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confbridge

Проблемы Asterisk без вэб-оболочек и их решения

Модераторы: april22, Zavr2008

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Aleks Second
Сообщения: 3
Зарегистрирован: 26 окт 2012, 09:48

confbridge

Сообщение Aleks Second »

Астериск 10.9.0 с confbrige. на нем конференции собираются нормально, но как только я пытаюсь подключить к конференции участника с другого Астериска 2, то соединение устанавливается и через 1 сек отключается. при sip debug видно что сообщения идут от пользователя anonymous. при этом обычные вызовы через Астериск 2 проходят отлично.


INVITE sip:777777@192.168.1.42;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK77723ddb;rport Max-Forwards: 70 From: "anonymous" <sip:760000@anonymous.invalid>;tag=as67e67632 To: <sip:777777@192.168.1.42;user=phone> Contact: <sip:760000@192.168.1.248:5060> Call-ID: 28402dff2e2642b718c8984037c5260d@111000.usar.ru CSeq: 102 INVITE User-Agent: 11111111 Date: Wed, 24 Oct 2012 07:29:28 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 227 v=0 o=root 1281397134 1281397134 IN IP4 192.168.1.248 s=111111111 c=IN IP4 192.168.1.248 t=0 0 m=audio 17872 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv
---

[Khosting*CLI>
[0K
<--- SIP read from UDP:192.168.1.42:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK77723ddb;received=192.168.1.248;rport=5060
From: "anonymous" <sip:760000@anonymous.invalid>;tag=as67e67632
To: <sip:777777@192.168.1.42;user=phone>
Call-ID: 28402dff2e2642b718c8984037c5260d@111000.usar.ru
CSeq: 102 INVITE
User-Agent: Softswitch Elcom SIPlant
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:777777@192.168.1.42>
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---

[Khosting*CLI>
[0K
<--- SIP read from UDP:192.168.1.24:5060 --->
jaK
<------------->

[Khosting*CLI>
[0K
<--- SIP read from UDP:192.168.1.42:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK77723ddb;received=192.168.1.248;rport=5060
From: "anonymous" <sip:760000@anonymous.invalid>;tag=as67e67632
To: <sip:777777@192.168.1.42;user=phone>;tag=as7305314a
Call-ID: 28402dff2e2642b718c8984037c5260d@111000.usar.ru
CSeq: 102 INVITE
User-Agent: Softswitch Elcom SIPlant
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:777777@192.168.1.42>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1736 1736 IN IP4 192.168.1.42
s=session
c=IN IP4 192.168.1.42
t=0 0
m=audio 15372 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---

[Khosting*CLI>
[0Klist_route: hop: <sip:777777@192.168.1.42>

[Khosting*CLI>
[0KFound RTP audio format 8
Found RTP audio format 101

[Khosting*CLI>
[0KFound audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.42:15372

[Khosting*CLI>
[0K
<--- SIP read from UDP:192.168.1.60:5060 --->


<------------->

[Khosting*CLI>
[0K
<--- SIP read from UDP:192.168.1.42:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK77723ddb;received=192.168.1.248;rport=5060
From: "anonymous" <sip:760000@anonymous.invalid>;tag=as67e67632
To: <sip:777777@192.168.1.42;user=phone>;tag=as7305314a
Call-ID: 28402dff2e2642b718c8984037c5260d@111000.usar.ru
CSeq: 102 INVITE
User-Agent: Softswitch Elcom SIPlant
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:777777@192.168.1.42>
Content-Type: application/sdp
Content-Length: 238

v=0
o=root 1736 1737 IN IP4 192.168.1.42
s=session
c=IN IP4 192.168.1.42
t=0 0
m=audio 15372 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
<------------->
--- (12 headers 12 lines) ---

[Khosting*CLI>
[0KFound RTP audio format 8
Found RTP audio format 101

[Khosting*CLI>
[0KFound audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.1.42:15372
list_route: hop: <sip:777777@192.168.1.42>
set_destination: Parsing <sip:777777@192.168.1.42> for address/port to send to
set_destination: set destination to 192.168.1.42:5060
Transmitting (NAT) to 192.168.1.42:5060:
ACK sip:777777@192.168.1.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK691637a3;rport Max-Forwards: 70 From: "anonymous" <sip:760000@anonymous.invalid>;tag=as67e67632 To: <sip:777777@192.168.1.42;user=phone>;tag=as7305314a Contact: <sip:760000@192.168.1.248:5060> Call-ID: 28402dff2e2642b718c8984037c5260d@111000.usar.ru CSeq: 102 ACK User-Agent: 11111111 Content-Length: 0
---

[Khosting*CLI>
[0K -- Auto fallthrough, channel 'SIP/freebsd-0000005f' status is 'UNKNOWN'

[Khosting*CLI>
[0KScheduling destruction of SIP dialog '28402dff2e2642b718c8984037c5260d@111000.usar.ru' in 32000 ms (Method: INVITE)

[Khosting*CLI>
[0Kset_destination: Parsing <sip:777777@192.168.1.42> for address/port to send to

