Regarding "sipsak" For AstLinux we have added sipsak to the SVN and is due to be official in the next AstLinux release, 1.2.4 . Thanks to nudging from Michael :-) -- # sipsak -V sipsak 0.9.6 by Nils Ohlmeier Copyright (C) 2002-2004 FhG Fokus Copyright (C) 2004-2005 Nils Ohlmeier compiled with DEFAULT_TIMEOUT=150, FQDN_SIZE=65, RAW_SUPPORT, LONG_OPTS, OPENSSL_MD5, OPENSSL_SHA1, STR_CASE_INSENSITIVE, CMP_CASE_INSENSITIVE --
Kudos the the sipsak authors, 10 years since last release, uses autoconf and still compiles clean today. This is unusual ! And only 61 KB in size. Lonnie On Sep 15, 2015, at 12:49 PM, The Cadillac Kid <eldorado...@yahoo.com> wrote: > will your service provider accept sip OPTIONS? (same as what setting > qualify=yes) in asterisk does for the PEER... > > I dont know if astlinux inscludes sipsak but on my large servers i use > sipsak and a script to test both a provider and my asterisk.. (we have had > SIP deadlock issues at one site).. OPTIONS would at least let you know if 2 > way sip traffic is progressing to and from the provider > -Christopher > > From: Shamus Rask <sha...@srask.ca> > To: astlinux-users@lists.sourceforge.net > Sent: Tuesday, September 15, 2015 10:40 AM > Subject: [Astlinux-users] how to monitor outbound trunks > > I’m posting this here as I find the AstLinux users offer the best and most > responsive insights to all things Asterisk. > > I’ve started a SaaS-based project that uses Asterisk to make outbound calls. > I’d like to implement some form of rigorous monitoring to ensure that my 2x > outbound SIP trunks are able to dial numbers on the PSTN. > > I’ve created a cron job on my Asterisk server that generates a .call file > every 10 minutes. This dials a separate DID (hosted by a different SIP > provider) on my personal AstLinux box. By doing an Answer() immediately > followed by a Hangup(), I’ve limited the duration of the call to < 1s. > However, I’m still being charged $0.005 for each call by my SIP provider. At > 2x calls every 10m, this adds up to $1.44/day or $43.20/month. > > Given the SIP signalling that is taking place, is there any way to parse the > SIP messages and terminate the call attempt when the Asterisk server sees a > “180 - ringing” message? This would eliminate the daily cost of the tests and > I would only incur the monthly DID charge. > > Thank you in advance, > Shamus > ------------------------------------------------------------------------------ > _______________________________________________ > Astlinux-users mailing list > Astlinux-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pay...@krisk.org. > > ------------------------------------------------------------------------------ > _______________________________________________ > Astlinux-users mailing list > Astlinux-users@lists.sourceforge.net > https://lists.sourceforge.net/lists/listinfo/astlinux-users > > Donations to support AstLinux are graciously accepted via PayPal to > pay...@krisk.org. ------------------------------------------------------------------------------ Monitor Your Dynamic Infrastructure at Any Scale With Datadog! Get real-time metrics from all of your servers, apps and tools in one place. SourceForge users - Click here to start your Free Trial of Datadog now! http://pubads.g.doubleclick.net/gampad/clk?id=241902991&iu=/4140 _______________________________________________ Astlinux-users mailing list Astlinux-users@lists.sourceforge.net https://lists.sourceforge.net/lists/listinfo/astlinux-users Donations to support AstLinux are graciously accepted via PayPal to pay...@krisk.org.
