3cx MP-118 Speech Server

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pranaywilson

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I have a SIP Phone registered as extension 100 on the 3cx system.
I also have all ports of a MP-118 gateway registered as lines of the 3cx system (10000 - 10007)
All the lines are successfully registered.

I have a speech application which receives the call from the gateway. After some speech activities, the speech application blind transfers to the SIP phone.
The uri used for the blind transfer is 100@3cx.
On receiving the REFER message the gateway sends an INVITE to the 3cx system, upon which the 3cx requests for authentication. Now the port on which the call has been received has user name: 10000 password: 10000, but the line I want to call is 100. So the gateway provides with the credentials configured for its port (10000), but the extension is 100. Therefore the authentication fails.
The logs are shown below.

1:29:43.046 AuthMgr::onAuthFailure [CM102001]: Authentication failed for SipReq: INVITE [email protected] tid=ac892143327 cseq=INVITE [email protected] / 2 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
11:29:42.750 evt::CheckAuth::not_handled [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:[email protected] SIP/2.0
Via: SIP/2.0/UDP 192.168.1.9;branch=z9hG4bKac892143327
Max-Forwards: 70
Contact: [sip:[email protected]]
To: [sip:[email protected]]
From: "ORDER CATCHER"[sip:[email protected]:5062];tag=1c891059758
Call-ID: [email protected]
CSeq: 2 INVITE
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
Proxy-Authorization: Digest username="100",realm="3CXPhoneSystem",nonce="12847825782:370101bb130a6161707d4b9c90d2821f",uri="sip:[email protected]",algorithm=MD5,response="1257de4950387160c6238f0dc6ae6161"
Supported: em, timer, replaces, path, resource-priority
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.017.001
Referred-By: [sip:[email protected]]
Content-Length: 0
Remote-Party-ID: "ORDER CATCHER" [sip:[email protected]];party=calling;privacy=off;screen=no;screen-ind=0;npi=0;ton=0, [sip:[email protected]];party=called;npi=0;ton=0

11:29:42.750 evt::CheckAuth::not_handled [CM302002]: Authentication failed due to unidentified source of: SipReq: INVITE [email protected] tid=ac892143327 cseq=INVITE [email protected] / 2 from(wire)
11:29:42.328 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 192.168.1.9;branch=z9hG4bKac891067054
Max-Forwards: 70
Contact: [sip:[email protected]]
To: [sip:[email protected]]
From: "ORDER CATCHER"[sip:[email protected]:5062];tag=1c891059758
Call-ID: [email protected]
CSeq: 1 INVITE
Allow: REGISTER, OPTIONS, INVITE, ACK, CANCEL, BYE, NOTIFY, PRACK, REFER, INFO, SUBSCRIBE, UPDATE
Supported: em, timer, replaces, path, resource-priority
User-Agent: Audiocodes-Sip-Gateway-MP-118 FXO/v.5.20A.017.001
Referred-By: [sip:[email protected]]
Content-Length: 0
Remote-Party-ID: "ORDER CATCHER" [sip:[email protected]];party=calling;privacy=off;screen=no;screen-ind=0;npi=0;ton=0, [sip:[email protected]];party=called;npi=0;ton=0

11:29:42.328 evt::CheckIfAuthIsRequired::not_handled [CM302001]: Authorization system can not identify source of: SipReq: INVITE [email protected] tid=ac891067054 cseq=INVITE [email protected] / 1 from(wire)
 
bump?

Any suggestions?
 
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