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Help. Неожиданно пропала передача звука на всех протокалах.

sn
Сообщений: 6

Help. Неожиданно пропала передача звука на всех протокалах.

Чудеса творятся. Еще вчера вечером все работало.
Сегодня утром на при звонках SIP - SIP, ни при SIP - ZAP, на при IAX2 - SIP нет звука.
Т.е. соединение происходит, но абоненты друг друга не слышат.

Подскажите на что обратить внимание.

На всякий случай логи:

--- SIP - SIP----
-- Executing Dial("SIP/315-c99c", "SIP/314|15|rtTwW") in new stack
We're at 192.168.0.100 port 14860
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.0.114:5060:
INVITE sip:314@192.168.0.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e;rport
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
To: <sip:314@192.168.0.114:5060>
Contact: <sip:315@192.168.0.100>
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 25 Jan 2006 11:27:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 3685 3685 IN IP4 192.168.0.100
s=session
c=IN IP4 192.168.0.100
t=0 0
m=audio 14860 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 314

<-- SIP read from 192.168.0.114:5060:
SIP/2.0 100 Trying
To: <sip:314@192.168.0.114:5060>
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e
Server: Sipura/SPA841-0.9.1
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 192.168.0.114:5060:
SIP/2.0 180 Ringing
To: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e
Server: Sipura/SPA841-0.9.1
Call-Info: <sip:192.168.0.100>;appearance-index=1
Content-Length: 0


--- (9 headers 0 lines)---
-- SIP/314-7cb8 is ringing

<-- SIP read from 192.168.0.114:5060:
SIP/2.0 200 OK
To: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e
Contact: 314 <sip:314@192.168.0.114:5060>
Server: Sipura/SPA841-0.9.1
Call-Info: <sip:192.168.0.100>;appearance-index=1
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 651124 651124 IN IP4 192.168.0.114
s=-
c=IN IP4 192.168.0.114
t=0 0
m=audio 16468 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.114:16468
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:314@192.168.0.114:5060>
set_destination: Parsing <sip:314@192.168.0.114:5060> for address/port to send to
set_destination: set destination to 192.168.0.114, port 5060
Transmitting (no NAT) to 192.168.0.114:5060:
ACK sip:314@192.168.0.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK78a80a73;rport
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
To: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
Contact: <sip:315@192.168.0.100>
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/314-7cb8 answered SIP/315-c99c
-- Attempting native bridge of SIP/315-c99c and SIP/314-7cb8

<-- SIP read from 192.168.0.114:5060:
BYE sip:315@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.114:5060;branch=z9hG4bK-5a5611dd
From: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
To: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Sipura/SPA841-0.9.1
Content-Length: 0


--- (9 headers 0 lines)---
Sending to 192.168.0.114 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.114:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.114:5060;branch=z9hG4bK-5a5611dd;received=192.168.0.114
From: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
To: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:315@192.168.0.100>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
== Spawn extension (sn, 314, 1) exited non-zero on 'SIP/315-c99c'
Destroying call '330d1fa43101f828640f584e058174ee@192.168.0.100'

----конец -----------------------


2006-01-25 11:36

sn
Сообщений: 6

Re: Help. Неожиданно пропала передача звука на всех протокалах.

Простите. Повторю еще раз.

Чудеса творятся. Еще вчера вечером все работало.
Сегодня утром на при звонках SIP - SIP, ни при SIP - ZAP, на при IAX2 - SIP нет звука.
Т.е. соединение происходит, но абоненты друг друга не слышат.

Подскажите на что обратить внимание. Заранее спасибо.

2006-01-25 11:37

sn
Сообщений: 6

Re: Help. Неожиданно пропала передача звука на всех протокалах.

На всякий случай логи:

--- SIP - SIP----
-- Executing Dial("SIP/315-c99c", "SIP/314|15|rtTwW") in new stack
We're at 192.168.0.100 port 14860
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
13 headers, 10 lines
Reliably Transmitting (no NAT) to 192.168.0.114:5060:
INVITE sip:314@192.168.0.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e;rport
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
To: <sip:314@192.168.0.114:5060>
Contact: <sip:315@192.168.0.100>
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Wed, 25 Jan 2006 11:27:20 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 216

v=0
o=root 3685 3685 IN IP4 192.168.0.100
s=session
c=IN IP4 192.168.0.100
t=0 0
m=audio 14860 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -

