Askozia. Why is the second outgoing call not going through?

Question

Asterisk

Good day, colleagues!
There is a server with Askozia and about 15 phones.
Faced the problem that if one phone makes an outgoing call, then the second phone can no longer get through. The problem is observed only with external calls, there are no problems with internal calls.

Log server-phone

--- (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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK78e7770b;rport
Max-Forwards: 70
From: "102" <sip:[email protected]>;tag=as192ef781
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
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:[email protected]>;tag=as192ef781
To: <sip:[email protected]:5060>;tag=as2c3458f2
Call-ID: [email protected]
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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK78e7770b;rport
Max-Forwards: 70
From: "102" <sip:[email protected]>;tag=as192ef781
To: <sip:[email protected]:5060>;tag=as2c3458f2
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK437a85b7;rport
Max-Forwards: 70
From: "102" <sip:[email protected]>;tag=as192ef781
To: <sip:[email protected]:5060>
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 INVITE
User-Agent: AskoziaPBX
Authorization: Digest username="LS6636186891", realm="VM-PBX-PUBLIC-1", algorithm=MD5, uri="sip:[email protected]: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:[email protected]>;tag=as192ef781
To: <sip:[email protected]:5060>
Call-ID: [email protected]
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:[email protected]: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:[email protected]>;tag=as192ef781
To: <sip:[email protected]:5060>;tag=as6a8d9103
Call-ID: [email protected]
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:[email protected]:5060 SIP/2.0
Via: SIP/2.0/UDP 79.111.156.4:5060;branch=z9hG4bK437a85b7;rport
Max-Forwards: 70
From: "102" <sip:[email protected]>;tag=as192ef781
To: <sip:[email protected]:5060>;tag=as6a8d9103
Contact: <sip:[email protected]:5060>
Call-ID: [email protected]
CSeq: 103 ACK
User-Agent: AskoziaPBX
Content-Length: 0

Answer 1

Sori, does the provider exactly support multi-channel? If the provider limits you to one connection, then you need to agree with him)))