[asterisk-bugs] [JIRA] (ASTERISK-24569) user=phone is not added to From, Contact and Diversion header

[Mark Petersen (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24569?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=223888#comment-223888 ] 

Mark Petersen edited comment on ASTERISK-24569 at 12/4/14 4:21 PM:
-------------------------------------------------------------------

Thanks
yes I'm working with the provider to remove that requirement
but as they are switching to a new huawei platform for their transition to VoLTE, where the new platform require that we send this option
they are not very forthcoming in recognizing that what they do is not optimal

my plan is to add a regx that will validate if it is a number

{noformat}
^\+?[0-9a-dA-D*#]{1,15}$
{noformat}

this will match the following
 *  optional first special character +
 *  the digits 0-9
 *  the special characters # and *
 *  the DTMF digits A-D
 *  number must be between 1 and 15 digits/char long, not including +

I'm not sure where I should allow "-" and space in the number
all the examples in the RFC's the SIP header newer uses them
they are only used in the description like this:

the phone number +1-415-767-6791
{noformat}
<sip:+14157676791 at example.com;user=phone>
{noformat}


was (Author: asterisk.org at zombie.dk):
Thanks
yes I'm working with the provider to remove that requirement
but as they are switching to a new huawei platform for their transition to VoLTE, where the new platform require that we send this option
they are not very forthcoming in recognizing that what they do is not optimal

my plan is to add a regx that will validate if it is a number

{noformat}
^\+?[0-9a-dA-D*#]{1,15}$
{noformat}

this will match the following
 *  optional first special character +
 *  the digits 0-9
 *  the special characters # and *
 *  the DTMF digits A-D
 *  number must be between 1 and 15 digits/char long, not including +

I'm not sure where I should allow "-" and space in the number
all the examples in the RFC the SIP header newer uses them
they are only used in the description like this:

the phone number +1-415-767-6791
{noformat}
<sip:+14157676791 at example.com;user=phone>
{noformat}

> 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
> {noformat}
> [general]
> usereqphone=yes
> {noformat}
> {noformat}
> 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);
> {noformat}
> {noformat}
> 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
> {noformat}



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