| Summary: | ASTERISK-26597: func_pjsip_contact: PJSIP_CONTACT function references MD5ed AOR contact | ||
| Reporter: | sharan h (sharkbitz) | Labels: | |
| Date Opened: | 2016-11-14 23:07:28.000-0600 | Date Closed: | |
| Priority: | Minor | Regression? | |
| Status: | Open/New | Components: | Channels/chan_pjsip |
| Versions: | 13.12.2 | Frequency of Occurrence | |
| Related Issues: | |||
| Environment: | n/a | Attachments: | |
| Description: | In reference to:
https://community.asterisk.org/t/cant-get-pjsip-contact-function-to-work/68740/2 I am requesting a change in the PJSIP_CONTACT function to reference the actual contact from the aor. Currently, using the PJSIP_CONTACT function requires the use of the PJSIP_AOR function to return the exact dynamic URI. It's not intuitive, and the documentation doesn't clarify how to use the function clearly. | ||
| Comments: | By: Asterisk Team (asteriskteam) 2016-11-14 23:07:29.519-0600 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: Luke Escude (lukeescude) 2022-06-13 15:52:29.051-0500 Bumping this - The PJSIP_CONTACT is fairly useless unless you use PJSIP_AOR, but this isn't easy if an AOR has multiple Contacts. By: Niklas Larsson (pnlarsson) 2023-01-17 05:20:22.777-0600 Bumping this as well - doing an ugly workaround of setting the hashes as "static" variables and checking status on the contact before dialing (and failing to a Reachable contact) We use one AOR with multiple contacts and dialing using: {noformat} Dial(PJSIP/siptrunk/sip:${DIALNUMBER}@X.X.X.X:5060,${DIALTIME},${PSTN_DIALOPTS}) {noformat} And to get some kind of working failover, is to look at {noformat}${PJSIP_CONTACT(${SIPTRUNK_X_X_X_X_HASH},status)}{noformat} before trying to dial | ||