[asterisk-users] checking if a phone number is UP

hi, We have a phone number from third party provider which is used for inbound calls. How could I monitor if this *phone number* is reachable? the initial idea doesn't sound elegant: - on my SIP server I set couple seconds of ringing before Answer(). - the monitoring server calls to that phone

Re: [asterisk-users] checking if a phone number is UP

tickets on issues consistently before they know they have an issue. Bryant -- *From*: Aurimas Skirgaila a.skirga...@gmail.com *Sent*: Thursday, February 09, 2012 9:34 AM *To*: Asterisk Users Mailing List - Non-Commercial Discussion asterisk-users@lists.digium.com

Re: [asterisk-users] Is this doable?

To UNSUBSCRIBE or update options visit: http://lists.digium.com/**mailman/listinfo/asterisk-**usershttp://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http

[asterisk-users] AMD tweaking

, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list

Re: [asterisk-users] AMD tweaking

-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ On May 16, 2011, at 8:56 AM, Aurimas Skirgaila a.skirga...@gmail.com wrote: Hi, long time ago, I came up with an optimal configuration set for my environment - good detection and little false positives. Unfortunately some

Re: [asterisk-users] load balance with 2 wan connections

/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http

[asterisk-users] malformed SIP / routing issue

to build SIP header specifying the final destination - the internal IP address? -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory

Re: [asterisk-users] malformed SIP / routing issue

://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided

[asterisk-users] Licensing of Default MOH

Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? -- Mvh, Aurimas Skirgaila

Re: [asterisk-users] Licensing of Default MOH

, Aurimas Skirgaila wrote: Hi, I wonder if I may freely use the default soundfiles that came with asterisk (fpm-world-mix, fpm-calm-river and fpm-sunshine) on production server? Are there any official sources of royalty free music? http://downloads.asterisk.org/pub/telephony/sounds

Re: [asterisk-users] E1 check with nagios, how to?

/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org

Re: [asterisk-users] AMD message

: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation

Re: [asterisk-users] Tweaking AMD in Asterisk

/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org

Re: [asterisk-users] Tweaking AMD in Asterisk

-- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila

Re: [asterisk-users] identify caller hangup or callee hangup?

: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided

Re: [asterisk-users] is it possible to connect Digium TE420 and Cisco card?

Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but same errors persist. -- Mvh, Aurimas Skirgaila -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com

Re: [asterisk-users] SIP Dialplan Failover Solution

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[asterisk-users] is it possible to connect Digium TE420 and Cisco card?

any effect, bchan=1-15,17-31 dchan=16 [zapata.conf] group=1 pridialplan = unknown switchtype=euroisdn context = trunk-1 signalling = pri_net channel = 1-15,17-31 Hardware - Dell PowerEdge R200. Now moved onto barebone test server, but same errors persist. -- Mvh, Aurimas Skirgaila

Re: [asterisk-users] Restarting Asterisk using a script - Thanks to all -

. Regards, Am -- Jose P. Espinal http://www.eSlackware.com http://www.eslackware.com/ IRC: Khratos @ #asterisk / -doc / -bugs -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http

Re: [asterisk-users] configure the sound for inbound calls

and name of clients: For your information there is no problem related to the outbound call An help would be appreciated Kind Regards Salah. -- Mvh, Aurimas Skirgaila -- _ -- Bandwidth and Colocation Provided by http://www.api

Re: [asterisk-users] IAX2 trunking with Older Asterisk version ?

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Re: [asterisk-users] SIP over VPN

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Re: [asterisk-users] SIP Trunk groups

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[asterisk-users] Difference between Transfer and Dial applications

Hi, I wonder what is the difference between Transfer and Dial applications? Could somebody give me an example of Transfer usage? (documentation and voip-info looks poor a bit). I'm using Asterisk 1.2.5 if it matters. -- Mvh, Aurimas Skirgaila

Re: [asterisk-users] QoS VPN

.) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Mvh, Aurimas Skirgaila

Re: [asterisk-users] QoS VPN

On Fri, May 8, 2009 at 3:45 PM, Jeff LaCoursiere j...@jeff.net wrote: On Fri, 8 May 2009, Aurimas Skirgaila wrote: Despite the VPN overhead, running VOIP through VPN is good idea because VPN reorders encapsulated UDP packets in correct order. Security matters as well. Reorders? How