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Обрыв звонка и Loose-Route - нужна помощь

Newbies/FAQ Forum 18 сообщений 19.11.2010 15:22 - 24.11.2010 10:33
#1 19.11.2010 15:22

Обрыв звонка и Loose-Route - нужна помощь

Когда-то поднимал тему с обрывом звонков, но никто не помог. Стал отслеживать что падает в консоль при таком звонке из вне, которые идет по SIP провайдеру и попадает на Наш астериск и дальше уже на внутренний номер.

Quote:
Девушка на телефоне поднимает трубку и звонок обрывается


Вот так она описывает, что слышет.

Ниже лог, что вижу я. Только не могу понять - это проблема у меня, или проблема из-за настроек провайдера.

Структура:
Хард телефон девушки подключен к Астериску без ната. Астериск как SIP клиент подключен к SIP провайдеру, Астериск находится за NAT.

Помогите определить кто виновник проблемы обрыва звонков и решить её.


Code:
voip*CLI>

INVITE sip:s@10.201.24.243 SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bK6a83.16f68c87.0
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=7F3E11ED-B95
To:
Date: Fri, 19 Nov 2010 12:59:58 GMT
Call-ID: BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 3211927788-4078113247-2515066006-4276178239
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 5
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Timestamp: 1290171598
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 518

v=0
o=CiscoSystemsSIP-GW-UserAgent 8363 8500 IN IP4 87.248.160.13
s=SIP Call
c=IN IP4 89.28.56.228
t=0 0
m=audio 58762 RTP/AVP 8 0 18 4 98 99 2 15 102 3 101
c=IN IP4 89.28.56.228
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:98 G726-16/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:102 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


--- (22 headers 21 lines) ---
Sending to 89.28.56.226 : 5060 (no NAT)
Using INVITE request as basis request - BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
Found peer 'starnet1'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 102
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 99
Found audio description format G726-32 for ID 2
Found unknown media description format G728 for ID 15
Found unknown media description format GSM-EFR for ID 102
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.28.56.228:58762
Looking for s in office (domain 10.201.24.243)
list_route: hop:


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bK6a83.16f68c87.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
Record-Route:
From: ;tag=7F3E11ED-B95
To:
Call-ID: BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Length: 0



Audio is at 10.201.24.243 port 15908
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.201.24.22:5060:
INVITE sip:5703@10.201.24.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK161f68ec;rport
From: "022922105" ;tag=as4525ae3b
To:
Contact:
Call-ID: 55a4de9b10426e6373b468dc5beef63a@10.201.24.243
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Fri, 19 Nov 2010 13:05:50 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 498

v=0
o=root 967 967 IN IP4 10.201.24.243
s=session
c=IN IP4 10.201.24.243
t=0 0
m=audio 15908 RTP/AVP 0 3 8 112 5 10 7 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
Audio is at 10.201.24.243 port 17182
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bK6a83.16f68c87.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
Record-Route:
From: ;tag=7F3E11ED-B95
To: ;tag=as3a3512b1
Call-ID: BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 967 967 IN IP4 10.201.24.243
s=session
c=IN IP4 10.201.24.243
t=0 0
m=audio 17182 RTP/AVP 3 0 8 101
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 - - - -
a=ptime:20
a=sendrecv


voip*CLI>

SIP/2.0 100 Trying
To:
From: "022922105" ;tag=as4525ae3b
Call-ID: 55a4de9b10426e6373b468dc5beef63a@10.201.24.243
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK161f68ec
Server: Sipura/SPA941-4.1.8
Content-Length: 0



--- (8 headers 0 lines) ---
voip*CLI>

SIP/2.0 180 Ringing
To: ;tag=764f886515716461i0
From: "022922105" ;tag=as4525ae3b
Call-ID: 55a4de9b10426e6373b468dc5beef63a@10.201.24.243
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK161f68ec
Server: Sipura/SPA941-4.1.8
Content-Length: 0



--- (8 headers 0 lines) ---
voip*CLI>

SIP/2.0 200 OK
To: ;tag=764f886515716461i0
From: "022922105" ;tag=as4525ae3b
Call-ID: 55a4de9b10426e6373b468dc5beef63a@10.201.24.243
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK161f68ec
Contact: "Call-Center"
Server: Sipura/SPA941-4.1.8
Content-Length: 208
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp

v=0
o=- 8595203 8595203 IN IP4 10.201.24.22
s=-
c=IN IP4 10.201.24.22
t=0 0
m=audio 16406 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv


