ASTERISK-24569: user=phone is not added to From, Contact and Diversion header

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Summary: ASTERISK-24569: user=phone is not added to From, Contact and Diversion header

Reporter: Mark Petersen (asterisk.org@zombie.dk) Labels:

Date Opened: 2014-12-01 03:46:57.000-0600 Date Closed:

Priority: Major Regression?

Status: Open/New Components: Channels/chan_sip/General

Versions: 11.14.1 13.18.4 Frequency of
Occurrence Constant

Related
Issues:

Environment: cento 6.5 Attachments: ( 0) chan_sip.24569.2.patch

Description: 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@192.168.0.2);
{noformat}
{noformat}
INVITE sip:55555555@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@192.168.0.1>;tag=as14ad576b
To: <sip:55555555@192.168.0.2;user=phone>
Contact: <sip:88888888@192.168.0.1:5060>
Call-ID: 566480180a198f053ee9ba1016c0aef8@192.168.0.1
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Diversion: <sip:66666666@192.168.0.1>;reason=unknown
{noformat}

Comments: By: Matt Jordan (mjordan) 2014-12-02 13:43:42.613-0600

I don't think this is a bug.

Asterisk is not a telephone. Hence, it placing the {{user=phone}} specifier in the SIP URI for the mentioned fields would be incorrect. The fact that your provider wants to pretend like Asterisk is a telephone is a limitation of your provider.

Someone may want to add this functionality as an improvement (and it would have to be configurable, as - again - Asterisk is not necessarily providing telephone numbers here in the mentioned headers). That being said, without a patch, I don't think this issue will remain open.
By: Rusty Newton (rnewton) 2014-12-03 17:40:04.933-0600

Yup I don't see how this is a bug. Closing out as "Not a Bug". If someone decides to write a feature for this then they should open a new issue.
By: Mark Petersen (asterisk.org@zombie.dk) 2014-12-04 02:47:02.452-0600

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
By: Matt Jordan (mjordan) 2014-12-04 09:26:11.374-0600

The RFCs aren't super explicit about the usage of {{user=phone}}, and certainly don't mandate its usage. The fact that your provider is requiring it is ... odd.

That being said, after talking with a few other folks, if the user portion of a SIP URI is *known* to map to a PSTN DID, then you can certainly include {{user=phone}}.

I'm not sure how you're going to be able to make this determination for all of the fields you mention, but if you have a patch in mind, we can certainly entertain it. Re-opening.
By: Mark Petersen (asterisk.org@zombie.dk) 2014-12-04 16:15:21.111-0600

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@example.com;user=phone>
{noformat}
By: Fazil (fas143) 2015-01-20 02:07:53.687-0600

Mark peterson:

Am facing this same issue. Need user=phone in From header in invite. This is hardly requested by provider
By: Mark Petersen (asterisk.org@zombie.dk) 2015-05-19 04:05:38.168-0500

patch that add missing urioptions to from headder
By: Mark Petersen (asterisk.org@zombie.dk) 2015-05-19 04:25:55.408-0500

ups fixed a small bug, we will now generate the from header like this
<sip:+14157676791@example.com;user=phone>;tag=23325dasd
By: Fazil (fas143) 2015-05-19 11:13:42.386-0500

Thanks you Mark Petersen. I will try it on tomorrow and will update.
By: Mark Petersen (asterisk.org@zombie.dk) 2015-05-20 06:10:05.646-0500

Patch is for Asterisk 11
By: Fazil (fas143) 2015-05-20 07:31:29.315-0500

Noted
By: Fazil (fas143) 2015-05-21 03:02:41.511-0500

Excellent its working fine

That provider also request for with Media Connection information

These are sip debugs from their end server
{noformat}
----------------------------------------------------------------------------Sip debugs from their end by an out going call from our asterisk server-----------------------
=== Session Description Protocol ===
Protocol Version : v=0
Session Owner/Creator and session identi : o=root 377537064 377537064 IN IP4 10.100.20.10
Session Name : s=Asterisk PBX 1.8.20.0
Session Connection Information : c=IN IP4 10.100.20.10
Time the session is active : t=0 0
Media Name : m=audio 14830 RTP/AVP 8 101
Media : audio
Port : 14830
Protocol type : RTP/AVP
Media Format : 8 = PCMA
Media Format : 101 = dynamic
Media Attribute : a=rtpmap:8 PCMA/8000
Media Attribute : a=rtpmap:101 telephone-event/8000
Media Attribute : a=fmtp:101 0-16
Media Attribute : a=ptime:20
Media Attribute : a=sendrecv
---------------------------------------------------Sip debugs from their end by an out going call from an server what they exepect-----------------------
Protocol Version : v=0
Session Owner/Creator and session identi : o=CiscoSystemsSIP-GW-UserAgent 6070 8175 IN IP4 178.16.8.7
Session Name : s=SIP Call
Session Connection Information : c=IN IP4 178.16.8.7
Time the session is active : t=0 0
Media Name : m=audio 32706 RTP/AVP 8 101
Media : audio
Port : 32706
Protocol type : RTP/AVP
Media Format : 8 = PCMA
Media Format : 101 = dynamic
Media Connection information : c=IN IP4 178.16.8.7 <- with Media Connection information
Media Attribute : a=rtpmap:8 PCMA/8000
Media Attribute : a=rtpmap:101 telephone-event/8000
Media Attribute : a=fmtp:101 0-15
Media Attribute : a=ptime:20

------------------------------------------------------------------------------------------------------------
{noformat}
Sip debug from our server as follows
{noformat}
NVITE sip:0553625772@172.16.84.6;user=phone SIP/2.0
Via: SIP/2.0/UDP 10.100.20.10:5060;branch=z9hG4bK3b61bdfd
Max-Forwards: 70
From: <sip:576981000@10.100.20.10;user=phone>;tag=as2f333a3c
To: <sip:0553625772@172.16.84.6;user=phone>
Contact: <sip:576981000@10.100.20.10:5060>
Call-ID: 65161339488954b43193f8562f8ac7d4@10.100.20.10:5060
CSeq: 102 INVITE
User-Agent: FPBX-2.8.1(1.8.20.0)
Date: Wed, 21 Jan 2015 09:32:12 GMT
Session-Expires: 1800
Min-SE: 1800
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
Allow-Events: telephone-event
Content-Disposition: session;handling=required
P-Asserted-Identity: <sip:576981000@10.100.20.10;user=phone>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 576210786 576210786 IN IP4 10.100.20.10
s=Asterisk PBX 1.8.20.0
c=IN IP4 10.100.20.10
t=0 0
m=audio 15996 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv
{noformat}