| Summary: | ASTERISK-28977: PJSIP can't SET CallerId | ||
| Reporter: | Michal Paplaczyk (paplaczyk) | Labels: | |
| Date Opened: | 2020-07-03 12:40:58 | Date Closed: | 2020-07-03 12:51:27 |
| Priority: | Trivial | Regression? | |
| Status: | Closed/Complete | Components: | Resources/res_pjsip |
| Versions: | 16.11.0 | Frequency of Occurrence | Constant |
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Hello
I have a problem in changing CallerID if I use PJSIP protocol. Belowe is my current configuration. I tried to use another versions of assterisk together with FreePBX and problem was the same. I think this is a protocol problem. Every time when i set callerid like: Set(CALLERID(num)=xxxxxxxxxx) i expect in INVITE in Header FROM: From: "xxxxxxxxxx" <sip:xxxxxxxxxxxx@xxx.xxx.xxx.xxx>;tag=c76d13d2 but i found From: <sip:xxxxxxxxxxxx@xxx.xxx.xxx.xxx>;tag=c76d13d2 Could somebody can help me with this issue? [trunk-1] type=registration transport=transport-udp outbound_auth=trunk-1-auth server_uri=sip:xxxxxxxxxxx.xx:5060 client_uri=sip:xxxxxxxxx@xxxxxxxxx.xx:5060 contact_user=xxxxxxxx retry_interval=60 forbidden_retry_interval=600 expiration=3600 line=yes outbound_proxy=sip:xxxxxxxxxxx.xx\;lr endpoint=trunk-1 [trunk-1] type=endpoint transport=transport-udp context=from-abc disallow=all allow=alaw,ulaw,g729,gsm aors=trunk-1 outbound_auth=trunk-1-auth from_user=xxxxxxxxx from_domain=xxxxxxx.xx [trunk-1] type=aor contact=sip:xxxxxxxxxxx.xx:5060 outbound_proxy=sip:xxxxxxxxxxxx.xx\;lr [trunk-1] type=identify endpoint=trunk-1 match=xxxxxxxxxxxxx.xx [trunk-1-auth] type=auth auth_type=userpass username=xxxxxxxx password=xxxxxxxxx | ||
| Comments: | By: Asterisk Team (asteriskteam) 2020-07-03 12:40:59.412-0500 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. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed. 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]. Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur. By: Joshua C. Colp (jcolp) 2020-07-03 12:51:27.389-0500 This is not an issue in Asterisk. You have set the "from_user" option on the PJSIP endpoint which sets the user portion of the From. Setting it in the dialplan will not override it. If you wish to convey callerid you will need to send it another way, such as using P-Asserted-Identity or Remote-Party-ID both of which can be enabled on the endpoint. For further help please use the community forum[1]. [1] https://community.asterisk.org/ | ||