If you add qualify=yes to the setting in sip.conf it will send a sip message to the peer every 60 seconds to check if it is alive. If you try to make a call while the peer is not alive it will fail immediatly rather than the caller hearing silence while your box waits for a reply timeout.
Andy Beak wrote: > Hi, > > No that is the correct address. I know it is an internal IP. > > We have our machine hosted in racks at our SIP providers data center. > > They've patched a new port to our cabinet and linked that to a gateway > (172.28.20.105). > > As long as we use that gateway (and the IP address they assigned to us) > our traffic will reach their SBC. > > I've confirmed that traceroute follows the path it is supposed to: > > traceroute to 192.168.34.1 (192.168.34.1), 30 hops max, 60 byte packets > 1 192.168.0.1 (192.168.0.1) 0.656 ms 0.562 ms 0.501 ms > 2 172.28.20.105 (172.28.20.105) 1.211 ms 1.209 ms 1.196 ms > 3 192.168.34.5 (192.168.34.5) 23.270 ms 23.269 ms 23.328 ms > 4 * * * > 5 * * * > 6 * * *^C > > Is there a way to test in Asterisk if it is able to reach a particular > IP address? Do you think that there is a routing problem here? > > Thanks, > Andy > > > > > On 20/07/2010 04:58 PM, Zeeshan Zakaria wrote: >> >> This "host=192.168.34.1" is where you'll put your provider's IP >> address. Currently you are using some local address which is not your >> provider's IP address. Where did you get it from? Call your providrr >> and ask them the IP address of the server where you'll be sending your >> calls. >> >> Zeeshan A Zakaria >> >> -- >> www.ilovetovoip.com <http://www.ilovetovoip.com> >> >>> On 2010-07-20 10:27 AM, "Andy Beak" <[email protected] >>> <mailto:[email protected]>> wrote: >>> >>> Hi, >>> >>> I set my list to subscribe to digest and I can't see how to reply to >>> your reply without starting a new thread. >>> >>> There is no need for SIP username and password because the provider >>> authenticates me on my IP address. >>> >>> I thought that "host=192.168.34.1" would be the sip provider IP address. >>> >>> At this point I don't need to accept incoming calls or place >>> VOIP-to-VOIP. All I need to do is connect to their PBX to place a >>> call to a cellphone. >>> >>> I reread all the documentation I could find and couldn't see where >>> else in sip.conf I should set the provider IP. >>> >>> Thanks for your reply, >>> Andy >>> >>> >>> >>> > In your sip.conf, there is no mention of your sip provider's IP >>> address, username and secret (pa... >>> >>> www.ilovetovoip.com <http://www.ilovetovoip.com> >>> <http://www.ilovetovoip.com> >>> >>> >>> >>> > On 2010-07-20 5:09 AM, "Andy Beak" <andr...@xxxxxxxxxxxxxxx >>> <mailto:andr...@xxxxxxxxxxxxxxx <mailto:andr...@xxxxxxxxxxxxxxx>>> wr... >>> >>> -- >>> _____________________________________________________________________ >>> -- Bandwidth and Colocation Provided byhttp://www.api-digital.com >>> <http://www.api-digital.com> -- >>> >>> >>> New to Asterisk? Join us for a live introductory webinar every Thurs: >>> http://www.aste... >>> >>> >>> -- >>> _____________________________________________________________________ >>> -- 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 >>> To UNSUBSCRIBE or update options visit: >>> http://lists.digium.com/mailman/listinfo/asterisk-users > -- _____________________________________________________________________ -- 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 To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
