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Internal dial error, unidentified incoming call

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et381

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Hi,

WIth our Linksys SPA942 phones, it cannot dial internally, the following is logged. The error is "Incoming call rejected" and "Unidentified incoming call". Anyone seen this?


17:17:17.718 CallCtrl::onIncomingCall [CM502001]: Source info: From: 20; To: "7900"[sip:[email protected]];tag=25c712c4307eefb0o3[sip:[email protected]]
17:17:17.718 CallCtrl::onIncomingCall [CM503012]: Call(20): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected]:7060 tid=-b5492afc cseq=INVITE [email protected]:5063 / 102 from(wire)
17:17:17.046 evt::CheckIfAuthIsRequired::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected]:7060 SIP/2.0
Via: SIP/2.0/UDP 10.1.2.103:5063;branch=z9hG4bK-96decfd0;rport=5063
Max-Forwards: 70
Contact: "7900"[sip:[email protected]:5063]
To: [sip:[email protected]]
From: "7900"[sip:[email protected]];tag=25c712c4307eefb0o3
Call-ID: [email protected]
CSeq: 101 INVITE
Expires: 240
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER
Supported: replaces
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0


17:17:17.046 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected]:7060 tid=-96decfd0 cseq=INVITE [email protected]:5063 / 101 from(wire)
 
So still couldn't get it to work for our evaluation of 3cx. It just fails to work with our Linksys SPA942. Soft phone seems to be working. Here is the SIP trace for the error. If there are any guru that can suggest what might be going on, its greatly appreciated.


INVITE sip:[email protected]:6999 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.120:5063;branch=z9hG4bK-f7983e3f;rport
From: "7001" <sip:[email protected]>;tag=e5b8d2e32cfa4edco3
To: <sip:[email protected]>
Remote-Party-ID: "7001" <sip:[email protected]:6999>;screen=yes;party=calling
Call-ID: [email protected]
CSeq: 101 INVITE
Max-Forwards: 70
Contact: "7001" <sip:[email protected]:5063>
Expires: 240
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp

v=0
o=- 2532 2532 IN IP4 192.168.10.120
s=-
c=IN IP4 192.168.10.120
t=0 0
m=audio 16480 RTP/AVP 8 0 18 2 4 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729a/8000
a=rtpmap:2 G726-32/8000
a=rtpmap:4 G723/8000
a=rtpmap:96 G726-40/8000
a=rtpmap:97 G726-24/8000
a=rtpmap:98 G726-16/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=ptime:20
a=sendrecv


SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.10.120:5063;branch=z9hG4bK-f7983e3f;rport=1161;received=10.1.2.254
Proxy-Authenticate: Digest nonce="12855223609:4ba37e1f3a6cdba1faa08998658e3f57",algorithm=MD5,realm="3CXPhoneSystem"
To: <sip:[email protected]>;tag=cf6b064c
From: "7001"<sip:[email protected]>;tag=e5b8d2e32cfa4edco3
Call-ID: [email protected]
CSeq: 101 INVITE
User-Agent: 3CXPhoneSystem 5.1.4510.0
Content-Length: 0


ACK sip:[email protected]:6999 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.120:5063;branch=z9hG4bK-f7983e3f;rport
From: "7001" <sip:[email protected]>;tag=e5b8d2e32cfa4edco3
To: <sip:[email protected]>;tag=cf6b064c
Call-ID: [email protected]
CSeq: 101 ACK
Max-Forwards: 70
Contact: "7001" <sip:[email protected]:5063>
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0


