Summary:	ASTERISK-02214: [patch] Add ";user=phone" when INVITE contain only phone number
Reporter:	fwittekind (fwittekind)	Labels:
Date Opened:	2004-08-12 14:02:47	Date Closed:	2008-01-15 15:15:28.000-0600
Priority:	Major	Regression?	No
Status:	Closed/Complete	Components:	Core/General
Versions:		Frequency of Occurrence
Related Issues:
Environment:		Attachments:	( 0) asterisk-1.0-RC2-chan_sip-userphone.patch ( 1) userphone.txt ( 2) userphone2.txt
Description:	Old description: add ;user=authname to SIP INVITEs   This was required for making outbound calls to PSTN gw service running Broadworks software. New description: For some services, we need to indicate that the user part of the URI we're calling is a phone number by adding ;user=phone to the URI. ****** ADDITIONAL INFORMATION ****** From the specs: "   To accomodate telephone addressing, the SIP specification includes a   provision to incorporate a tel: URI [4] telephone-subscriber   (everything following the tel: prefix) directly into the user part of   a sip: or sips: URI, by setting the "user" parameter to "phone"." and "The tel: URI telephone-subscriber can be either a global-number or a   local-number."
Comments:	By: Olle Johansson (oej) 2004-08-12 14:56:21 Can you explain more, give any references to why the require this? I need more information. user= is normally a hint if this a phone. Is there any reason to do this by default, as in your patch, or make it an option. Let's do some research on how to do this right! :-) By: fwittekind (fwittekind) 2004-08-12 16:06:16 http://www.voip-info.org/tiki-view_cache.php?url=http%3A%2F%2Fftp.isi.edu%2Fin-notes%2Frfc3261.txt Found on page 152, 19.1.3 Example SIP and SIPS URIs sip:+1-212-555-1212:1234@gateway.com;user=phone Found on page 222, 25.1 Basic Rules user-param        =  "user=" ( "phone" / "ip" / other-user) By: Olle Johansson (oej) 2004-08-12 16:11:40 user=phone is documented in several places, but I can't find anything on user equals something else... By: Olle Johansson (oej) 2004-08-12 16:17:26 For the record, since this may be useful another time, there's a clarification on "user=phone" in http://www.softarmor.com/wgdb/docs/draft-mahy-sipping-user-equals-phone-00.txt Still no mention of user=<somethingelse> anywhere except possibly Brian Rosen's proposal for user=dialstring By: Mark Spencer (markster) 2004-08-12 19:28:55 So what's the story here?  Maybe yet another option? By: Olle Johansson (oej) 2004-08-13 01:51:03 Markster, we are still trying to find out what the Broadworks stuff requires and the usage of the user= header. As soon as we know more, we'll come up with a proposal. Stay tuned. My guess is that we can add ";user=phone" when the username really is a phone number or a digit-only extension (DTMF "digit", including *#). We should not add it always, as in the patch. By: fwittekind (fwittekind) 2004-08-13 13:05:21 ";user=phone" is good enough for broadworks to accept. I uploaded a new version, that only adds ";user=phone" iff the username portion of the SIP URI matches a phone number or a digit-only extension. Should it also be a option in sip.conf to turn it on or off? By: Olle Johansson (oej) 2004-08-14 06:54:02 From IETF proceedings: "  agreed: the presence of the user=phone parameter implies that the user part conforms to the specification of the tel: URI." Also, some SIP user names are numbers-only without being a tel: uri phone number. To do this right, this would have to be an option to DIAL. We can however make it an option for a peer, saying that "everything we send to this peer is a phone number, since it's my PSTN gateway (provided that it consists of digits only, and DTMF allowed characters)" Conclusion: I would add a peer option to your patch, enabling it only for selected peers. Otherwise we could potentially cause problems. By: Olle Johansson (oej) 2004-08-14 07:54:34 Modified patch * Allows + as first character according to Tel uri: rfc * Adds config option for [general] and [peer]  usereqphone = yes | no Yes means that *if* the username part is a valid tel uri, we add ;user=phone to the uri before sending it to the proxy. Use this for proxies (pstn providers and gateways) if it's required by the provider. Fwittekind: Please test and report if this works for you. By: fwittekind (fwittekind) 2004-08-16 14:02:47 sip debug shows ;user=phone missing from first INVITE packet, yet sip show peer shows UserEqPhone  : Yes Minor bug fix required, added r->usereqphone = p->usereqphone; to create_addr function. By: Olle Johansson (oej) 2004-08-16 14:53:30 Great. Added userphone2.txt includes your patch to my patch to your patch, as well as the changes to sip.conf.sample. Ready for Mark and possibly CVS integration. Fwittekind: Thank you for working with us to solve this! By: Mark Spencer (markster) 2004-08-31 15:19:38 Did the recent SIP contact changes help this in any way? By: Olle Johansson (oej) 2004-08-31 15:21:32 Mark, they're not related. By: Olle Johansson (oej) 2004-09-05 13:56:23 Mark, find me on the IRC to discuss this patch. By: Mark Spencer (markster) 2004-10-14 10:49:23 Okay, your turn, you find me! By: Olle Johansson (oej) 2004-10-15 02:34:17 ...will do... By: twisted (twisted) 2004-10-27 17:16:43 How's the hide-and-seek game going? By: Olle Johansson (oej) 2004-11-11 15:31:57.000-0600 Ping, Markster. If you need to discuss this more, find me on IRC By: Olle Johansson (oej) 2004-11-14 02:56:09.000-0600 Keep-alive-bug-tracker-packet /O :-) By: Olle Johansson (oej) 2004-11-21 04:17:08.000-0600 Updated patch to current CVS head By: Mark Spencer (markster) 2004-12-02 18:31:05.000-0600 Added to CVS, thanks olle! By: Russell Bryant (russell) 2004-12-02 19:13:39.000-0600 not in 1.0 By: Digium Subversion (svnbot) 2008-01-15 15:15:28.000-0600 Repository: asterisk Revision: 4372 U   trunk/channels/chan_sip.c U   trunk/configs/sip.conf.sample ------------------------------------------------------------------------ r4372 | markster | 2008-01-15 15:15:28 -0600 (Tue, 15 Jan 2008) | 2 lines Add user=phone option (bug ASTERISK-2214, thanks oej) ------------------------------------------------------------------------ http://svn.digium.com/view/asterisk?view=rev&revision=4372