--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.201.24.22:16406
list_route: hop:
set_destination: Parsing for address/port to send to
set_destination: set destination to 10.201.24.22, port 5060
Transmitting (no NAT) to 10.201.24.22:5060:
ACK sip:5703@10.201.24.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK5c2813f8;rport
From: "022922105" ;tag=as4525ae3b
To: ;tag=764f886515716461i0
Contact:
Call-ID: 55a4de9b10426e6373b468dc5beef63a@10.201.24.243
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
Audio is at 10.201.24.243 port 17182
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bK6a83.16f68c87.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
Record-Route:
From: ;tag=7F3E11ED-B95
To: ;tag=as3a3512b1
Call-ID: BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 967 968 IN IP4 10.201.24.243
s=session
c=IN IP4 10.201.24.243
t=0 0
m=audio 17182 RTP/AVP 3 0 8 101
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 - - - -
a=ptime:20
a=sendrecv


voip*CLI>

ACK sip:s@89.28.59.10:56504 SIP/2.0
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bK6a83.16f68c87.2
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=7F3E11ED-B95
To: ;tag=as3a3512b1
Date: Fri, 19 Nov 2010 12:59:58 GMT
Call-ID: BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
Max-Forwards: 5
Content-Length: 0
CSeq: 101 ACK
P-hint: Loose-Route - fixcontact,setflag6,mediaproxy



--- (11 headers 0 lines) ---
voip*CLI>

BYE sip:s@89.28.59.10:56504 SIP/2.0
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bK3a83.edefbcf4.0
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=7F3E11ED-B95
To: ;tag=as3a3512b1
Date: Fri, 19 Nov 2010 12:59:58 GMT
Call-ID: BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 5
Timestamp: 1290171605
CSeq: 102 BYE
Content-Length: 0
P-hint: Loose-Route - fixcontact,setflag6,mediaproxy



--- (13 headers 0 lines) ---
Sending to 89.28.56.226 : 5060 (NAT)
voip*CLI>

SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bK3a83.edefbcf4.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=7F3E11ED-B95
To: ;tag=as3a3512b1
Call-ID: BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



Scheduling destruction of SIP dialog '55a4de9b10426e6373b468dc5beef63a@10.201.24.243' in 32000 ms (Method: INVITE)
set_destination: Parsing for address/port to send to
set_destination: set destination to 10.201.24.22, port 5060
Reliably Transmitting (no NAT) to 10.201.24.22:5060:
BYE sip:5703@10.201.24.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK3d5fd3d0;rport
From: "022922105" ;tag=as4525ae3b
To: ;tag=764f886515716461i0
Call-ID: 55a4de9b10426e6373b468dc5beef63a@10.201.24.243
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
voip*CLI>

SIP/2.0 200 OK
To: ;tag=764f886515716461i0
From: "022922105" ;tag=as4525ae3b
Call-ID: 55a4de9b10426e6373b468dc5beef63a@10.201.24.243
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK3d5fd3d0
Server: Sipura/SPA941-4.1.8
Content-Length: 0



--- (8 headers 0 lines) ---
Really destroying SIP dialog '55a4de9b10426e6373b468dc5beef63a@10.201.24.243' Method: INVITE
Really destroying SIP dialog 'BF7359D5-F31311DF-95EBDC96-FEE14D3F@87.248.160.13' Method: BYE



Спасибо!
#2 21.11.2010 13:36
ни у кого мыслей нет?

жаль... у меня тоже идей нет Sad
#3 21.11.2010 14:38
поэкпериментируйте с кодеками
#4 22.11.2010 07:59
zlat wrote:
поэкпериментируйте с кодеками


в смысле?
#5 22.11.2010 08:08
tv.vldmr, не надо экспериментов - это он не подумав ляпнул.

SIP BYE приходит от прова после установления соединения. Жаль в логе нет временных меток - не видна разница между ACK со стороны * и BYE от прова.
Так или иначе, надо спрашивать прова, почему ихая сиська разрывает соединение.

P.S.: Есть подозрение, что сиська не хочет слать rtp на c=IN IP4 10.201.24.243.
Может все-таки поднапряжешься и будешь валидные ипы слать прову, а не локальные?
#6 22.11.2010 09:43
Alekz wrote:
tv.vldmr, не надо экспериментов - это он не подумав ляпнул.

