Имеется Asterisk v11 (без GUI) и Panasonic KX-NS500. Друг другу звонят. Стык организован по SIP.
Проблема заключается в том, что когда с asterisk-а звонят на panasonic не отображается display name, а номер отображается. В обратную сторону все нормально.
При трассировке исходящего вызова с asterisk-а в дампе видно, что display name с asterisk-а отправляется.
Человек, который настраивал asterisk утверждает, что display name у него в формате utf-8
Со своей стороны ранее я стыковал этот же panasonic с FreePBX 13. Там таких проблем не было.
Вот дамп вызова с asterisk-а на panasonic:
- Код: Выделить всё
-- Executing [2555@DID_avaya:2] NoOp("SIP/avaya_procr-00000f18", "Result: Bobrov D.V.") in new stack
-- Executing [2555@DID_avaya:3] Set("SIP/avaya_procr-00000f18", "CALLERID(name)=Bobrov D.V.") in new stack
-- Executing [2555@DID_avaya:4] Set("SIP/avaya_procr-00000f18", "CALLERID(all)=Bobrov D.V. <1236>") in new stack
-- Executing [2555@DID_avaya:5] Macro("SIP/avaya_procr-00000f18", "trunkdial,SIP/tyva/2555,SIP/tyva/2555") in new stack
-- Executing [s@macro-trunkdial:1] NoOp("SIP/avaya_procr-00000f18", "arg1 = SIP/tyva/2555 ") in new stack
-- Executing [s@macro-trunkdial:2] NoOp("SIP/avaya_procr-00000f18", "arg2 = SIP/tyva/2555 ") in new stack
-- Executing [s@macro-trunkdial:3] Dial("SIP/avaya_procr-00000f18", "SIP/tyva/2555") in new stack
== Using SIP RTP CoS mark 5
Audio is at 19706
Adding codec 100004 (alaw) to SDP
Adding codec 100003 (ulaw) to SDP
Adding codec 100008 (g729) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 10.117.33.2:35060:
INVITE sip:2555@10.117.33.2 SIP/2.0
Via: SIP/2.0/UDP 172.31.43.20:5060;branch=z9hG4bK227f22d9
Max-Forwards: 70
From: "Bobrov D.V." <sip:1236@172.31.43.20>;tag=as60cd0154
To: <sip:2555@10.117.33.2>
Contact: <sip:1236@172.31.43.20:5060>
Call-ID: 1f2b219b7078bba73acbcde9738a7a6a@172.31.43.20:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 11.25.3
Date: Wed, 23 Jan 2019 08:15:02 GMT
Session-Expires: 1800
Min-SE: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
P-Asserted-Identity: "Bobrov D.V." <sip:1236@172.31.43.20>
Content-Type: application/sdp
Content-Length: 305
v=0
o=root 487893661 487893661 IN IP4 172.31.43.20
s=Asterisk PBX 11.25.3
c=IN IP4 172.31.43.20
t=0 0
m=audio 19706 RTP/AVP 8 0 18 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
---
-- Called SIP/tyva/2555
<--- SIP read from UDP:10.117.33.2:35060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 172.31.43.20:5060;branch=z9hG4bK227f22d9;received=172.31.43.20
Call-ID: 1f2b219b7078bba73acbcde9738a7a6a@172.31.43.20:5060
From: "Bobrov D.V." <sip:1236@172.31.43.20>;tag=as60cd0154
To: <sip:2555@10.117.33.2>
CSeq: 102 INVITE
Content-Length: 0
<------------->
--- (7 headers 0 lines) ---
<--- SIP read from UDP:10.117.33.2:35060 --->
SIP/2.0 502 Bad Gateway
Via: SIP/2.0/UDP 172.31.43.20:5060;branch=z9hG4bK227f22d9;received=172.31.43.20
Call-ID: 1f2b219b7078bba73acbcde9738a7a6a@172.31.43.20:5060
From: "Bobrov D.V." <sip:1236@172.31.43.20>;tag=as60cd0154
To: <sip:2555@10.117.33.2>;tag=468963829
CSeq: 102 INVITE
Allow: INVITE,ACK,CANCEL,BYE,PRACK,INFO,UPDATE,OPTIONS,REGISTER,NOTIFY
Server: Panasonic-MPR15-V006.01079/VSIPGW-V3.0000
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
-- Got SIP response 502 "Bad Gateway" back from 10.117.33.2:35060
Transmitting (no NAT) to 10.117.33.2:35060:
ACK sip:2555@10.117.33.2 SIP/2.0
Via: SIP/2.0/UDP 172.31.43.20:5060;branch=z9hG4bK227f22d9
Max-Forwards: 70
From: "Bobrov D.V." <sip:1236@172.31.43.20>;tag=as60cd0154
To: <sip:2555@10.117.33.2>;tag=468963829
Contact: <sip:1236@172.31.43.20:5060>
Call-ID: 1f2b219b7078bba73acbcde9738a7a6a@172.31.43.20:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 11.25.3
Content-Length: 0