[asterisk-bugs] [JIRA] (ASTERISK-26063) ${PJSIP_HEADER(read, Call-ID)} does not work - documentation needs clarification for when read/write is possible

Private Name (JIRA) noreply at issues.asterisk.org
Sat Jun 18 23:34:56 CDT 2016


    [ https://issues.asterisk.org/jira/browse/ASTERISK-26063?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=231072#comment-231072 ] 

Private Name commented on ASTERISK-26063:
-----------------------------------------

The function CHANNEL(PJSIP,call-id) only works in Asterisk 13. It does not appear in the help
core show function CHANNEL in either version 11 or 12, and of course does not work.
I hereby request that this critical property be added to version 11 or 12, or both.
There is another critical property that is missing from Version 11 and 12, pjsip rtp_timeout.
Without this variable, pjsip cannot be used for serious business. I wonder if somebody could add this to version 11 and 12. Thousands, maybe millions of companies cannot use version 13 yet. We do business only with 11 and 12, more 11 than 12.
I would hate to go back and start using the old SIP channel again.



> ${PJSIP_HEADER(read,Call-ID)} does not work - documentation needs clarification for when read/write is possible
> ---------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-26063
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-26063
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_pjsip
>    Affects Versions: 13.9.1
>         Environment: Linux 64 bits
>            Reporter: Private Name
>            Assignee: Rusty Newton
>         Attachments: asterisk-error-1.txt, asterisk-error.txt
>
>
> I run the code below inside a b(handler) parameter in the Dial function.
> The code is excuted but it does not return the header Call-ID, which I need to capture for billing purposes. I mean on the outbound channel. In the old SIP channel, there was a variable that held that information. No it is empty, and I guess the code below should read the header in question.
> {noformat}
> exten => s,n,Set(SIPCALLID=${PJSIP_HEADER(read,Call-ID)}
> {noformat}



--
This message was sent by Atlassian JIRA
(v6.2#6252)



More information about the asterisk-bugs mailing list