[asterisk-bugs] [JIRA] (ASTERISK-30053) CHANGE SIP OPTIONS a=

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30053?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259124#comment-259124 ] 

Asterisk Team commented on ASTERISK-30053:
------------------------------------------

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.

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> CHANGE SIP OPTIONS a=
> ---------------------
>
>                 Key: ASTERISK-30053
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30053
>             Project: Asterisk
>          Issue Type: Information Request
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 13.32.0
>         Environment: CentOS Linux release 7.9.2009 
> Standard Serve Xeon 2 CPU / 8GB RAM
>            Reporter: Gerald
>
> 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 at 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.



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