# Sip usereqphone = yes does not add user=phone in From header

**URL:** <https://community.asterisk.org/t/sip-usereqphone-yes-does-not-add-user-phone-in-from-header/44177>\
**Category:** Asterisk General\
**Created:** [October 14, 2013, 7:44pm UTC](https://community.asterisk.org/t/sip-usereqphone-yes-does-not-add-user-phone-in-from-header/44177 "2013-10-14T19:44:01Z")\
**Posts on this page:** 1\
**Showing post:** 4

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**Author:** ![Triki](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/triki/32/2688_2.png) [@Triki](https://community.asterisk.org/u/Triki)\
**Post date:** [November 30, 2017, 1:11pm UTC](https://community.asterisk.org/t/sip-usereqphone-yes-does-not-add-user-phone-in-from-header/44177/4 "2017-11-30T13:11:23Z")

</div>

There is a workaround until “usereqphone” variable is available for From header.

You can use fromdomain variable on your sip\*.conf.

For instance:

fromuser=yourSIPuser  
[fromdomain=yourdomain.com](http://fromdomain=yourdomain.com);user=phone

If you find something like this:

From: sip:yourSIPuser@yourdomain.com;user=phone:5160;tag=as695bc59b

You’ll have to change your Asterisk listening port for your trunk and for your extensions.

Now:  
Trunk 5060  
Extensions 5160

Trunk:  
Edit your file for “bind” variable (in my case: pjsip.transports.conf)  
Change bind=0.0.0.0:5060 to bind=0.0.0.0:5160

Extensions:  
Edit your file for “bindport” variable (in my case: sip\_general\_additional.conf)  
Change bindport=5160 to bindport=5060

Don’t forget to REGISTER your UA to new Extensions listening port (5060) and to reload these files

Now new From header will be as this (std.5060 port is avoided as is considered a well known port):

From: sip:yourSIPuser@yourdomain.com;user=phone;tag=as695bc59b

Hope to be useful

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_[View the full topic](https://community.asterisk.org/t/sip-usereqphone-yes-does-not-add-user-phone-in-from-header/44177)._