AF
Asterisk Forum
обсуждения телефонии, VoIP и IP-PBX
12разделов
5 423тем
34 385сообщений
← К списку тем

record.wav

Asterisk IP PBX 12 сообщений -
#1
Дано: asterisk 1.0.9
ОС: Linux Gentoo 2005.0
Ядро: 2.6.11-gentoo-r3

Никак не могу разобраться со следующей багой, подозреваю что дело в звуковой карте.
Пытаюсь записать голос с телефонной трубки:

=== cut ===
*CLI> -- Executing Answer("OH323/R17736", "") in new stack
-- Executing VoiceMail("OH323/R17736", "297101") in new stack
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/other/297101/INBOX/msg0005 format: wav49, 0xb6b01318
-- x=1, open writing: /var/spool/asterisk/voicemail/other/297101/INBOX/msg0005 format: gsm, 0xb6b01178
-- x=2, open writing: /var/spool/asterisk/voicemail/other/297101/INBOX/msg0005 format: wav, 0xb6b051b0
Killed
Press any key to continue...
=== cut ===

Звукашка определяется след. образом:

intel8x0_measure_ac97_clock: measured 50046 usecs
intel8x0: clocking to 48000
ALSA device list:
#0: Intel ICH5 with ALC658 at 0xfc001000, irq 17

Помогите люди добрые Smile
_________________
I want to change the world! (c)
#2
1) что написано в логах Asterisk'a?
2) Происходит ли тоже самое при записи только в один из форматов, а не в 3 сразу?
#3
вы меня оба удивляете Wink
при чем тут звуковая карта вообще если пишем звук не с микрофона а с телефона?? Shocked
p.s. imho если звук не нужен в консоли то на звуковуху вообще можно забить (отключить в биосе).
_________________
«Choose a job you love, and you will never have to work a day in your life» — Confucius
#4
Поставил ядро 2.4
#uname -a
Linux asterisk 2.4.28-gentoo-r8

Теперь при записи вываливается с криками:

=== cut ===
-- Executing Answer("OH323/R4767", ""Wink in new stack
-- Executing VoiceMail("OH323/R4767", "297101"Wink in new stack
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/other/297101/INBOX/msg0012 format: gsm, 0x8180598
Segmentation fault
=== cut ===

Иногда говорит вот это:

=== cut ===
-- x=0, open writing: /var/spool/asterisk/voicemail/other/297101/INBOX/msg0013 format: gsm, 0x8166fa0
Ouch ... error while writing audio data: : Broken pipe
Segmentation fault
=== cut ===

В "/var/log/asterisk/messages" ничего по этому поводу не пишется Sad

Где грабли? Sad
_________________
I want to change the world! (c)
#5
на мой взгляд это ошибка в mpg123
если достаточно опыта чтобы ставить генту то я думаю что его хватит и на астериск под ним. в любом случае нехилый секс гарантирован Wink

хотя в данном случае буквально 15 секунд гугления выдал твой диагноз:
You probably have a configuration error.

http://lists.digium.com/pipermail/asteri ... 08393.html
_________________
«Choose a job you love, and you will never have to work a day in your life» — Confucius
#6
еще может это поможет:
Код:
+-------------------------------------------+
+ +
+ ** NOTE FOR DOWNGRADING FROM CVS HEAD ** +
+ +
+ If you are downgrading from CVS HEAD to +
+ a stable release, remember to delete +
+ everything from your asterisk modules +
+ directory (/usr/lib/asterisk/modules/) +
+ and the asterisk header directory +
+ (/usr/include/asterisk/) +
+ before doing a 'make install'. +
+ +
+-------------------------------------------+

_________________
«Choose a job you love, and you will never have to work a day in your life» — Confucius
#7
В общем, откатился я обратно на 2.6, думаю дело не в ядре (?)
#uname -a
Linux asterisk 2.6.11-gentoo-r3

Решил сделать так называемый "asterisk mpg123 fake" - http://www.voip-info.org/tiki-index.php? ... aking%20it

Заменил его cat`ом и переконвертил mp3-ки в .raw формат.
Теже "яйца", вид сбоку:

=== cut ===
-- Executing Answer("OH323/R11783", "") in new stack
-- Executing VoiceMail("OH323/R11783", "297101") in new stack
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing: /var/spool/asterisk/voicemail/other/297101/INBOX/msg0000 format: wav, 0xb7b67a08
cat: write error: Broken pipe
Killed
=== cut ===

То же самое говорит, если звонить с SIP'а.

