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Summary:ASTERISK-28071: chan_sip: ignores the "fromdomain" option if "fromuser" option is presented
Reporter:Aliaksandr Hrechny (ahrechny)Labels:
Date Opened:2018-09-25 04:03:25Date Closed:2018-09-26 02:31:37
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Registration
Versions:13.23.1 Frequency of
Occurrence
Related
Issues:
Environment:CentOS Linux release 7.5.1804 (Core)Attachments:
Description:My sip provider requires domain for a registration. But I faced with issue when `fromdomain` option is ignored if `fromuser` option is presented.
For example with the following configuration
{noformat}
username=123456789012
type=peer
transport=udp
secret=sip-secret
qualify=yes
outboundproxy=sip.server:5060
insecure=invite,port
host=sip.server
fromdomain=10.10.10.10
disallow=all
context=from-trunk-sip-custom
allow=alaw
{noformat}
result is next:
{noformat}
To: <sip:sip.server>
From: "Unknown"<sip:Unknown@10.10.10.10>
{noformat}

But if I add to the config `fromuser=+123456789012` the result is as follows:
{noformat}
From: "Unknown"<sip:+123456789012@sip.server>
To: <sip:+123456789012@sip.server>
{noformat}
Comments:By: Asterisk Team (asteriskteam) 2018-09-25 04:03:28.776-0500

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].
By: Joshua C. Colp (jcolp) 2018-09-25 11:54:28.816-0500

Thank you for taking the time to report this bug and helping to make Asterisk better. Unfortunately, we cannot work on this bug because your description did not include enough information. Please read over the Asterisk Issue Guidelines [1] which discusses the information necessary for your issue to be resolved and the format that information needs to be in. We would be grateful if you would then provide a more complete description of the problem. At a minimum, we need:

1. The specific steps or actions you took that caused you to encounter the problem.
2. The behavior you expected and the location of documentation that led you to that expectation.
3. The behavior you actually encountered.

To demonstrate the issue in detail, please include Asterisk log files generated per the instructions on the wiki [2]. If applicable, please ensure that protocol-level trace debugging is enabled, e.g., 'sip set debug on' if the issue involves chan_sip, and configuration information such as dialplan and channel configuration.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

[2] https://wiki.asterisk.org/wiki/display/AST/Collecting+Debug+Information

For example you state this is for registration but your configuration is incomplete. As well complete console output is needed.

I should also state that chan_sip is community supported, so there is no timeframe on if or when someone may look at this if it is accepted.