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Summary:ASTERISK-30053: CHANGE SIP OPTIONS a=
Reporter:Gerald (gfesj)Labels:
Date Opened:2022-05-10 19:48:40Date Closed:2022-05-10 19:48:42
Priority:TrivialRegression?
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:13.32.0 Frequency of
Occurrence
Related
Issues:
Environment:CentOS Linux release 7.9.2009 Standard Serve Xeon 2 CPU / 8GB RAMAttachments:
Description:I need to change SIP OPTIONS from a=fmtp:102 0-16 to a=fmtp:1010-15, but when we set __SIP_URI_OPTIONS=a=fmtp:1010-15, it does not take place at SIP OPTIONS, as we can se bellow

[May 10 21:43:48] SIP/2.0 183 Session Progress
[May 10 21:43:48] Via: SIP/2.0/UDP 192.168.69.5:63666;branch=z9hG4bKPjpOCvKsZK.hEa5POCJtKWOZBCPzysvzvT;received=192.168.69.5;rport=63666
[May 10 21:43:48] From: "4209 @ CGP" <sip:4209@192.168.0.4>;tag=aWLWBAEGdcyMFHFHT-Au2tqyGYHuOiNT
[May 10 21:43:48] To: sip:9303*****@192.168.0.4;tag=as5153ca66
[May 10 21:43:48] Call-ID: T1El.P6vIbATynFygsTlt1gs7hb7Sak-
[May 10 21:43:48] CSeq: 28156 INVITE
[May 10 21:43:48] Server: Asterisk PBX 13.32.0
[May 10 21:43:48] Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
[May 10 21:43:48] Supported: replaces, timer
[May 10 21:43:48] Contact: <sip:9303*****@192.168.0.4:5060>
[May 10 21:43:48] Content-Type: application/sdp
[May 10 21:43:48] Content-Length: 260
[May 10 21:43:48]
[May 10 21:43:48] v=0
[May 10 21:43:48] o=root 782853149 782853149 IN IP4 192.168.0.4
[May 10 21:43:48] s=Asterisk PBX 13.32.0
[May 10 21:43:48] c=IN IP4 192.168.0.4
[May 10 21:43:48] t=0 0
[May 10 21:43:48] m=audio 12928 RTP/AVP 0 8 102
[May 10 21:43:48] a=rtpmap:0 PCMU/8000
[May 10 21:43:48] a=rtpmap:8 PCMA/8000
[May 10 21:43:48] a=rtpmap:102 telephone-event/8000
[May 10 21:43:48] a=fmtp:102 0-16
[May 10 21:43:48] a=maxptime:150
[May 10 21:43:48] a=sendrecv

How can we set this options to take place at channel.

Comments:By: Asterisk Team (asteriskteam) 2022-05-10 19:48:41.720-0500

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines
By: Asterisk Team (asteriskteam) 2022-05-10 19:48:42.378-0500

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].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

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