Привожу листинг команд:
=== cut ===
#lspci
0000:00:00.0 Host bridge: Intel Corp. 82865G/PE/P DRAM Controller/Host-Hub Interface (rev 02)
0000:00:01.0 PCI bridge: Intel Corp. 82865G/PE/P PCI to AGP Controller (rev 02)
0000:00:1d.0 USB Controller: Intel Corp. 82801EB/ER (ICH5/ICH5R) USB UHCI Controller #1 (rev 02)
0000:00:1d.1 USB Controller: Intel Corp. 82801EB/ER (ICH5/ICH5R) USB UHCI Controller #2 (rev 02)
0000:00:1d.2 USB Controller: Intel Corp. 82801EB/ER (ICH5/ICH5R) USB UHCI #3 (rev 02)
0000:00:1d.3 USB Controller: Intel Corp. 82801EB/ER (ICH5/ICH5R) USB UHCI Controller #4 (rev 02)
0000:00:1d.7 USB Controller: Intel Corp. 82801EB/ER (ICH5/ICH5R) USB2 EHCI Controller (rev 02)
0000:00:1e.0 PCI bridge: Intel Corp. 82801 PCI Bridge (rev c2)
0000:00:1f.0 ISA bridge: Intel Corp. 82801EB/ER (ICH5/ICH5R) LPC Interface Bridge (rev 02)
0000:00:1f.1 IDE interface: Intel Corp. 82801EB/ER (ICH5/ICH5R) IDE Controller (rev 02)
0000:00:1f.3 SMBus: Intel Corp. 82801EB/ER (ICH5/ICH5R) SMBus Controller (rev 02)
0000:00:1f.5 Multimedia audio controller: Intel Corp. 82801EB/ER (ICH5/ICH5R) AC'97 Audio Controller (rev 02)
0000:01:00.0 VGA compatible controller: ATI Technologies Inc RV280 [Radeon 9200] (rev 01)
0000:01:00.1 Display controller: ATI Technologies Inc RV280 [Radeon 9200] (Secondary) (rev 01)
0000:02:01.0 Ethernet controller: Realtek Semiconductor Co., Ltd. RTL-8139/8139C/8139C+ (rev 10)

#cat /proc/interrupts
CPU0 CPU1
0: 259074467 0 IO-APIC-edge timer
1: 8 0 IO-APIC-edge i8042
7: 0 0 IO-APIC-edge parport0
9: 0 0 IO-APIC-level acpi
14: 2861 0 IO-APIC-edge ide0
15: 13 0 IO-APIC-edge ide1
16: 0 0 IO-APIC-level uhci_hcd, uhci_hcd
17: 0 0 IO-APIC-level Intel ICH5
18: 0 0 IO-APIC-level uhci_hcd
19: 0 0 IO-APIC-level uhci_hcd
21: 363922 0 IO-APIC-level eth0
23: 0 0 IO-APIC-level ehci_hcd
NMI: 0 0
LOC: 259083765 259083764
ERR: 0
MIS: 0
=== cut ===

Надеюсь на вашу помощь Smile
_________________
I want to change the world! (c)
#8
2.: kuzdu не ставил, гента с "нуля" Smile Только midnight commander Smile
_________________
I want to change the world! (c)
#9
Пытаюсь звонить c SIP устройства: (включен sip debug)

=== cut ===
-- Saved useragent "Linksys/PAP2-2.0.10(LSc)" for peer siplinksys

*CLI> sip debug
SIP Debugging Enabled
*CLI>

Sip read:
INVITE sip:297101@xxx.xxx.xxx.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.3:5060;branch=z9hG4bK-d432f28e
From: siplinksys ;tag=c34a0fe7fba9e426o0
To:
Call-ID: 16b437bb-dd87be62@192.168.xxx.3
CSeq: 101 INVITE
Max-Forwards: 70
Contact: siplinksys
Expires: 240
User-Agent: Linksys/PAP2-2.0.10(LSc)
Content-Length: 399
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: x-sipura
Content-Type: application/sdp

v=0
o=- 86595682 86595682 IN IP4 192.168.xxx.3
s=-
c=IN IP4 192.168.xxx.3
t=0 0
m=audio 16478 RTP/AVP 18 0 2 4 8 96 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

14 headers, 18 lines
Using latest request as basis request
Sending to 192.168.xxx.3 : 5060 (non-NAT)
Reliably Transmitting (no NAT):
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.xxx.3:5060;branch=z9hG4bK-d432f28e
From: siplinksys ;tag=c34a0fe7fba9e426o0
To: ;tag=as31b5ea16
Call-ID: 16b437bb-dd87be62@192.168.xxx.3
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Proxy-Authenticate: Digest realm="asterisk", nonce="1b32e34f"
Content-Length: 0


to 192.168.xxx.3:5060
Scheduling destruction of call '16b437bb-dd87be62@192.168.xxx.3' in
15000 ms
Found user 'siplinksys'


Sip read:
ACK sip:297101@xxx.xxx.xxx.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.3:5060;branch=z9hG4bK-d432f28e
From: siplinksys ;tag=c34a0fe7fba9e426o0
To: ;tag=as31b5ea16
Call-ID: 16b437bb-dd87be62@192.168.xxx.3
CSeq: 101 ACK
Max-Forwards: 70
Contact: siplinksys
User-Agent: Linksys/PAP2-2.0.10(LSc)
Content-Length: 0