[Khosting*CLI>
[0Kset_destination: set destination to 192.168.1.42:5060

[Khosting*CLI>
[0KReliably Transmitting (NAT) to 192.168.1.42:5060:
BYE sip:777777@192.168.1.42 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK4bcecddc;rport Max-Forwards: 70 From: "anonymous" <sip:760000@anonymous.invalid>;tag=as67e67632 To: <sip:777777@192.168.1.42;user=phone>;tag=as7305314a Call-ID: 28402dff2e2642b718c8984037c5260d@111000.usar.ru CSeq: 103 BYE User-Agent: 11111111 X-Asterisk-HangupCause: Normal Clearing X-Asterisk-HangupCauseCode: 16 Content-Length: 0
---

[Khosting*CLI>
[0K -- Auto fallthrough, channel 'SIP/100-0000005b' status is 'UNKNOWN'

<--- SIP read from UDP:192.168.1.42:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.248:5060;branch=z9hG4bK4bcecddc;received=192.168.1.248;rport=5060
From: "anonymous" <sip:760000@anonymous.invalid>;tag=as67e67632
To: <sip:777777@192.168.1.42;user=phone>;tag=as7305314a
Call-ID: 28402dff2e2642b718c8984037c5260d@111000.usar.ru
CSeq: 103 BYE
User-Agent: Softswitch Elcom SIPlant
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Supported: replaces
Contact: <sip:777777@192.168.1.42>
Content-Length: 0



если вызов совершить по обычному (не в режиме конференции), то все ок.

sip.conf
[freebsd]
secret=12345
fromuser=760000
type=friend
host=dynamic ; This device needs to register
dtmfmode=rfc2833
context=from-gsm
trunk=yes
disallow=all
allow=alaw
insecure=invite,port

extension.conf


exten => 101,1,Originate(SIP/101,exten,conferencia,902,1)
;exten => 101,2,Hangup
exten => 92,1,Originate(SIP/92,exten,conferencia,902,1)
exten => 92,2,Hangup
exten => 50,1,Originate(SIP/50,exten,conferencia,902,1)
;exten => 50,2,Hangup
exten => 92,1,Dial(SIP/92,60,tT)
exten => 92,2,Hangup

;exten => _XXXXXX,1,Set(CALLERID(name)=1721617216)
exten => _XXXXXX,1,Originate(SIP/${EXTEN}@freebsd,exten,conferencia,902,2)
exten => _XXXXXX,2,Hangup

;exten => 3000,1,Dial(SIP/${EXTEN}@cisco,100,tT))
;exten => 3000,2,Hangup

[conferencia]

exten => 902,1,Confbridge(902,conf_test,user_test,menu_test)
;exten => 902,n,Hangup

exten => 903,1,Set(CONFBRIDGE_JOIN_SOUND=conf-join).
exten => 903,n,ConfBridge(${BLINDTRANSFER:4:3},M1)

confbridge.conf





[conf_test].
type=bridge.
;max_members=20.
max_members=20.
record_conference=no.
internal_sample_rate=auto.
mixing_interval=20.
sound_join= beep.
;sound=leave= beeperr.
sound_leave= beeperr.
pin=456.

[user_test].
type=user.
admin=yes.
marked=no.
startmuted=no.
music_on_hold_when_empty=yes.
;denoise=yes ;need codec_speex.
denoise=no.
;announce_user_count=yes
;announce_user_count_all=yes

[menu_test].
type= menu.
*=playback_and_continue(press&digits/1&press&digits/2&press&digits/3&press&digits/4&press&digits/5&press&digits/6&press&digits/7&press&digits/8press&digits/9&
1=toggle_mute.
2=leave_conference.
3=dialplan_exec(invitar,1234,1).
4=decrease_listening_volume.
5=reset_listening_volume.
6=increase_listening_volume.
7=decrease_talking_volume.
8=reset_talking_volume.
9=increase_talking_volume.
0=no_op


почему могут сбрасываться соединения
Вернуться к началу
ded
Сообщения: 15630
Зарегистрирован: 26 авг 2010, 19:00

Re: confbridge

Сообщение ded »

The quote of Changelog of 1.8:
2010-03-31 22:35 +0000 [r255701] Mark Michelson <mmichelson@digium.com>

* channels/chan_sip.c: Fix improper comaparison of anonymous URI
when getting P-Asserted-Identity. There was a bug where we split
the URI on the @ sign and then attempted to compare to
"anonymous@anonymous.invalid" afterwards. This comparison could
never evaluate true. So now we keep a copy of the URI prior to
the split so that the comparison is valid.
chan_sip.c:

Код: Выделить всё

/*!
 * \pre if p->owner exists, it must be locked
 * \brief Add Remote-Party-ID header to SIP message
 */
static int add_rpid(struct sip_request *req, struct sip_pvt *p)
{
        struct ast_str *tmp = ast_str_alloca(256);
        char tmp2[256];
        char *lid_num = NULL;
        char *lid_name = NULL;
        int lid_pres;
        const char *fromdomain;
        const char *privacy = NULL;
        const char *screen = NULL;
        const char *anonymous_string = "\"Anonymous\" <sip:anonymous@anonymous.invalid>";
Вернуться к началу
Aleks Second
Сообщения: 3
Зарегистрирован: 26 окт 2012, 09:48

Re: confbridge

Сообщение Aleks Second »

вы бы не могли поподробней рассказать, что надо сделать.
Вернуться к началу
Aleks Second
Сообщения: 3
Зарегистрирован: 26 окт 2012, 09:48

Re: confbridge

Сообщение Aleks Second »

разобрался, заработало
Вернуться к началу
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