Problems with Avaya CS1000 adding phone-context=cdp.udp

[BAYSHORE]

I have a SIP trunk between AVAYA CS1000 and 3cx.....when a call comes in from the CS1000 it is adding phone-context=cdp.udp to the TO: Field in the INVITE.....Does anyone know to strip this out of the TO: Field


Here is the LOG

I replaced the Hostname and the IP addresses to protect the innocent....

16:19:15.585 [CM102001]: Authentication failed for SipReq: INVITE 1384;[email protected]:5060 tid=AC1F287B26254FF30787782-AP cseq=INVITE contact=46539;[email protected]:5060 / 2 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
16:19:15.585 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:1384;[email protected]:5060;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0
Via: SIP/2.0/UDP 172.31.xxx.xxx;rport=5060;branch=z9hG4bKAC1F287B26254FF30787782-AP;ft=172.31.xxx.xxx~13c4
Via: SIP/2.0/UDP 172.31.xxx.xxx:15070;branch=z9hG4bKAC1F287B26254FF30787782
Via: SIP/2.0/UDP 172.31.xxx.xxx:15070;branch=z9hG4bKAC1F287B26254FF31787780
Via: SIP/2.0/UDP 172.31.xxx.xxx:15070;branch=z9hG4bKAC1F287B26254FF31787779
Via: SIP/2.0/UDP 172.31.xxx.xxx;rport;branch=z9hG4bK-1567241-39ae4f73-11632959-AP;ft=172.31.xxx.xxx~13c4
Via: SIP/2.0/UDP 172.31.xxx.xxx:5060;branch=z9hG4bK-1567241-39ae4f73-11632959
Max-Forwards: 66
Route: <sip:xxx.xxx.xxx.xxx;lr;phase=terminating>
Record-Route: <sip:[email protected];transport=udp;lr>
Record-Route: <sip:172.31.xxx.xxx:15060;lr;sap=-1867748178*1*016asm-callprocessing.sar2066226860~1360102755617~161401105~1>
Record-Route: <sip:[email protected];transport=udp;lr>
Contact: <sip:46539;[email protected]:5060;maddr=172.31.xxx.xxx;transport=udp;user=phone>
To: <sip:1384;[email protected];user=phone>
From: "RUDY'S TEST PHONE"<sip:46539;[email protected];user=phone>;tag=84010a8-5281fac-13c4-40030-1567241-3ebc44a-1567241
Call-ID: afd2d560-5281fac-13c4-40030-1567241-5528cce4-1567241
CSeq: 2 INVITE
Min-SE: 0
Alert-Info: <cid:[email protected]>
Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE
Proxy-Authorization: Digest username="ACH_1000E",realm="3CXPhoneSystem",nonce="414d535c0722166347:339e340d3980e02080dbad9f0719ebb7",uri="sip:1384;[email protected]:5060;maddr=172.31.xxx.xxx;transport=udp;user=phone;x-nt-redirect=redirect-server",response="d9ddce97f9c2f8d5623e338a27583544",algorithm=MD5
Supported: 100rel, x-nortel-sipvc, replaces, timer
User-Agent: Nortel CS1000 SIP GW release_6.0 version_ssLinux-6.00.18 AVAYA-SM-6.1.5.0.615006
Privacy: none
P-Asserted-Identity: "RUDY'S TEST PHONE" <sip:46539;[email protected];user=phone>
Content-Length: 0
x-nt-corr-id: 001f9f5a1013100502@00247f969c52-54cdc122
History-Info: <sip:1384;[email protected];user=phone>;index=1,<sip:1384;[email protected];user=phone>;index=1.1
P-Location: SM;origlocname="NRS";termlocname="NNC"
________________________________________

 

[SY]

when a call comes in from the CS1000 it is adding phone-context=cdp.udp to the TO: Field in the INVITE.....Does anyone know to strip this out of the TO: Field
...

Contact: <sip:46539;[email protected]:5060;maddr=172.31.xxx.xxx;transport=udp;user=phone>
To: <sip:1384;[email protected];user=phone>
From: "RUDY'S TEST PHONE"<sip:46539;[email protected];user=phone>;tag=84010a8-5281fac-13c4-40030-1567241-3ebc44a-1567241

3CX PhoneSystem is very tolerant to interpretations of RFC 3261 (definition of SIP protocol), but these headers are going far beyond the bounds of possible tolerance...

Sorry, but it will not work even if you remove that "parameter"

 

[71jav2t]

On the 3CX server, try putting this into the SIP ID of the extension. It is located in the tab labeled "Other"

1384;phone-context=cdp.udp

I think this will work if you are running version 11.