SIP BYE приходит от прова после установления соединения. Жаль в логе нет временных меток - не видна разница между ACK со стороны * и BYE от прова.
Так или иначе, надо спрашивать прова, почему ихая сиська разрывает соединение.

P.S.: Есть подозрение, что сиська не хочет слать rtp на c=IN IP4 10.201.24.243.
Может все-таки поднапряжешься и будешь валидные ипы слать прову, а не локальные?


хм... странно, вроде бы выставил externip=xxx.xxx.xxx.xxx

где еще прописывать? потому как астериск у меня за НАТом по отношению к провайдеру...

Added after 11 minutes:

Хотя вру, externip не был выставлен, сейчас поставил и получаю, что

когда трубку поднимаешь, человек начинает говорит и сбивается во время разговора....
#7 22.11.2010 09:53
Во-о-о... А теперь смотрим, как работать за натом на voip-info.
#8 22.11.2010 10:07
ну проброс портов то я сделал....

пробросил порты на внутренний сервере 5060 и 10000-20000
#9 22.11.2010 10:23
Покажи сессию.
#10 22.11.2010 10:47
Alekz wrote:
Покажи сессию.


эмм в смысле? чуток не понял запроса, показать лог любого звонка ?

Added after 15 minutes:

Code:
[general]
externip=xxx.xxx.xxx.xxx
localnet=10.201.24.0/255.255.255.0
context=office
allowguest=no
allowtransfer=yes
;autocreatepeer=yes
bindport=5060
bindaddr=0.0.0.0
allowoverlap=no
srvlookup=no
g726nonstandart=yes
Language=ru
defaultexpiry=3600


на всяк случай выставил general из sip.conf и настройка сип-провайдера

Code:
type=friend
insecure = invite
secret = secret
nat = yes
username = username
fromuser = fromuser
fromdomain = sip.md
host = sip.md
port = 5060
dtmfmode = rfc2833
canreinvite = no
qualify=yes
#11 22.11.2010 10:49
tv.vldmr wrote:
эмм в смысле?
Покажи sip debug проблемного вызова.
#12 22.11.2010 10:51
Alekz wrote:
tv.vldmr wrote:
эмм в смысле?
Покажи sip debug проблемного вызова.

сейчас буду ловить.
#13 23.11.2010 11:23
не было ... не было.. и вот снова пошло Sad


Code:



--- (8 headers 0 lines) ---
Really destroying SIP dialog '60e4219d705fe92d5ad4b96f3ba5d762@10.201.24.243' Method: OPTIONS
voip*CLI>

INVITE sip:s@10.201.24.243 SIP/2.0
Record-Route:
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bKef9c.a109d1d3.0
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=933FC18B-2574
To:
Date: Tue, 23 Nov 2010 10:14:18 GMT
Call-ID: 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
Supported: timer,100rel
Min-SE: 1800
Cisco-Guid: 1142819941-4129362399-3117341846-4276178239
User-Agent: Cisco-SIPGateway/IOS-12.x
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO
CSeq: 101 INVITE
Max-Forwards: 5
Remote-Party-ID: ;party=calling;screen=yes;privacy=off
Timestamp: 1290507258
Contact:
Expires: 180
Allow-Events: telephone-event
Content-Type: application/sdp
Content-Length: 518

v=0
o=CiscoSystemsSIP-GW-UserAgent 2003 6502 IN IP4 87.248.160.13
s=SIP Call
c=IN IP4 89.28.56.228
t=0 0
m=audio 52274 RTP/AVP 8 0 18 4 98 99 2 15 102 3 101
c=IN IP4 89.28.56.228
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=yes
a=rtpmap:4 G723/8000
a=fmtp:4 annexa=yes
a=rtpmap:98 G726-16/8000
a=rtpmap:99 G726-24/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:15 G728/8000
a=rtpmap:102 GSM-EFR/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16


--- (22 headers 21 lines) ---
Sending to 89.28.56.226 : 5060 (no NAT)
Using INVITE request as basis request - 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
Found peer 'starnet1'
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 18
Found RTP audio format 4
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 2
Found RTP audio format 15
Found RTP audio format 102
Found RTP audio format 3
Found RTP audio format 101
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format G729 for ID 18
Found audio description format G723 for ID 4
Found unknown media description format G726-16 for ID 98
Found unknown media description format G726-24 for ID 99
Found audio description format G726-32 for ID 2
Found unknown media description format G728 for ID 15
Found unknown media description format GSM-EFR for ID 102
Found audio description format GSM for ID 3
Found audio description format telephone-event for ID 101
Capabilities: us - 0x8000e (gsm|ulaw|alaw|h263), peer - audio=0x90f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0xe (gsm|ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 89.28.56.228:52274
Looking for s in office (domain 10.201.24.243)
list_route: hop:


