<html><body><div id="zimbraEditorContainer" style="font-family: arial, helvetica, sans-serif; font-size: 10pt; color: #000000" class="2"><div></div><div data-marker="__SIG_PRE__"></div><div data-marker="__HEADERS__"><blockquote style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;" data-mce-style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><b>From: </b>"John Hughes" <john@calva.com><br><b>To: </b>"Asterisk Users Mailing List, Non-Commercial Discussion" <asterisk-users@lists.digium.com><br><b>Sent: </b>Thursday, May 14, 2020 2:10:45 AM<br><b>Subject: </b>[asterisk-users] I can do alaw, ulaw and gsm; remote can do g729 and alaw; asterisk wants to translate g729 -> alaw. WHY?<br></blockquote></div><div data-marker="__QUOTED_TEXT__"><blockquote style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;" data-mce-style="border-left: 2px solid #1010FF; margin-left: 5px; padding-left: 5px; color: #000; font-weight: normal; font-style: normal; text-decoration: none; font-family: Helvetica,Arial,sans-serif; font-size: 12pt;"><p>I am having a problem with one of my callers who is using either g729 or alaw. I can do alaw but not g729 so asterisk should negotiate alaw right? In fact from the sip debug it looks like it does, but then I get the dreaded "channel.c:5630 set_format: Unable to find a codec translation path: (g729) -> (alaw)" and the call hangs up. Why?</p><p>Last minute thought: Is it possible that the caller is sending g729 in RTP even though the SIP negotiation clearly chooses alaw? Maybe I need some RTP debugging.<br></p><p>Asterisk 13.14.1 on Debian, using chan_sip.</p></blockquote><div>Hi John,</div><div><br data-mce-bogus="1"></div><div>Maybe a newer version of Asterisk would help? The latest release for 13 is version 13.33. The version you are on was released 3 years ago.</div><div><br></div><div>Here is an issue which looks like what you describe and was fixed in 13.16</div><div><a href="https://issues.asterisk.org/jira/browse/ASTERISK-26143" data-mce-href="https://issues.asterisk.org/jira/browse/ASTERISK-26143">https://issues.asterisk.org/jira/browse/ASTERISK-26143</a></div><div><br data-mce-bogus="1"></div><div>Not sure if this is the answer to your problem but thought that I would throw that out there.</div><div><br data-mce-bogus="1"></div><div>Michael L. Young</div><div><br data-mce-bogus="1"></div><div>(elguero)</div></div></div></body></html>