---
-- Called 314

<-- SIP read from 192.168.0.114:5060:
SIP/2.0 100 Trying
To: <sip:314@192.168.0.114:5060>
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e
Server: Sipura/SPA841-0.9.1
Content-Length: 0


--- (8 headers 0 lines)---

<-- SIP read from 192.168.0.114:5060:
SIP/2.0 180 Ringing
To: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e
Server: Sipura/SPA841-0.9.1
Call-Info: <sip:192.168.0.100>;appearance-index=1
Content-Length: 0


--- (9 headers 0 lines)---
-- SIP/314-7cb8 is ringing

<-- SIP read from 192.168.0.114:5060:
SIP/2.0 200 OK
To: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 INVITE
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK3938000e
Contact: 314 <sip:314@192.168.0.114:5060>
Server: Sipura/SPA841-0.9.1
Call-Info: <sip:192.168.0.100>;appearance-index=1
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 651124 651124 IN IP4 192.168.0.114
s=-
c=IN IP4 192.168.0.114
t=0 0
m=audio 16468 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:30
a=sendrecv

--- (12 headers 11 lines)---
Found RTP audio format 0
Found RTP audio format 101
Peer audio RTP is at port 192.168.0.114:16468
Found description format PCMU
Found description format telephone-event
Capabilities: us - 0x4 (ulaw), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
list_route: hop: <sip:314@192.168.0.114:5060>
set_destination: Parsing <sip:314@192.168.0.114:5060> for address/port to send to
set_destination: set destination to 192.168.0.114, port 5060
Transmitting (no NAT) to 192.168.0.114:5060:
ACK sip:314@192.168.0.114:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.100:5060;branch=z9hG4bK78a80a73;rport
From: "315" <sip:315@192.168.0.100>;tag=as1f332df8
To: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
Contact: <sip:315@192.168.0.100>
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/314-7cb8 answered SIP/315-c99c
-- Attempting native bridge of SIP/315-c99c and SIP/314-7cb8

<-- SIP read from 192.168.0.114:5060:
BYE sip:315@192.168.0.100 SIP/2.0
Via: SIP/2.0/UDP 192.168.0.114:5060;branch=z9hG4bK-5a5611dd
From: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
To: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 101 BYE
Max-Forwards: 70
User-Agent: Sipura/SPA841-0.9.1
Content-Length: 0


--- (9 headers 0 lines)---
Sending to 192.168.0.114 : 5060 (non-NAT)
Transmitting (no NAT) to 192.168.0.114:5060:
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.0.114:5060;branch=z9hG4bK-5a5611dd;received=192.168.0.114
From: <sip:314@192.168.0.114:5060>;tag=18aba2456e47dc72i0
To: "315" <sip:315@192.168.0.100>;tag=as1f332df8
Call-ID: 330d1fa43101f828640f584e058174ee@192.168.0.100
CSeq: 101 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Contact: <sip:315@192.168.0.100>
Content-Length: 0
X-Asterisk-HangupCause: Normal Clearing


---
== Spawn extension (sn, 314, 1) exited non-zero on 'SIP/315-c99c'
Destroying call '330d1fa43101f828640f584e058174ee@192.168.0.100'

----конец -----------------------


2006-01-25 11:39

meral
Откуда: Киев
Сообщений: 749

Re: Help. Неожиданно пропала передача звука на всех протокалах.

не поверишь - у меня та же фигня)) версия 1.2.2
лечиться так - берешь 1.2.1 - на ней работает
я не понял почему
заметь при этом звук от астериска к коробке работет
не работает сразу после того как сделаеться bridge
похоже на чьюто шутку в связи с днем студента

http://www.getafreelancer.com/users/feedback_385837.html
2006-01-25 13:28

abab
Сообщений: 1

Re: Help. Неожиданно пропала передача звука на всех протокалах.

Баг версии 1.2.2
0006349: [patch] No audio on bridges from jan 25 2006
уже описан есть пач
http://bugs.digium.com/view.php?id=6349
быстрое решение ОТКАТИТЬ ДАТУ НА СЕРВАКЕ!!!
нормально решение поставить пач или откат на версию 1.2.1
Сильный BUG !!! у нас соседи попали, у самих стояло 1.2.1 (пронесло)
2006-01-25 13:34

sn
Сообщений: 6

Re: Help. Неожиданно пропала передача звука на всех протокалах.

Ребята, большое спасибо!
2006-01-25 14:01

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