INVITE sip:[email protected]:6999 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.120:5063;branch=z9hG4bK-42627973;rport
From: "7001" <sip:[email protected]>;tag=e5b8d2e32cfa4edco3
To: <sip:[email protected]>
Remote-Party-ID: "7001" <sip:[email protected]:6999>;screen=yes;party=calling
Call-ID: [email protected]
CSeq: 102 INVITE
Max-Forwards: 70
Proxy-Authorization: Digest username="7001",realm="3CXPhoneSystem",nonce="12855223609:4ba37e1f3a6cdba1faa08998658e3f57",uri="sip:[email protected]:6999",algorithm=MD5,response="eb0eccb10d922bfbaaa42512dcee46ac"
Contact: "7001" <sip:[email protected]:5063>
Expires: 240
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 395
Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, PRACK, REFER
Supported: 100rel, replaces
Content-Type: application/sdp

v=0
o=- 2532 2532 IN IP4 192.168.10.120
s=-
c=IN IP4 192.168.10.120
t=0 0
m=audio 16480 RTP/AVP 8 0 18 2 4 96 97 98 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:18 G729a/8000
a=rtpmap:2 G726-32/800...


SIP/2.0 403 Forbidden
Via: SIP/2.0/UDP 192.168.10.120:5063;branch=z9hG4bK-42627973;rport=1161;received=10.1.2.254
To: <sip:[email protected]>;tag=546c0764
From: "7001"<sip:[email protected]>;tag=e5b8d2e32cfa4edco3
Call-ID: [email protected]
CSeq: 102 INVITE
User-Agent: 3CXPhoneSystem 5.1.4510.0
Warning: 499 csmpbx01 "Caller is forbidden"
Content-Length: 0


ACK sip:[email protected]:6999 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.120:5063;branch=z9hG4bK-42627973;rport
From: "7001" <sip:[email protected]>;tag=e5b8d2e32cfa4edco3
To: <sip:[email protected]>;tag=546c0764
Call-ID: [email protected]
CSeq: 102 ACK
Max-Forwards: 70
Proxy-Authorization: Digest username="7001",realm="3CXPhoneSystem",nonce="12855223609:4ba37e1f3a6cdba1faa08998658e3f57",uri="sip:[email protected]:6999",algorithm=MD5,response="eb0eccb10d922bfbaaa42512dcee46ac"
Contact: "7001" <sip:[email protected]:5063>
User-Agent: Linksys/SPA942-5.2.8
Content-Length: 0
 
Show extension settings for ext. 7001 and 7900.
 
Here's the extension settings on 3CX from the settings backup file.

<row>
<field name="idextension">8</field>
<field name="fkiddn">15</field>
<field name="authid">7900</field>
<field name="authpswd">7900</field>
<field name="outcid">Conference Room 1</field>
<field name="recrdout">0</field>
<field name="recrdin">0</field>
<field name="accstatus">0</field>
<field name="accconfig">0</field>
<field name="fwdallcalls">0</field>
<field name="noanswtimeout">15</field>
<field name="noanswringing">1</field>
<field name="busydetection">2</field>
<field name="isinternal">1</field>
<field name="isbindtoms">0</field>
<field name="issupportreinvite">1</field>
<field name="issupportreplaces">1</field>
<field name="castatus">0</field>
<field name="qstatus">0</field>
<field name="macaddress"></field>
<field name="templateused"></field>
<field name="filename1"></field>
<field name="filename2"></field>
<field name="interface"></field>
</row>

<row>
<field name="idextension">2</field>
<field name="fkiddn">5</field>
<field name="authid">7001</field>
<field name="authpswd">7001</field>
<field name="outcid"></field>
<field name="recrdout">0</field>
<field name="recrdin">0</field>
<field name="accstatus">0</field>
<field name="accconfig">0</field>
<field name="fwdallcalls">0</field>
<field name="noanswtimeout">15</field>
<field name="noanswringing">1</field>
<field name="busydetection">2</field>
<field name="isinternal">1</field>
<field name="isbindtoms">0</field>
<field name="issupportreinvite">1</field>
<field name="issupportreplaces">1</field>
<field name="castatus">0</field>
<field name="qstatus">0</field>
<field name="macaddress"></field>
<field name="templateused"></field>
<field name="filename1"></field>
<field name="filename2"></field>
<field name="interface"></field>
</row>
 
Two questions:
1) at what port your 3CX SIP server is working? First listing shows 7060, second - 6999. Can you just run it on 5060?
2) those phones from another network (192.168.x.x), and PBX is at network 10.x.x.x. Moreover, they address their requests to 10.1.2.50:6999, but PBX receives them from 10.1.2.254:1161 (which is supposedly a router between those networks). Moreover, From URI is <sip:[email protected]> although PBX runs on non default port. In this case From URI should be <sip:[email protected]:6999> (if PBX runs on port 6999), and also should have Request line and To URI as <sip:[email protected]:6999>

Should warn you that after you will fix it and phones will register you may still experience troubles calling to that phones in case of inappropriate settings of 10.1.2.254 router.
 