SIP/2.0 100 Trying
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bKef9c.a109d1d3.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
Record-Route:
From: ;tag=933FC18B-2574
To:
Call-ID: 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Length: 0



-- Executing [s@office:1] System("SIP/starnet1-0000201b", "echo """ - SIP/starnet1-0000201b" >> /var/log/asterisk/callinfo") in new stack
-- Executing [s@office:2] Dial("SIP/starnet1-0000201b", "SIP/5703|3600|otmw") in new stack
Audio is at 10.201.24.243 port 10708
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x2 (gsm) to SDP
Adding codec 0x8 (alaw) to SDP
Adding codec 0x10 (g726aal2) to SDP
Adding codec 0x20 (adpcm) to SDP
Adding codec 0x40 (slin) to SDP
Adding codec 0x80 (lpc10) to SDP
Adding codec 0x200 (speex) to SDP
Adding codec 0x400 (ilbc) to SDP
Adding codec 0x800 (g726) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.201.24.22:5060:
INVITE sip:5703@10.201.24.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK1db28eda;rport
From: "079288125" ;tag=as6e9dbd28
To:
Contact:
Call-ID: 004b97e35154e93024758b62057f5a6f@10.201.24.243
CSeq: 102 INVITE
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Nov 2010 10:21:04 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 498

v=0
o=root 967 967 IN IP4 10.201.24.243
s=session
c=IN IP4 10.201.24.243
t=0 0
m=audio 10708 RTP/AVP 0 3 8 112 5 10 7 110 97 111 101
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:112 AAL2-G726-32/8000
a=rtpmap:5 DVI4/8000
a=rtpmap:10 L16/8000
a=rtpmap:7 LPC/8000
a=rtpmap:110 speex/8000
a=rtpmap:97 iLBC/8000
a=fmtp:97 mode=30
a=rtpmap:111 G726-32/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv

---
-- Called 5703
Audio is at 10.201.24.243 port 17740
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bKef9c.a109d1d3.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
Record-Route:
From: ;tag=933FC18B-2574
To: ;tag=as21396d09
Call-ID: 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 967 967 IN IP4 10.201.24.243
s=session
c=IN IP4 10.201.24.243
t=0 0
m=audio 17740 RTP/AVP 3 0 8 101
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 - - - -
a=ptime:20
a=sendrecv


-- Started music on hold, class 'default', on SIP/starnet1-0000201b
voip*CLI>

SIP/2.0 100 Trying
To:
From: "079288125" ;tag=as6e9dbd28
Call-ID: 004b97e35154e93024758b62057f5a6f@10.201.24.243
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK1db28eda
Server: Sipura/SPA941-4.1.8
Content-Length: 0



--- (8 headers 0 lines) ---
voip*CLI>

SIP/2.0 180 Ringing
To: ;tag=afeaa5e8397da51i0
From: "079288125" ;tag=as6e9dbd28
Call-ID: 004b97e35154e93024758b62057f5a6f@10.201.24.243
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK1db28eda
Server: Sipura/SPA941-4.1.8
Content-Length: 0



--- (8 headers 0 lines) ---
-- SIP/5703-0000201c is ringing
voip*CLI>



voip*CLI>

SIP/2.0 200 OK
To: ;tag=afeaa5e8397da51i0
From: "079288125" ;tag=as6e9dbd28
Call-ID: 004b97e35154e93024758b62057f5a6f@10.201.24.243
CSeq: 102 INVITE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK1db28eda
Contact: "Call-Center"
Server: Sipura/SPA941-4.1.8
Content-Length: 210
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Content-Type: application/sdp
voip*CLI>
v=0
o=- 42161306 42161306 IN IP4 10.201.24.22
s=-
c=IN IP4 10.201.24.22
t=0 0
m=audio 16462 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv


