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Summary:ASTERISK-29956: chan_sip: settings not set correctly - useragent, sdpsession,sdpowner
Reporter:Mihail Belobrov (brost1986)Labels:
Date Opened:2022-03-07 09:53:22.000-0600Date Closed:2022-03-21 12:00:01
Priority:MinorRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/General
Versions:16.24.1 Frequency of
Occurrence
Related
Issues:
Environment:Debian 10Attachments:
Description:settings not set correctly
useragent=dlink 12-3892-6657-1.3.3.198-ON201LW                                                                                                                                                                                                                                  
sdpsession=dlink 12-3892-6657-1.3.3.198-ON201LW                                                                                                                                                                                                                                
sdpowner=Dlink  


Global Settings:
----------------
 UDP Bindaddress:        0.0.0.0:5060
 TCP SIP Bindaddress:    0.0.0.0:5060
 TLS SIP Bindaddress:    Disabled
 RTP Bindaddress:        Disabled
 Videosupport:           Yes
 Textsupport:            No
 Ignore SDP sess. ver.:  No
 AutoCreate Peer:        Off
 Match Auth Username:    Yes
 Allow unknown access:   No
 Allow subscriptions:    Yes
 Allow overlap dialing:  No
 Allow promisc. redir:   No
 Enable call counters:   No
 SIP domain support:     No
 Path support :          No
 Realm. auth:            No
 Our auth realm          dlink
 Use domains as realms:  No
 Call to non-local dom.: Yes
 URI user is phone no:   No
 Always auth rejects:    Yes
 Direct RTP setup:       No
 User Agent:             dlink 12-3892-6657-1.3.3.198-ON201LW
 SDP Session Name:       dlink 12-3892-6657-1.3.3.198-ON201LW
 SDP Owner Name:         Dlink
 Reg. context:           (not set)
 Regexten on Qualify:    No
 Trust RPID:             No
 Send RPID:              No
 Legacy userfield parse: No
 Send Diversion:         Yes
 Caller ID:              asterisk
 From: Domain:          
 Record SIP history:     Off
 Auth. Failure Events:   Off
 T.38 support:           No
 T.38 EC mode:           Unknown
 T.38 MaxDtgrm:          4294967295
 SIP realtime:           Disabled
 Qualify Freq :          60000 ms
 Q.850 Reason header:    Yes
 Store SIP_CAUSE:        No

But when call from one exten to other, in sip dump i see "Server: Asterisk PBX 16.24.1" for all reply packet
If my Asterisk send Register to other Sip Provider, i see correctly user agent "User-Agent: dlink 12-3892-6657-1.3.3.198-ON201LW  "
Comments:By: Asterisk Team (asteriskteam) 2022-03-07 09:53:22.767-0600

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

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By: Joshua C. Colp (jcolp) 2022-03-07 10:05:06.090-0600

We require additional debug to continue with triage of your issue. Please follow the instructions on the wiki [1] for how to collect debugging information from Asterisk. For expediency, where possible, attach the debug with a '.txt' file extension so that the debug will be usable for further analysis.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information


By: Asterisk Team (asteriskteam) 2022-03-21 12:00:00.675-0500

Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1].

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines