[asterisk-bugs] [JIRA] (ASTERISK-24569) user=phone is not added to From, Contact and Diversion header
Mark Petersen (JIRA)
noreply at issues.asterisk.org
Thu Dec 4 02:47:29 CST 2014
[ https://issues.asterisk.org/jira/browse/ASTERISK-24569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223869#comment-223869 ]
Mark Petersen commented on ASTERISK-24569:
------------------------------------------
Hmm I do not agree as asterisk already has an option to add the user=phone
it is just not doing it on all the required headers
;usereqphone = no ; If yes, ";user=phone" is added to uri that contains a valid phone number
just because my provide require it, and they are annoying, do not mean that it is not a bug
user=phone is well described in RFC3261 & RFC3666 for how to interact with the PSTN world
witch asterisk has implement with the sip setting usereqphone
that you are not going to fix it without someone providing a patch, is a different thing, and I'm working on getting that done
but the first step is to report the bug, so others know that it exist
http://tools.ietf.org/html/rfc3261
http://tools.ietf.org/html/rfc3666
http://tools.ietf.org/html/rfc4967
> user=phone is not added to From, Contact and Diversion header
> -------------------------------------------------------------
>
> Key: ASTERISK-24569
> URL: https://issues.asterisk.org/jira/browse/ASTERISK-24569
> Project: Asterisk
> Issue Type: Bug
> Security Level: None
> Components: Channels/chan_sip/General
> Affects Versions: 11.14.1
> Environment: cento 6.5
> Reporter: Mark Petersen
>
> the problem is that user=phone is not added to the From, Contact and Diversion header, but is correctly added to the INVITE and To header
> This is a major problem as our Provider is switching to a new platform where they require these header, in order for os to set the outgoing CALLERID on our trunk
> [general]
> usereqphone=yes
> Set(CALLERID(name)=77777777);
> Set(CALLERID(num)=88888888);
> Set(CALLERID(ANI-num)=99999999);
> Set(CALLERID(rdnis)=66666666);
> Dial(SIP/55555555 at 192.168.0.2);
> INVITE sip:55555555 at 192.168.0.1;user=phone SIP/2.0
> Via: SIP/2.0/UDP 192.168.0.1:5060;branch=z9hG4bK1b298676;rport
> From: "77777777" <sip:88888888 at 192.168.0.1>;tag=as14ad576b
> To: <sip:55555555 at 192.168.0.2;user=phone>
> Contact: <sip:88888888 at 192.168.0.1:5060>
> Call-ID: 566480180a198f053ee9ba1016c0aef8 at 192.168.0.1
> CSeq: 102 INVITE
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Diversion: <sip:66666666 at 192.168.0.1>;reason=unknown
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