The different IP addresses and port was me trying out different settings to make it work. But none worked, all same error of unidentified incoming call.

Interestingly, I downgraded the phone firmware to 5.1.18 and it worked straight away. So for some reason 5.2.5 and 5.2.8 firmware of the Linksys SPA942 won't work with 3CX. When I get more time next week, I'll try again with 5.2.8 firmware and see what's going on.
 
I am a enterprise version v5 user, and i have the same problem with you but on v6 RC and v6 final version only...(on v5 I have no this problem)
But the most strange thing is, I have 3 external , these are 13, 14 and 15
When i use deskphone 13, I can't call 14 and 15 , but phone 14,15 can call to 13...
and I have tried 3CX VOIP phone to login as 13, It work fine...@@

I think that may the bug on v6 version...maybe you should try v5 final version first...


as list is my log info
-----------------------------------------------------------------------------------------------
01:35:42.034 CallCtrl::onIncomingCall [CM503013]: Call(4): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=9bc96c7426 cseq=INVITE [email protected]:5060 / 802 from(wire)
01:35:42.034 CallLeg::onNewCall [CM500002]: Info on incoming INVITE:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;rport=5060;branch=z9hG4bK9bc96c7426
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: "Ken@VMS3CX"[sip:[email protected]];tag=486b4c86
Call-ID: [email protected]
CSeq: 802 INVITE
Allow: INVITE, CANCEL, ACK, BYE, NOTIFY, REFER, OPTIONS, INFO, MESSAGE
Proxy-Authorization: Digest username="13",realm="3CXPhoneSystem",nonce="12858888941:34a34074f9217428ab1bf0dfc844e0df",response="9d5a7415c8bbe96f11023144323974f9",uri="sip:[email protected]",algorithm=MD5
Supported: replaces
User-Agent: DGP306 (711090)
Content-Length: 0
 
Remove @ from display name, please. If problem persist - tell us what phone are you using.
 
yeh, thx I've remove the @ char, but the problem still a live...
here's are new test log...
==============================================================
09:48:23.425 Call::Terminate [CM503008]: Call(3): Call is terminated
09:48:23.425 CallCtrl::onIncomingCall [CM502001]: Source info: From: 3; To: "Ken_VMS3CX"[sip:[email protected]];tag=197e5f8e[sip:[email protected]]
09:48:23.425 CallCtrl::onIncomingCall [CM503013]: Call(3): Incoming call rejected, caller is unknown; msg=SipReq: INVITE [email protected] tid=006920903c cseq=INVITE [email protected]:5060 / 802 from(wire)
09:48:22.925 evt::CheckIfAuthIsRequired::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.0.110:5060;rport=5060;branch=z9hG4bK6e3dd029f0
Max-Forwards: 70
Contact: [sip:[email protected]:5060]
To: [sip:[email protected]]
From: "Ken_VMS3CX"[sip:[email protected]];tag=197e5f8e
Call-ID: [email protected]
CSeq: 801 INVITE
Allow: INVITE, CANCEL, ACK, BYE, NOTIFY, REFER, OPTIONS, INFO, MESSAGE
Supported: replaces
User-Agent: DGP306 (711090)
Content-Length: 0


09:48:22.925 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected] tid=6e3dd029f0 cseq=INVITE [email protected]:5060 / 801 from(wire)
 
execuse me, my big fault...@@
maybe last time I tried v6 RC1 version, I make the same fault..@@
thx again...
 
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