--- (11 headers 11 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - 0x3f1fff (g723|gsm|ulaw|alaw|g726|adpcm|slin|lpc10|g729|speex|ilbc|g726aal2|g722|jpeg|png|h261|h263|h263p|h264), peer - audio=0x4 (ulaw)/video=0x0 (nothing), combined - 0x4 (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event)
Peer audio RTP is at port 10.201.24.22:16462
list_route: hop:
set_destination: Parsing for address/port to send to
set_destination: set destination to 10.201.24.22, port 5060
Transmitting (no NAT) to 10.201.24.22:5060:
ACK sip:5703@10.201.24.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK763a7840;rport
From: "079288125" ;tag=as6e9dbd28
To: ;tag=afeaa5e8397da51i0
Contact:
Call-ID: 004b97e35154e93024758b62057f5a6f@10.201.24.243
CSeq: 102 ACK
User-Agent: Asterisk PBX
Max-Forwards: 70
Content-Length: 0


---
-- SIP/5703-0000201c answered SIP/starnet1-0000201b
-- Stopped music on hold on SIP/starnet1-0000201b
Audio is at 10.201.24.243 port 17740
Adding codec 0x2 (gsm) to SDP
Adding codec 0x4 (ulaw) to SDP
Adding codec 0x8 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP


SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bKef9c.a109d1d3.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
Record-Route:
From: ;tag=933FC18B-2574
To: ;tag=as21396d09
Call-ID: 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
CSeq: 101 INVITE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact:
Content-Type: application/sdp
Content-Length: 285

v=0
o=root 967 968 IN IP4 10.201.24.243
s=session
c=IN IP4 10.201.24.243
t=0 0
m=audio 17740 RTP/AVP 3 0 8 101
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 - - - -
a=ptime:20
a=sendrecv


voip*CLI>

ACK sip:s@89.28.59.10:56504 SIP/2.0
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bKef9c.a109d1d3.2
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=933FC18B-2574
To: ;tag=as21396d09
Date: Tue, 23 Nov 2010 10:14:18 GMT
Call-ID: 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
Max-Forwards: 5
Content-Length: 0
CSeq: 101 ACK
P-hint: Loose-Route - fixcontact,setflag6,mediaproxy



--- (11 headers 0 lines) ---
voip*CLI>

BYE sip:s@89.28.59.10:56504 SIP/2.0
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bKbf9c.1be6973.0
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=933FC18B-2574
To: ;tag=as21396d09
Date: Tue, 23 Nov 2010 10:14:18 GMT
Call-ID: 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
User-Agent: Cisco-SIPGateway/IOS-12.x
Max-Forwards: 5
Timestamp: 1290507262
CSeq: 102 BYE
Content-Length: 0
P-hint: Loose-Route - fixcontact,setflag6,mediaproxy



--- (13 headers 0 lines) ---
Sending to 89.28.56.226 : 5060 (NAT)
voip*CLI>

SIP/2.0 200 OK
Via: SIP/2.0/UDP 89.28.56.226;branch=z9hG4bKbf9c.1be6973.0;received=89.28.56.226
Via: SIP/2.0/UDP 87.248.160.13:5060
From: ;tag=933FC18B-2574
To: ;tag=as21396d09
Call-ID: 441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13
CSeq: 102 BYE
User-Agent: Asterisk PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0



Scheduling destruction of SIP dialog '004b97e35154e93024758b62057f5a6f@10.201.24.243' in 32000 ms (Method: INVITE)
set_destination: Parsing for address/port to send to
set_destination: set destination to 10.201.24.22, port 5060
Reliably Transmitting (no NAT) to 10.201.24.22:5060:
BYE sip:5703@10.201.24.22:5060 SIP/2.0
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK2ef299db;rport
From: "079288125" ;tag=as6e9dbd28
To: ;tag=afeaa5e8397da51i0
Call-ID: 004b97e35154e93024758b62057f5a6f@10.201.24.243
CSeq: 103 BYE
User-Agent: Asterisk PBX
Max-Forwards: 70
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---
== Spawn extension (office, s, 2) exited non-zero on 'SIP/starnet1-0000201b'
voip*CLI>

SIP/2.0 200 OK
To: ;tag=afeaa5e8397da51i0
From: "079288125" ;tag=as6e9dbd28
Call-ID: 004b97e35154e93024758b62057f5a6f@10.201.24.243
CSeq: 103 BYE
Via: SIP/2.0/UDP 10.201.24.243:5060;branch=z9hG4bK2ef299db
Server: Sipura/SPA941-4.1.8
Content-Length: 0



--- (8 headers 0 lines) ---
Really destroying SIP dialog '004b97e35154e93024758b62057f5a6f@10.201.24.243' Method: INVITE
Really destroying SIP dialog '441FE176-F62111DF-B9D1DC96-FEE14D3F@87.248.160.13' Method: BYE
voip*CLI>
Disconnected from Asterisk server