10 headers, 0 lines


Sip read:
INVITE sip:297101@xxx.xxx.xxx.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.3:5060;branch=z9hG4bK-ab77451a
From: siplinksys ;tag=c34a0fe7fba9e426o0
To:
Call-ID: 16b437bb-dd87be62@192.168.xxx.3
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest
username="siplinksys",realm="asterisk",nonce="1b32e34f",uri="sip:297101@xxx.xxx.xxx.39",algorithm=MD5,response="4a55ce825a7aa5938963ad30ac30641f"
Contact: siplinksys
Expires: 240
User-Agent: Linksys/PAP2-2.0.10(LSc)
Content-Length: 399
Content-Type: application/sdp

v=0
o=- 86595682 86595682 IN IP4 192.168.xxx.3
s=-
c=IN IP4 192.168.xxx.3
t=0 0
m=audio 16478 RTP/AVP 18 0 2 4 8 96 98 100 101
a=rtpmap:18 G729a/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:100 NSE/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

13 headers, 18 lines
Using latest request as basis request
Sending to 192.168.xxx.3 : 5060 (non-NAT)
Found user 'siplinksys'
Found RTP audio format 18
Found RTP audio format 0
Found RTP audio format 2
Found RTP audio format 4
Found RTP audio format 8
Found RTP audio format 96
Found RTP audio format 98
Found RTP audio format 100
Found RTP audio format 101
Peer audio RTP is at port 192.168.xxx.3:16478
Found description format G729a
Found description format PCMU
Found description format G726-32
Found description format G723
Found description format PCMA
Found description format G726-40
Found description format G726-16
Found description format NSE
Found description format telephone-event
Capabilities: us - 0x8010e (gsm|ulaw|alaw|g729|h263), peer - audio=0x11d
(g723|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x10c
(ulaw|alaw|g729)
Non-codec capabilities: us - 0x1 (g723), peer - 0x1 (g723), combined -
0x1 (g723)
Looking for 297101 in internal
list_route: hop:
Transmitting (no NAT):
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.xxx.3:5060;branch=z9hG4bK-ab77451a
From: siplinksys ;tag=c34a0fe7fba9e426o0
To:
Call-ID: 16b437bb-dd87be62@192.168.xxx.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Length: 0


to 192.168.xxx.3:5060
-- Executing Answer("SIP/siplinksys-a460", ""Wink in new stack
We're at xxx.xxx.xxx.39 port 17790
Answering with preferred capability 0x100 (g729)
Answering with capability 0x2 (gsm)
Answering with capability 0x4 (ulaw)
Answering with capability 0x8 (alaw)
Answering with non-codec capability 0x1 (telephone-event)
Reliably Transmitting (no NAT):
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.xxx.3:5060;branch=z9hG4bK-ab77451a
From: siplinksys ;tag=c34a0fe7fba9e426o0
To: ;tag=as37e83764
Call-ID: 16b437bb-dd87be62@192.168.xxx.3
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER
Contact:
Content-Type: application/sdp
Content-Length: 287

v=0
o=root 7585 7585 IN IP4 xxx.xxx.xxx.39
s=session
c=IN IP4 xxx.xxx.xxx.39
t=0 0
m=audio 17790 RTP/AVP 18 3 0 8 101
a=rtpmap:18 G729/8000
a=rtpmap:3 GSM/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

to 192.168.xxx.3:5060
-- Executing VoiceMail("SIP/siplinksys-a460", "297101"Wink in new stack
-- Playing 'vm-intro' (language 'en')


Sip read:
ACK sip:297101@xxx.xxx.xxx.39 SIP/2.0
Via: SIP/2.0/UDP 192.168.xxx.3:5060;branch=z9hG4bK-ab057897
From: siplinksys ;tag=c34a0fe7fba9e426o0
To: ;tag=as37e83764
Call-ID: 16b437bb-dd87be62@192.168.xxx.3
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest
username="siplinksys",realm="asterisk",nonce="1b32e34f",uri="sip:297101@xxx.xxx.xxx.39",algorithm=MD5,response="8c11e61baa16bf10c3cea84e8d1a6f83"
Contact: siplinksys
User-Agent: Linksys/PAP2-2.0.10(LSc)
Content-Length: 0


11 headers, 0 lines
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/other/297101/INBOX/msg0006 format: wav,
0x8153588
Killed
=== cut ===
_________________
I want to change the world! (c)
#10
очень. очень странный глюк..

попробуй переставить астерсик но конфиги на свои старые не заменять. тоесть все сделать с нуля. у меня было что астериск валился с мессагой Broken pipe как у тебя с (старым) кривым конфигом. но правда не во время записи а сразу на старте. а без Monitor всё работает нормально? если так то остается только сам monitor выходит..
как бы повторить твою ситуацию у себя - помог бы подебажить..
_________________
«Choose a job you love, and you will never have to work a day in your life» — Confucius
#11
2anest: переставил asterisk с нуля - толку ноль Sad
_________________
I want to change the world! (c)
#12
Проблема решилась! Дело было в кодеке, оптимизированным под pIV...
Спасибо anest за помощь!
_________________
I want to change the world! (c)