Askozia. Почему не проходит второй исходящий звонок?

Question

[neonox]

Доброго времени суток, коллеги!

Есть сервер с Askozia и порядка 15 телефонов.
Столкнулся с проблемой, что если один телефон совершает исходящий звонок, то со второго телефона уже не дозвониться. Проблема наблюдается только с внешними звонками, при внутренних звонках проблем нет.

Лог сервер-телефон

--- (12 headers 0 lines) ---
Audio is at 10108
Adding codec 100003 (ulaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 109.94.93.125:5060:
INVITE sip:89065552269@109.94.93.125:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK78e7770b;rport
Max-Forwards: 70
From: "102" <sip:LS6636186891@109.94.93.125>;tag=as192ef781
To: <sip:89065552269@109.94.93.125:5060>
Contact: <sip:LS6636186891@79.111.156.4:5060>
Call-ID: 58ed98ff2a2c77ed1175286b71e1c050@109.94.93.125
CSeq: 102 INVITE
User-Agent: AskoziaPBX
Date: Thu, 06 Aug 2015 07:27:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 1707180845 1707180845 IN IP4 79.111.156.4
s=Asterisk PBX 10.9.0
c=IN IP4 79.111.156.4
t=0 0
m=audio 10108 RTP/AVP 0 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:109.94.93.125:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK78e7770b;received=192.168.1.25;rport=5060
From: "102" <sip:LS6636186891@109.94.93.125>;tag=as192ef781
To: <sip:89065552269@109.94.93.125:5060>;tag=as2c3458f2
Call-ID: 58ed98ff2a2c77ed1175286b71e1c050@109.94.93.125
CSeq: 102 INVITE
Server: MERA MVTS v.9.3.0-17c
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="VM-PBX-PUBLIC-1", nonce="75a11cc2"
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Transmitting (NAT) to 109.94.93.125:5060:
ACK sip:89065552269@109.94.93.125:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK78e7770b;rport
Max-Forwards: 70
From: "102" <sip:LS6636186891@109.94.93.125>;tag=as192ef781
To: <sip:89065552269@109.94.93.125:5060>;tag=as2c3458f2
Contact: <sip:LS6636186891@79.111.156.4:5060>
Call-ID: 58ed98ff2a2c77ed1175286b71e1c050@109.94.93.125
CSeq: 102 ACK
User-Agent: AskoziaPBX
Content-Length: 0


---
Audio is at 10108
Adding codec 100003 (ulaw) to SDP
Adding codec 100012 (g722) to SDP
Adding codec 100004 (alaw) to SDP
Adding codec 100002 (gsm) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 109.94.93.125:5060:
INVITE sip:89065552269@109.94.93.125:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK437a85b7;rport
Max-Forwards: 70
From: "102" <sip:LS6636186891@109.94.93.125>;tag=as192ef781
To: <sip:89065552269@109.94.93.125:5060>
Contact: <sip:LS6636186891@79.111.156.4:5060>
Call-ID: 58ed98ff2a2c77ed1175286b71e1c050@109.94.93.125
CSeq: 103 INVITE
User-Agent: AskoziaPBX
Authorization: Digest username="LS6636186891", realm="VM-PBX-PUBLIC-1", algorithm=MD5, uri="sip:89065552269@109.94.93.125:5060", nonce="75a11cc2", response="84c805102f425dfff6f3dc13d96285e7"
Date: Thu, 06 Aug 2015 07:27:19 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 306

v=0
o=root 1707180845 1707180846 IN IP4 79.111.156.4
s=Asterisk PBX 10.9.0
c=IN IP4 79.111.156.4
t=0 0
m=audio 10108 RTP/AVP 0 9 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:109.94.93.125:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK437a85b7;received=192.168.1.25;rport=5060
From: "102" <sip:LS6636186891@109.94.93.125>;tag=as192ef781
To: <sip:89065552269@109.94.93.125:5060>
Call-ID: 58ed98ff2a2c77ed1175286b71e1c050@109.94.93.125
CSeq: 103 INVITE
Server: MERA MVTS v.9.3.0-17c
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Session-Expires: 1800;refresher=uas
Contact: <sip:89065552269@109.94.93.125:5060>
Content-Length: 0

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

<--- SIP read from UDP:109.94.93.125:5060 --->
SIP/2.0 603 Declined
Via: SIP/2.0/UDP 192.168.1.25:5060;branch=z9hG4bK437a85b7;received=192.168.1.25;rport=5060
From: "102" <sip:LS6636186891@109.94.93.125>;tag=as192ef781
To: <sip:89065552269@109.94.93.125:5060>;tag=as6a8d9103
Call-ID: 58ed98ff2a2c77ed1175286b71e1c050@109.94.93.125
CSeq: 103 INVITE
Server: MERA MVTS v.9.3.0-17c
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Transmitting (NAT) to 109.94.93.125:5060:
ACK sip:89065552269@109.94.93.125:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK437a85b7;rport
Max-Forwards: 70
From: "102" <sip:LS6636186891@109.94.93.125>;tag=as192ef781
To: <sip:89065552269@109.94.93.125:5060>;tag=as6a8d9103
Contact: <sip:LS6636186891@79.111.156.4:5060>
Call-ID: 58ed98ff2a2c77ed1175286b71e1c050@109.94.93.125
CSeq: 103 ACK
User-Agent: AskoziaPBX
Content-Length: 0

Comments

[Владимир]

ваш дебаг - просто мусор
зачем снимать дебаг на внутреннем пире если не проходит на внешний?

[neonox]

Владимир: да, Вы правы. Поменял. По дебагу идет SIP/2.0 603 Decline. Буду пробовать мучать поставщика услуги.

[mureevms]

Так у Вас сколько транков подключено? Не один ли?

Answer 1

[Vl@d1m1r]

Сори , а провайдер точно поддерживает многоканальность? Если провайдер вас ограничивает одним соединением то нужно с ним договориться )))