peer не даёт работать user`у
Добавлено: 30 окт 2014, 17:35
Есть номер телефонный 28ХХХХХ.
Выходит ошибка Out of capacity
Тип соединения sip account.
Если поставить user only, входящие звонки работают.
Если поставить peer, исходящие работают, входящие нет.
Если поставить friend, то исходящие работают, а входящие нет.
-- Called SIP/28XXXXX_peer/28XXXXX
Выходит ошибка Out of capacity
Тип соединения sip account.
Если поставить user only, входящие звонки работают.
Если поставить peer, исходящие работают, входящие нет.
Если поставить friend, то исходящие работают, а входящие нет.
-- Called SIP/28XXXXX_peer/28XXXXX
Код: Выделить всё
<--- SIP read from UDP:195.64.217.115:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.45.1.13:5060;branch=z9hG4bK64efd16c;rport=5060;received=92.242.6.110
From: <sip:28XXXXX@sip6.ural.net>;tag=as04ef2cda
To: <sip:28XXXXX@sip6.ural.net>
Call-ID: 53f7f2b4199f3c204b9e32c03874b13d@sip6.ural.net
CSeq: 102 INVITE
Contact: <sip:28XXXXX@195.64.217.115:5060>
Server: TS-v4.5.1-17W
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from UDP:195.64.217.115:5060 --->
SIP/2.0 603 Decline
Via: SIP/2.0/UDP 10.45.1.13:5060;branch=z9hG4bK64efd16c;rport=5060;received=92.242.6.110
From: <sip:28XXXXX@sip6.ural.net>;tag=as04ef2cda
To: <sip:28XXXXX@sip6.ural.net>;tag=547663675-3826336096-620815540-3932369040
Call-ID: 53f7f2b4199f3c204b9e32c03874b13d@sip6.ural.net
CSeq: 102 INVITE
Contact: <sip:28XXXXX@195.64.217.115:5060>
Server: TS-v4.5.1-17W
Reason: Centrex;cause=181;text="Out of capacity"
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
-- Got SIP response 603 "Decline" back from 195.64.217.115:5060
Transmitting (NAT) to 195.64.217.115:5060:
ACK sip:28XXXXX@sip6.ural.net SIP/2.0
Via: SIP/2.0/UDP 10.45.1.13:5060;branch=z9hG4bK64efd16c;rport
Max-Forwards: 70
From: <sip:28XXXXX@sip6.ural.net>;tag=as04ef2cda
To: <sip:28XXXXX@sip6.ural.net>;tag=547663675-3826336096-620815540-3932369040
Contact: <sip:28XXXXX@10.45.1.13:5060>
Call-ID: 53f7f2b4199f3c204b9e32c03874b13d@sip6.ural.net
CSeq: 102 ACK
User-Agent: FPBX-12.0.1rc34(11.7.0)
Content-Length: 0
---
-- SIP/28XXXXX_peer-0000005f is busy
== Everyone is busy/congested at this time (1:1/0/0)
-- Executing [s@macro-dialout-trunk:23] NoOp("SIP/28XXXXX_peer-0000005e", "Dial failed for some reason with DIALSTATUS = BUSY and HANGUPCAUSE = 21") in new stack
-- Executing [s@macro-dialout-trunk:24] GotoIf("SIP/28XXXXX_peer-0000005e", "1?continue,1:s-BUSY,1") in new stack
-- Goto (macro-dialout-trunk,continue,1)
-- Executing [continue@macro-dialout-trunk:1] NoOp("SIP/28XXXXX_peer-0000005e", "TRUNK Dial failed due to BUSY HANGUPCAUSE: 21 - failing through to other trunks") in new stack
-- Executing [continue@macro-dialout-trunk:2] Set("SIP/28XXXXX_peer-0000005e", "CALLERID(number)=") in new stack
-- Executing [28XXXXX@from-internal:6] Macro("SIP/28XXXXX_peer-0000005e", "outisbusy,") in new stack
-- Executing [s@macro-outisbusy:1] Progress("SIP/28XXXXX_peer-0000005e", "") in new stack
Audio is at 13230
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Transmitting (NAT) to 195.64.217.115:5060 --->
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP 195.64.217.115:5060;branch=z9hG4bK-1491512635-3826336096-620815540-39323690401;received=195.64.217.115;rport=5060
Via: SIP/2.0/UDP 195.64.217.115:5061;rport=5061;branch=z9hG4bK-1491512635-3826336096-620815540-3932369040;received=195.64.217.115
From: <sip:9625576738@195.64.217.115:5061;user=phone>;tag=2530913595-3826336096-620815540-3932369040
To: <sip:28XXXXX@195.64.217.115;user=phone>;tag=as13260d5d
Call-ID: 1BD7818D263914C6FE97462877D82ECC
CSeq: 1 INVITE
Server: FPBX-12.0.1rc34(11.7.0)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Contact: <sip:28XXXXX@10.45.1.13:5060>
Content-Type: applica