The cdp.udp context can also be removed in the CS1000, but I would not recommend unless you know what you are doing.

 

[SY]

On the 3CX server, try putting this into the SIP ID of the extension. It is located in the tab labeled "Other"

1384;phone-context=cdp.udp

I think this will work if you are running version 11.

The cdp.udp context can also be removed in the CS1000, but I would not recommend unless you know what you are doing.

... and what will it resolve?

 

[71jav2t]

I have a SIP trunk between AVAYA CS1000 and 3cx.....when a call comes in from the CS1000 it is adding phone-context=cdp.udp to the TO: Field in the INVITE.....Does anyone know to strip this out of the TO: Field


Here is the LOG

I replaced the Hostname and the IP addresses to protect the innocent....

16:19:15.585 [CM102001]: Authentication failed for SipReq: INVITE 1384;[email protected]:5060 tid=AC1F287B26254FF30787782-AP cseq=INVITE contact=46539;[email protected]:5060 / 2 from(wire); Reason: Credentials don't match, check that authorization-ID and password match the ones in extension settings
16:19:15.585 [CM500002]: Unidentified incoming call. Review INVITE and adjust source identification:
INVITE sip:1384;[email protected]:5060;transport=udp;user=phone;x-nt-redirect=redirect-server SIP/2.0
Via: SIP/2.0/UDP 172.31.xxx.xxx;rport=5060;branch=z9hG4bKAC1F287B26254FF30787782-AP;ft=172.31.xxx.xxx~13c4
Via: SIP/2.0/UDP 172.31.xxx.xxx:15070;branch=z9hG4bKAC1F287B26254FF30787782
Via: SIP/2.0/UDP 172.31.xxx.xxx:15070;branch=z9hG4bKAC1F287B26254FF31787780
Via: SIP/2.0/UDP 172.31.xxx.xxx:15070;branch=z9hG4bKAC1F287B26254FF31787779
Via: SIP/2.0/UDP 172.31.xxx.xxx;rport;branch=z9hG4bK-1567241-39ae4f73-11632959-AP;ft=172.31.xxx.xxx~13c4
Via: SIP/2.0/UDP 172.31.xxx.xxx:5060;branch=z9hG4bK-1567241-39ae4f73-11632959
Max-Forwards: 66
Route: <sip:xxx.xxx.xxx.xxx;lr;phase=terminating>
Record-Route: <sip:[email protected];transport=udp;lr>
Record-Route: <sip:172.31.xxx.xxx:15060;lr;sap=-1867748178*1*016asm-callprocessing.sar2066226860~1360102755617~161401105~1>
Record-Route: <sip:[email protected];transport=udp;lr>
Contact: <sip:46539;[email protected]:5060;maddr=172.31.xxx.xxx;transport=udp;user=phone>
To: <sip:1384;[email protected];user=phone>
From: "RUDY'S TEST PHONE"<sip:46539;[email protected];user=phone>;tag=84010a8-5281fac-13c4-40030-1567241-3ebc44a-1567241
Call-ID: afd2d560-5281fac-13c4-40030-1567241-5528cce4-1567241
CSeq: 2 INVITE
Min-SE: 0
Alert-Info: <cid:[email protected]>
Allow: INVITE, ACK, BYE, REGISTER, REFER, NOTIFY, CANCEL, PRACK, OPTIONS, INFO, SUBSCRIBE, UPDATE
Proxy-Authorization: Digest username="ACH_1000E",realm="3CXPhoneSystem",nonce="414d535c0722166347:339e340d3980e02080dbad9f0719ebb7",uri="sip:1384;[email protected]:5060;maddr=172.31.xxx.xxx;transport=udp;user=phone;x-nt-redirect=redirect-server",response="d9ddce97f9c2f8d5623e338a27583544",algorithm=MD5
Supported: 100rel, x-nortel-sipvc, replaces, timer
User-Agent: Nortel CS1000 SIP GW release_6.0 version_ssLinux-6.00.18 AVAYA-SM-6.1.5.0.615006
Privacy: none
P-Asserted-Identity: "RUDY'S TEST PHONE" <sip:46539;[email protected];user=phone>
Content-Length: 0
x-nt-corr-id: 001f9f5a1013100502@00247f969c52-54cdc122
History-Info: <sip:1384;[email protected];user=phone>;index=1,<sip:1384;[email protected];user=phone>;index=1.1
P-Location: SM;origlocname="NRS";termlocname="NNC"
________________________________________

It should allow for extensions on the Avaya CS1000 to make calls to extensions on the 3CX. Specifically in the example above, extension 46539 on the CS1000 should be able to call extension 1384 on the 3CX.