4dfb0
[Jan 22 14:39:20] DEBUG[15032]: rtp_engine.c:1299 ast_rtp_codecs_payloads_set_m_type: Don't have a default tx payload type 2 format for m type on 0x7ff628a4dfb0
[Jan 22 14:39:20] DEBUG[15032]: rtp_engine.c:1304 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 18 based on m type on 0x7ff628a4dfb0
[Jan 22 14:39:20] DEBUG[15032]: res_pjsip_session.c:836 handle_negotiated_sdp_session_media: Applied negotiated SDP media stream 'audio' using audio SDP handler
[Jan 22 14:39:20] DEBUG[14477]: chan_mobile.c:1913 sco_accept: Incoming Audio Connection from device 00:00:00:00:00:00 MTU is 64
[Jan 22 14:39:20] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 2,2
[Jan 22 14:39:20] DEBUG[14536]: chan_mobile.c:3616 handle_response_ciev: [N9005] outgoing call
[Jan 22 14:39:20] DEBUG[15038][C-00000005]: res_rtp_asterisk.c:4884 ast_rtp_write: Ooh, format changed from none to ulaw
[Jan 22 14:39:20] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 2,3
[Jan 22 14:39:20] DEBUG[14536]: chan_mobile.c:3624 handle_response_ciev: [N9005] remote alerting
[Jan 22 14:39:21] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,0
[Jan 22 14:39:21] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,2
[Jan 22 14:39:24] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,3
[Jan 22 14:39:24] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,4
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: chan_mobile.c:1136 mbl_read: [N9005] read error 104, going to wait for new connection
[Jan 22 14:39:34] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 2,0
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: channel.c:2566 ast_hangup: Channel 0x7ff66c0035e0 'Mobile/N9005-de3c' hanging up.  Refs: 2
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: chan_mobile.c:1037 mbl_hangup: [N9005] hanging up device
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: channel.c:2211 ast_channel_destructor: Channel 0x7ff66c0035e0 'Mobile/N9005-de3c' destroying
[Jan 22 14:39:34] DEBUG[14460]: cdr.c:1463 cdr_object_finalize: Finalized CDR for PJSIP/200-00000004 - start 1548142758.580472 answer 0.000000 end 1548142774.580882 dispo BUSY
[Jan 22 14:39:34] DEBUG[14460]: cdr.c:1463 cdr_object_finalize: Finalized CDR for Mobile/N9005-de3c - start 1548142758.580751 answer 0.000000 end 1548142774.580897 dispo BUSY
[Jan 22 14:39:34] DEBUG[14460]: cdr.c:1296 cdr_object_create_public_records: CDR for Mobile/N9005-de3c is dialed and has no Party B; discarding
[Jan 22 14:39:34] DEBUG[14451]: chan_mobile.c:1217 mbl_devicestate: Checking device state for device N9005
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: app_dial.c:3215 dial_exec_full: Exiting with DIALSTATUS=BUSY.
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: pbx.c:2917 pbx_extension_helper: Launching 'Hangup'
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: channel.c:2475 ast_softhangup_nolock: Soft-Hanging (0x20) up channel 'PJSIP/200-00000004'
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: pbx.c:4392 __ast_pbx_run: Spawn extension (phones,79000000000,2) exited non-zero on 'PJSIP/200-00000004'
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: channel.c:2475 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'PJSIP/200-00000004'
[Jan 22 14:39:34] DEBUG[15038][C-00000005]: channel.c:2566 ast_hangup: Channel 0x7ff6540235c0 'PJSIP/200-00000004' hanging up.  Refs: 2
[Jan 22 14:39:34] DEBUG[15032]: rtp_engine.c:440 instance_destructor: Destroyed RTP instance '0x7ff65401a250'
[Jan 22 14:39:34] DEBUG[15032]: channel.c:2211 ast_channel_destructor: Channel 0x7ff6540235c0 'PJSIP/200-00000004' destroying
[Jan 22 14:39:34] DEBUG[14460]: cdr.c:1463 cdr_object_finalize: Finalized CDR for PJSIP/200-00000004 - start 1548142774.580982 answer 0.000000 end 1548142774.581270 dispo BUSY
[Jan 22 14:39:34] DEBUG[14460]: cdr.c:3507 post_cdr: Skipping CDR for PJSIP/200-00000004 since we weren't answered
[Jan 22 14:39:35] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,0
[Jan 22 14:39:35] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,3
[Jan 22 14:39:35] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,2
[Jan 22 14:39:38] DEBUG[14536]: chan_mobile.c:3926 do_monitor_phone: [N9005] read +CIEV: 4,1
[Jan 22 14:39:54] DEBUG[15043]: threadpool.c:1166 worker_idle: Worker thread idle timeout reached. Dying.

pjsip

Код: Выделить всё

[transport-udp]
type=transport
protocol=udp
bind=0.0.0.0

[200]
type=auth
auth_type=userpass
username=200
password=123456

[200]
type=aor
max_contacts=1

[200]
type=endpoint
transport=transport-udp
context=phones
auth=200
aors=200
disallow=all
allow=ulaw
allow=alaw
allow=g722
allow=g726
allow=g729

sip

Код: Выделить всё

[200]
type=friend
defaultuser=200
secret=123456
context=phones
host=dynamic
nat=no
canreinvite=no
disallow=all
allow=ulaw
allow=alaw
allow=g722
allow=g726
allow=g729

[ded]

1) Сделайте тестовую терминацию каждого плеча на эхо тест, последовательно,
exten => _X.,1,Answer
exten => _X.,n,Echo()
и протестируйте работу каждого плеча со звуком.

2) При этом полезно включить rtp set debug on
чтобы видеть, собственно, голос - в обе стороны? В одну сторону?

3) PJSIP сложнее, чем старый добрый chan_sip, зачем он вам?

4) http://asterisk.ru/knowledgebase/debug

[yandrey]

1,2) rtp ходит в обе стороны, между внутренними номерами и с внешним голос есть во все стороны
оказалось что молчит только входящий с chan_mobile голос через gigaset ip, а софтовым sip телефоном проблемы нет, смена кодека не помогает, наверно, что то с потоком и гигасет не может его воспроизвести

3) особой нужды в PJSIP нет, ради современности, интереса и опыта перешел на него

[ded]

1) chan_mobile - реликт и архаизм, все юзают chan_dongle

[yandrey]

Ниже, в теме про еще одну особенность chan_mobile не давать засыпать всем ядрам cpu, я писал почему он не заменим.

Буду искать что-нибудь на intel atom или под openwrt, персональное для chan_sip + chan_mobile.

[ded]

Это уже музейные вещи. Я, правда, тоже собираю
У меня есть загружаемая Live USB-флэш, на которой SPBlinux + иксы, браузер, почта, и Астериск без вэб-оболочки. И это всё работает.
Размер флэшки - 128 МБ