<div dir="ltr">Another thing is storage... if your DB structure is assuming UUIDs are of the normal length, then longer Call-IDs are going to cause you truncation problems.</div><div class="gmail_extra"><br><br><div class="gmail_quote">
On 28 August 2013 20:14, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div dir="ltr">Correct, you are at the mercy of the call-id being unique and uniqueness cannot be guaranteed.<div>Many sip endpoints at least try pretty hard to make sure they are unique but there is really nothing to enforce it.</div>
<div><br></div><div>Alternatively, you do have the sip_call_id variable present so its not that hard to correlate them.</div><div><div><br></div></div></div><div class="gmail_extra"><div><div class="h5"><br><br><div class="gmail_quote">
On Wed, Aug 28, 2013 at 1:40 PM, Ben Langfeld <span dir="ltr"><<a href="mailto:ben@langfeld.co.uk" target="_blank">ben@langfeld.co.uk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">I would imagine the issue is that an incoming INVITE CallID might not strictly be unique. I'm not sure if "<span style="font-family:arial,sans-serif;font-size:13px">use-callid-as-uuid" deals with this at all...</span></div>
<div><div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On 28 August 2013 06:51, Kim Rostgaard Christensen <span dir="ltr"><<a href="mailto:krc@retrospekt.dk" target="_blank">krc@retrospekt.dk</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">-----BEGIN PGP SIGNED MESSAGE-----<br>
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<br>
Hi list<br>
<br>
First post, so I would just like to give a big hat tip to the people<br>
involved - extremely nice project. Well documented, nice community<br>
We recently moved from Asterisk to FreeSWITCH, and have not looked<br>
back since. Asterisk is a very nice project, but AMI is less suited<br>
our needs than ESL.<br>
<br>
Today, while mucking about with PJSUA, I discovered the call_id to be<br>
(entirely) different from the uuid assigned to the channel in FS.<br>
While this appears to remedied by the *-use-callid-as-uuid options in<br>
sofia.xml I still wonder if there are any potential<br>
complications/slowdown associated with using the option?<br>
<br>
Best<br>
- --<br>
Kim Rostgaard Christensen | <a href="mailto:krc@retrospekt.dk" target="_blank">krc@retrospekt.dk</a> | <a href="http://retrospekt.dk" target="_blank">http://retrospekt.dk</a><br>
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