Summary:	ASTERISK-30266: Voicemail aborts when playing caller ID that includes + before digits
Reporter:	Misha Vodsedalek (vmisha)	Labels:	patch
Date Opened:	2022-10-17 16:45:42	Date Closed:	2022-10-17 17:24:12
Priority:	Minor	Regression?
Status:	Closed/Complete	Components:	Applications/app_voicemail
Versions:	18.14.0	Frequency of Occurrence
Related Issues:	duplicates ASTERISK-30248 ast_get_digit_str adds bogus initial delimiter if first character not to be spoken
Environment:		Attachments:	( 0) ASTERISK-30266.patch
Description:	From a SIP trunk provider, we receive caller ID that includes + before digits in the caller ID number.  The caller leaves a voicemail message.  In the msgXXXX.txt file, the caller ID field looks something like callerid="Name of the caller" <+16135551234> The mailbox option saycid for the extension with the mailbox is set to saycid=yes When the user attempts to listen to the message, the session is aborted and the following message are logged: File  does not exist in any format Unable to open  (format (ulaw|alaw|g729)): No such file or directory I tracked the problem to say.c line 339.  Here, the assumption is made that the first character must be a valid digit.  When it is the plus sign, & is put at the beginning of the filenames string and that causes the empty file name issue logged during this scenario. The attached patch solves the issue.
Comments:	By: Asterisk Team (asteriskteam) 2022-10-17 16:45:47.961-0500 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. Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/]. By: Joshua C. Colp (jcolp) 2022-10-17 17:24:12.089-0500 This is already being tracked at ASTERISK-30248 and a patch is already up for review.