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Summary:ASTERISK-26902: chan_sip: Unnecessary changes to Domain Part of Request URI when setting User part
Reporter:Leo Brown (netfuse)Labels:
Date Opened:2017-03-29 04:13:00Date Closed:
Priority:MinorRegression?No
Status:Open/NewComponents:Channels/chan_sip/Interoperability
Versions:14.3.0 Frequency of
Occurrence		Constant
Related
Issues:
Environment:CentOS6.7 x86_x64Attachments:
Description:Lets say we have a registered SIP peer, sip_123.

{{*CLI> sip show peer sip_123}}

{{Name       : sip_123}}
{{Addr->IP     : 195.95.76.61:5060}}
{{Reg. Contact : sip:sip_123@10.1.10.51:54547}}

As we can see the location is our serving proxy 195.95.76.61. If we dial out to this peer like:
{{Dial(SIP/sip_123)}}

We then correctly have an INVITE sent to 195.95.76.61:5060 which looks like:
{{INVITE sip:sip_123@10.1.10.51:54547}}

All is well. However, if we dial specifying a username part like:
{{Dial(SIP/sip_123/567)}}

Our INVITE looks like this:
{{INVITE 567@195.95.76.61:5060}}

That is to say, the contact domain and port have been lost by virtue of setting the user part of the dialled contact. The user part has been changed correctly.

If it were possible to collect all this data from Asterisk and construct a the INVITE based on the correct fields, that would be fine. However, as far as I understand there is no way to separately specify the Request URI and target IP address for a call, the assumption is that these are bound up in chan_sip’s processing.

I am sure PJSip contains more granularity over this but it’s simply not possible to switch out for PJSip at this time due to the number of bindings and new interop challenges.

I'm also aware that it's possible to change the To headers by using exclamation mark syntax in the dial string, however it is the Request URI which needs to be correct in this case.
Comments:By: Asterisk Team (asteriskteam) 2017-03-29 04:13:03.683-0500

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].