SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP and receiving rport=52891

[dsobrinho]

Hi,

We have a problem with ITSP. My outbound calls cannot be completed. The ITSP monitor the calls outbound and report me that we are not sent the correct port sip 5060.

So in the wireshark I see that port 52891 is a source port from gateway.

<------------->

Following the calls:

--- (22 headers 11 lines) ---
Sending to 10.50.70.28:52891 (NAT)
Sending to 10.50.70.28:52891 (NAT)
Using INVITE request as basis request - C7F6BE6-F90011E5-9F95B8C1-B0528DA7@10.50.70.28
No matching peer for '17480670' from '10.50.70.28:52891'

<--- Reliably Transmitting (NAT) to 10.50.70.28:52891 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 10.50.70.28:5060;branch=z9hG4bK7BD88;received=10.50.70.28;rport=52891
From: "Ruben Salazar" <sip:17480670@172.20.2.1>;tag=4C0A4EC-1CF7
To: <sip:6163131@172.20.2.1>;tag=as2b515ecd
Call-ID: C7F6BE6-F90011E5-9F95B8C1-B0528DA7@10.50.70.28
CSeq: 101 INVITE
Server: Asterisk360EE-access1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="access1", nonce="094a3c46"
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog 'C7F6BE6-F90011E5-9F95B8C1-B0528DA7@10.50.70.28' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:10.50.70.28:52891 --->
ACK sip:6163131@172.20.2.1:5060 SIP/2.0
Via: SIP/2.0/UDP 10.50.70.28:5060;branch=z9hG4bK7BD88
From: "Ruben Salazar" <sip:17480670@172.20.2.1:5060>;tag=4C0A4EC-1CF7
To: <sip:6163131@172.20.2.1>;tag=as2b515ecd
Date: Sun, 03 Apr 2016 18:23:50 GMT
Call-ID: C7F6BE6-F90011E5-9F95B8C1-B0528DA7@10.50.70.28
Max-Forwards: 70
CSeq: 101 ACK
Allow-Events: telephone-event
Content-Length: 0

<------------->

[Amit Sharawat · Accepted solution]

Hi,

The source port could be anything, but the destination port should be 5060 in case of SIP, which correct in our case.

Please ask your service provider which port are they talking about?

Please rate if helpful

~Amit

[dsobrinho]

Hi Amit Sharawat,

You are correct. I have a lot of problem with ITSP. then I had to change the header of SIP invite and apply a command in sip configuration to work (connection-reuse). The ITSP did not help me in nothing.

voice class sip-profiles 1
request INVITE sip-header From modify "<sip:*.*>" "<sip:12333339@10.50.X0.YY:5060>"
!
!
sip-ua
calling-info pstn-to-sip from number set 12333339
registrar ipv4:172.20.Y.X:5060 expires 3600
sip-server ipv4:172.20.Y.X:5060
connection-reuse

After that the outgoing  and inbound calls works fine.

Thank you

Best regards,

Daniel Sobrinho

[kalungiabrahams]

so how do you change this if you are using XML document to configure sip phones

[Roger Kallberg]

Your question is off topic of the OP. Please create your own question instead of posting your question on an old market as answered post.