[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=259123#comment-259123 ] 

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

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

> 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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