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
