07-27-2019 09:13 AM - edited 07-27-2019 04:53 PM
Hi Experts,
I recently bought a Cisco CP-8945 phone for home use and loaded the latest SIP 9-4-2SR3-1 firmware. Unfortunately, I have found that all outbound calls fail immediately after pressed the call button, inbound calls however working perfectly.
I did a traffic dump and it looks like my caller number doesn't show properly in the From field, instead it shows Anonymous. As a consequence, I have received a 403 Spoofed From-URI detected from the SIP server.
Session Initiation Protocol (INVITE) Request-Line: INVITE sip:[CALLEE]@[SIP_SERVER];user=phone SIP/2.0 Message Header Via: SIP/2.0/UDP [IP_PHONE]:5060;branch=z9hG4bK58106e15 From: "Anonymous" <sip:Anonymous@[SIP_SERVER]>;tag=203a0783aecd0009224ea09a-376c7fae SIP Display info: "Anonymous" SIP from address: sip:Anonymous@[SIP_SERVER] SIP from tag: 203a0783aecd0009224ea09a-376c7fae To: <sip:[CALLEE]@[SIP_SERVER]> SIP to address: sip:[CALLEE]@[SIP_SERVER] Call-ID: 203a0783-aecd0004-56539248-7cd3edc8@[IP_PHONE] [Generated Call-ID: 203a0783-aecd0004-56539248-7cd3edc8@[IP_PHONE]] Max-Forwards: 70 Date: Sat, 27 Jul 2019 15:42:00 GMT CSeq: 101 INVITE User-Agent: Cisco-CP8945/9.4.2 Contact: <sip:[CALLER]@[IP_PHONE]:5060;transport=udp> Contact URI: sip:[CALLER]@[IP_PHONE]:5060;transport=udp Expires: 180 Accept: application/sdp Allow: ACK,BYE,CANCEL,INVITE,NOTIFY,OPTIONS,REFER,REGISTER,UPDATE,SUBSCRIBE,INFO Remote-Party-ID: "[CALLER]" <sip:[CALLER]@[SIP_SERVER]>;party=calling;id-type=subscriber;privacy=full;screen=yes
I can confirm the outbound calls are working fine if an appropriate phone number is included in the From header. Any ideas on how to make this work by manipulating the from header? I don't have CUCM set up yet because it might be too overkill for a home user.
Thanks in advance
07-27-2019 05:54 PM
07-27-2019 05:56 PM
Yes, my SIP server is Asterisk.
07-27-2019 08:21 PM
@GreenVine wrote:
Unfortunately, I have found that all outbound calls fail immediately after pressed the call button, inbound calls however working perfectly.
Post the dial plan.
07-27-2019 08:25 PM - edited 07-28-2019 01:12 AM
Thanks Leo. My dial plan is for Australian numbers, as follows:
<DIALTEMPLATE> <TEMPLATE MATCH="12...." TIMEOUT="0"/> <TEMPLATE MATCH="13...." TIMEOUT="0"/> <TEMPLATE MATCH="1300......" TIMEOUT="0"/> <TEMPLATE MATCH="1800......" TIMEOUT="0"/> <TEMPLATE MATCH="0........." TIMEOUT="0"/> <TEMPLATE MATCH="4......." TIMEOUT="0"/> <TEMPLATE MATCH="5......." TIMEOUT="0"/> <TEMPLATE MATCH="6......." TIMEOUT="0"/> <TEMPLATE MATCH="7......." TIMEOUT="0"/> <TEMPLATE MATCH="8......." TIMEOUT="0"/> <TEMPLATE MATCH="9......." TIMEOUT="0"/> </DIALTEMPLATE>
07-28-2019 01:04 AM
07-28-2019 01:11 AM - edited 07-28-2019 01:12 AM
Yes, incoming calls are fine. The phone is placed in a NAT-ed network and I have set my public IP address in the config file as well.
<natEnabled>true</natEnabled> <natReceivedProcessing>true</natReceivedProcessing> <natAddress>[PUBLIC_IP]</natAddress>
Wasn't sure whether this is caused by NAT issue as my softphone on iOS works for both directions perfectly.
INVITE traffic dump for normal outbound calls from my iOS:
From: "03xxxxxxxx"<sip:03xxxxxxxx@sip.redacted.com>;tag=8cd5b213
Comparatively, Cisco sends the following header:
From: "Anonymous" <sip:Anonymous@sip.redacted.com>;tag=203a0783aecd0009224ea09a-376c7fae
The SIP server immediately responds with:
SIP 450 Status: 403 Spoofed From-URI detected
07-28-2019 01:16 AM
@GreenVine wrote:
The phone is placed in a NAT-ed network and I have set my public IP address in the config file as well.
If you can receive incoming call but cannot make any outgoing calls, this is usually a NAT-related issue.
07-28-2019 01:19 AM
Thanks for your insights Leo. Any ideas on how to make NATed phone work? I did port forward 5060 to the phone though and it looks like my iOS soft phone is working inside the same NAT network.
Do you mind share an approach to modify the from header in the INVITE request?
07-28-2019 04:08 AM
12-30-2019 09:32 PM
Hello my friends. I've recently bought a 8945 and I installed the latest sip load 9.4.2SR3 and I have the exact problem as you had. Did you find a way to solve that?
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