COMNET PHONE 有一個很嚴重的問題..

lamsoft

-- Executing [44444444@PSTN_AutoRetryCall:5] Dial("SIP/XXXX-000000d6", "SIP/COMNET_PSTN/44444444") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/COMNET_PSTN/44444444
    -- Call on SIP/COMNET_PSTN-000000d7 placed on hold

打不通的號碼會被placed on hold!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!
垃圾COMNET PHONE, 陣間打去插佢


* 此處以 4444-4444 作測試, 實際環境如要打信用卡book飛熱線時, 會不能運作！

lamsoft

Audio is at 20000
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 202.0.179.3:5060:
INVITE sip:44444444@202.0.179.3 SIP/2.0
Via: SIP/2.0/UDP ASTERISK_SERVER_IP:5060;branch=z9hG4bK3fadc397
Max-Forwards: 70
From: "" <sip:852HK_PSTN_PHONE_NO@202.0.179.3>;tag=as3065e440
To: <sip:44444444@202.0.179.3>
Contact: <sip:852HK_PSTN_PHONE_NO@ASTERISK_SERVER_IP:5060>
Call-ID: 4c0de08d59e0d92716ac1c4d4b7e2387@202.0.179.3
CSeq: 102 INVITE
User-Agent: Linksys/SPA3102-5.1.10(GW)
Date: Fri, 16 Aug 2013 07:37:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 222

v=0
o=Linksys 156856094 156856094 IN IP4 ASTERISK_SERVER_IP
s=Linksys
c=IN IP4 ASTERISK_SERVER_IP
t=0 0
m=audio 20000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---
    -- Called SIP/COMNET_PSTN/44444444

<--- SIP read from UDP:202.0.179.3:5060 --->
SIP/2.0 100 Trying
From: "" <sip:852HK_PSTN_PHONE_NO@202.0.179.3>;tag=as3065e440
To: <sip:44444444@202.0.179.3>
CSeq: 102 INVITE
Call-ID: 4c0de08d59e0d92716ac1c4d4b7e2387@202.0.179.3
Via: SIP/2.0/UDP ASTERISK_SERVER_IP:5060;branch=z9hG4bK3fadc397;rport=5060
Content-Length: 0

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

<--- SIP read from UDP:202.0.179.3:5060 --->
SIP/2.0 407 Proxy Authentication Required
From: "" <sip:852HK_PSTN_PHONE_NO@202.0.179.3>;tag=as3065e440
To: <sip:44444444@202.0.179.3>;tag=38848e84
CSeq: 102 INVITE
Call-ID: 4c0de08d59e0d92716ac1c4d4b7e2387@202.0.179.3
Via: SIP/2.0/UDP ASTERISK_SERVER_IP:5060;branch=z9hG4bK3fadc397;rport=5060
Proxy-Authenticate: Digest realm="huawei.com",nonce="15:37:25:23905", stale=false,algorithm=MD5
Reason: Q.850;cause="0";text="unknown"
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
Transmitting (no NAT) to 202.0.179.3:5060:
ACK sip:44444444@202.0.179.3 SIP/2.0
Via: SIP/2.0/UDP ASTERISK_SERVER_IP:5060;branch=z9hG4bK3fadc397
Max-Forwards: 70
From: "" <sip:852HK_PSTN_PHONE_NO@202.0.179.3>;tag=as3065e440
To: <sip:44444444@202.0.179.3>;tag=38848e84
Contact: <sip:852HK_PSTN_PHONE_NO@ASTERISK_SERVER_IP:5060>
Call-ID: 4c0de08d59e0d92716ac1c4d4b7e2387@202.0.179.3
CSeq: 102 ACK
User-Agent: Linksys/SPA3102-5.1.10(GW)
Content-Length: 0


---
Audio is at 20000
Adding codec 100004 (alaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 202.0.179.3:5060:
INVITE sip:44444444@202.0.179.3 SIP/2.0
Via: SIP/2.0/UDP ASTERISK_SERVER_IP:5060;branch=z9hG4bK2f4e3093
Max-Forwards: 70
From: "" <sip:852HK_PSTN_PHONE_NO@202.0.179.3>;tag=as3065e440
To: <sip:44444444@202.0.179.3>
Contact: <sip:852HK_PSTN_PHONE_NO@ASTERISK_SERVER_IP:5060>
Call-ID: 4c0de08d59e0d92716ac1c4d4b7e2387@202.0.179.3
CSeq: 103 INVITE
User-Agent: Linksys/SPA3102-5.1.10(GW)
Proxy-Authorization: Digest username="852HK_PSTN_PHONE_NO", realm="huawei.com", algorithm=MD5, uri="sip:44444444@202.0.179.3", nonce="15:37:25:23905", response="AUTH_RESPONSE"
Date: Fri, 16 Aug 2013 07:37:24 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 222

v=0
o=Linksys 156856094 156856095 IN IP4 ASTERISK_SERVER_IP
s=Linksys
c=IN IP4 ASTERISK_SERVER_IP
t=0 0
m=audio 20000 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

---

<--- SIP read from UDP:202.0.179.3:5060 --->
SIP/2.0 100 Trying
From: "" <sip:852HK_PSTN_PHONE_NO@202.0.179.3>;tag=as3065e440
To: <sip:44444444@202.0.179.3>
CSeq: 103 INVITE
Call-ID: 4c0de08d59e0d92716ac1c4d4b7e2387@202.0.179.3
Via: SIP/2.0/UDP ASTERISK_SERVER_IP:5060;branch=z9hG4bK2f4e3093;rport=5060
Content-Length: 0

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

<--- SIP read from UDP:202.0.179.3:5060 --->
SIP/2.0 183 Session Progress
From: "" <sip:852HK_PSTN_PHONE_NO@202.0.179.3>;tag=as3065e440
To: <sip:44444444@202.0.179.3>;tag=006585d6
CSeq: 103 INVITE
Call-ID: 4c0de08d59e0d92716ac1c4d4b7e2387@202.0.179.3
Via: SIP/2.0/UDP ASTERISK_SERVER_IP:5060;branch=z9hG4bK2f4e3093;rport=5060
Contact: <sip:44444444@202.0.179.3:5060;user=phone>
Content-Length: 210
Content-Type: application/sdp

v=0
o=HuaweiSoftX3000 6684583 6684583 IN IP4 10.0.1.36
s=Sip Call
c=IN IP4 202.0.179.3
t=0 0
m=audio 18774 RTP/AVP 8 97
a=rtpmap:8 PCMA/8000
a=rtpmap:97 telephone-event/8000
a=fmtp:97 0-15
a=sendonly
<------------->
--- (9 headers 10 lines) ---
list_route: hop: <sip:44444444@202.0.179.3:5060;user=phone>
Found RTP audio format 8
Found RTP audio format 97
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 97
Capabilities: us - (alaw), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 202.0.179.3:18774
    -- Call on SIP/COMNET_PSTN-000000d9 placed on hold
    -- Started music on hold, class 'default', on SIP/XXXX-000000d8


一個打不通的空號, 撥號時會回傳Session Progress...
簡直不能接受！

lamsoft

已經投訴, 佢地電話回覆工程師盡量去攪... 但唔確保可以解到... 暈~