| Summary: | ASTERISK-26957: manipulating ToHeader number with CallerID(DNID) | ||
| Reporter: | Yasin CANER (ycaner) | Labels: | fax pjsip |
| Date Opened: | 2017-04-24 08:59:26 | Date Closed: | 2018-07-11 04:37:43 |
| Priority: | Minor | Regression? | |
| Status: | Closed/Complete | Components: | Applications/app_dial Resources/res_pjsip_session |
| Versions: | 13.15.0 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | Centos 6.7, Asterisk 13.5 | Attachments: | ( 0) Asterik_13_Toph.jpg |
| Description: | Hello all,
Some kind of devices like FXS needs to wait Request Uri Number and ToHeader number must be same.if not , it declines the calls. in chan_sip, we can solve this problem with exclamation mark in Dial app but chan_pjsip doesnt have ability. So i added a function to set toheader with setting Callerid(DNID) variable in res_pjsip_session.c My toph {noformat} A caller ---> Asterisk-Trunk-1 || Asterisk-Trunk-2 --> Kamailio --> B callee INVITE --> Ruri : 08503027337 From : 08503023423 To : 08503027337 ---> Asterisk Trunk -1 --> Asterisk Trunk -2 ---> Ruri : 56428503027337 From : 908503023423 To : 8503027337----> Kamailio(carrier module) ----> Ruri : 8503027337 From : 908503023423 To : 8503023423 {noformat} How is it work. Before dial application , setting Callerid(DNID) in dialplan , it effects. i tested function as below 1- 2 side PJSIP_SEND_SESSION_REFRESH(), it send re-invite to outgoing and incoming channels. No problem 2- Fax Re-invite, it send so many re-invite to negotiation for fax. No problem 3- Standard Call . No Problem 4- without setting Callerid(DNID) . No problem ,there isnt null pointer exception or crash 5- Setting 2 side Callerid(DNID) . there isnt problem. 6- chan_pjsip , CDR is checked , no problem. 7- chan_sip , it tested with or without Callerid(DNID) setting. there is not side effect. No problem. it didn't test on chan_sip Forum topic : https://community.asterisk.org/t/way-to-get-toheader-name-or-number/68717/6?u=ycaner commit is updated for format and fixed memory leak. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2017-04-24 08:59:27.764-0500 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: Yasin CANER (ycaner) 2017-04-24 09:20:07.346-0500 logs are removed. By: Yasin CANER (ycaner) 2017-04-24 09:21:28.531-0500 Here is topology By: Yasin CANER (ycaner) 2017-04-27 01:05:12.275-0500 i pushed the patch to gerrit By: Yasin CANER (ycaner) 2017-05-03 02:28:24.808-0500 do i need to do smth for appling ? By: Joshua C. Colp (jcolp) 2017-05-03 04:31:33.349-0500 It is in queue to get reviewed. By: Joshua C. Colp (jcolp) 2017-12-19 07:55:28.813-0600 Did you ever end up pursuing a different change based on what Richard said? asterisk-dev list thread: http://lists.digium.com/pipermail/asterisk-dev/2017-May/076322.html By: Joshua C. Colp (jcolp) 2018-07-11 04:37:43.582-0500 Suspended due to lack of activity. This issue will be automatically re-opened if the reporter posts a comment. If you are not the reporter and would like this re-opened please create a new issue instead. If the new issue is related to this one a link will be created during the triage process. Further information on issue tracker usage can be found in the Asterisk Issue Guidlines [1]. [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines | ||