На текущий момент sip.conf такой:

Code:
[general]
;externip=xx.xx.xx.xx
;localnet=10.201.24.0/255.255.255.0
context=office
allowguest=no
allowtransfer=yes
;autocreatepeer=yes
bindport=5060
bindaddr=0.0.0.0
allowoverlap=no
srvlookup=no
g726nonstandart=yes
Language=ru
defaultexpiry=3600
register = username:secret@sip.md

[starnet1]
type=friend
externip=xx.xx.xx.xx
localnet=10.201.24.0/24
insecure = invite
secret = secret
nat = yes
username = username
fromuser = user
fromdomain = sip.md
host = sip.md
port = 5060
dtmfmode = rfc2833
canreinvite = no
qualify=yes
#14 23.11.2010 13:27
Ку.

Для начала в [starnet1] :

disallow=all
allow=g729,alaw,ulaw

чтоб отсеять всю ботву, типа GSM кодека, который той стороной шлется в SDP

и что делает:
externip=xx.xx.xx.xx
localnet=10.201.24.0/24

в настройках пира? Это должно жить в секции [general]
_________________
ys
http://voip.rus.net/
#15 23.11.2010 13:35
ys прав. Потому и SDP без изменений:
Quote:
c=IN IP4 10.201.24.243
#16 23.11.2010 15:57
ys wrote:
Ку.

Для начала в [starnet1] :

disallow=all
allow=g729,alaw,ulaw

чтоб отсеять всю ботву, типа GSM кодека, который той стороной шлется в SDP

и что делает:
externip=xx.xx.xx.xx
localnet=10.201.24.0/24

в настройках пира? Это должно жить в секции [general]


по поводу что делают в настройках пира...посоветовали здеся Sad а я поверил)

поправлю Wink

Added after 12 minutes:

изменил и сразу заметил вот такую ошибку в логах

Code:
== Spawn extension (office, s, 2) exited non-zero on 'SIP/starnet1-0000207b'
Really destroying SIP dialog '6d5220490fed143f1f6e9c69280e321d@10.201.24.243' Method: BYE
[Nov 23 14:55:40] WARNING[967]: chan_sip.c:2017 retrans_pkt: Maximum retries exceeded on transmission D32CC973-F63611DF-A549B09D-35E175E@89.28.25.114 for seqno 101 (Critical Response) -- See doc/sip-retransmit.txt.
Really destroying SIP dialog 'D32CC973-F63611DF-A549B09D-35E175E@89.28.25.114' Method: CANCEL
Reliably Transmitting (NAT) to 89.28.56.226:5060:
OPTIONS sip:sip.md SIP/2.0
Via: SIP/2.0/UDP xx.xxx.xxx.xxx:5060;branch=z9hG4bK6bc00f5e;rport
From: "Unknown" ;tag=as497c1ec1
To:
Contact:
Call-ID: 3f43af4937a05a2a57a32ceb30fe87b5@xx.xxx.xxx.xxx
CSeq: 102 OPTIONS
User-Agent: Asterisk PBX
Max-Forwards: 70
Date: Tue, 23 Nov 2010 12:55:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Length: 0


Не понятно только почему From Unknown ....

Added after 12 minutes:

20 секунд и разрывается....

Added after 1 hours 50 minutes:

мне тут сказали, что за НАТом работать полюбому не будет нормально....

поэтому зря старался Sad
#17 24.11.2010 07:46
tv.vldmr wrote:
мне тут сказали, что за НАТом работать полюбому не будет нормально....
То-то у всех работает =) Или думаешь врут?

Я уже давал ссылку на voip-info.org - там расписано все. По твоей проблеме - у тебя ACK от прова приходит не на тот порт. Что-то мне кажется, нифига ты не прокинул порты до *.
#18 24.11.2010 10:33
Alekz wrote:
tv.vldmr wrote:
мне тут сказали, что за НАТом работать полюбому не будет нормально....
То-то у всех работает =) Или думаешь врут?

Я уже давал ссылку на voip-info.org - там расписано все. По твоей проблеме - у тебя ACK от прова приходит не на тот порт. Что-то мне кажется, нифига ты не прокинул порты до *.

сделал прямой редирект, просто редирект делал с sip.md на свой ... а там смотрю есть прокси Opensys или как там его еще...

тоже думаю, что косяк в правилах фаера.

можешь привести пример своих правил? Чтобы я сверился...