This will help...

-- sipsak, version 0.9.6-1, new package known as SIP-swiss-army-knife.
   More Info: https://web.archive.org/web/20130728223800/http://sipsak.org/   
(via the Wayback Machine)
   More Info: http://linux.die.net/man/1/sipsak


Lonnie


On Sep 18, 2015, at 4:53 PM, Michael Knill <michael.kn...@ipcsolutions.com.au> 
wrote:

> Cool. I can’t access their MAN page though!
> 
> Regards
> Michael Knill
> 
> 
> 
> 
> On 17 Sep 2015, at 9:20 am, Lonnie Abelbeck <li...@lonnie.abelbeck.com> wrote:
> 
> 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
>> ------------------------------------------------------------------------------
>> _______________________________________________
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>> 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.
>> 
>> ------------------------------------------------------------------------------
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>> 
>> Donations to support AstLinux are graciously accepted via PayPal to 
>> pay...@krisk.org.
> 
> 
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> 
> 
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> pay...@krisk.org.
> 
> 


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