Attached is a sanitized configuration file that was sent to Ranga back in May/09 that works.
This is using user/auth, my apologies if I mislead somewhere on how it is configured. -----Original Message----- From: Michael Scheidell [mailto:[email protected]] Sent: Wednesday, February 10, 2010 7:09 PM To: Nathaniel Watkins; Todd Hodgen; 'Michael Scheidell' Cc: [email protected] Subject: RE: [sipx-users] voip.ms: tries to connect to random port Actually Todd says he figured it out. Once we get it documented, it will be fine. Unless voip.ms is scrdeey . I don't think their user/auth will ever work. They bind to the port that registers, not the one that does keepalives. -----Original Message----- From: Nathaniel Watkins <[email protected]> Sent: Wednesday, February 10, 2010 9:14 PM To: Todd Hodgen <[email protected]>; 'Michael Scheidell' <[email protected]> Cc: [email protected] <[email protected]> Subject: RE: [sipx-users] voip.ms: tries to connect to random port If you guys figure out what Todd is doing wrong, let me know - because I'm probably doing it wrong as well. Nathaniel Watkins IT Director Garrett County Government 316 East Alder Street, Suite 2 Oakland, MD 21550 Telephone: 301-334-5001 Fax: 301-334-5021 E-mail: [email protected]<mailto:[email protected]> From: [email protected] [mailto:[email protected]] On Behalf Of Todd Hodgen Sent: Wednesday, February 10, 2010 5:47 PM To: 'Michael Scheidell' Cc: [email protected] Subject: Re: [sipx-users] voip.ms: tries to connect to random port Doh! Didn't see this email, I'll call you in a few minutes. From: Michael Scheidell [mailto:[email protected]] Sent: Wednesday, February 10, 2010 1:55 PM To: Todd Hodgen Cc: [email protected] Subject: Re: [sipx-users] voip.ms: tries to connect to random port On 2/10/10 4:49 PM, Todd Hodgen wrote: I am using VOIP.MS for inbound and outbound calling, and I know of at least one other party doing the same on sipXecs standard install. Worked on 4.0, 4.0.2, 4.0.4 and on 4.1.xxxxxxxx. And, it worked on the first attempt, with nothing special being done to make it work. Happy to call you over it if you like. that would be great. they look like the most flexible of all services. 561-939-7259. I'd suggest you have a setup issue on your end. most likely. :-).. and with the randomness, its probably something I keep fixing then breaking. -- Michael Scheidell, CTO Phone: 561-999-5000, x 1259 > | SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program 2009, VARBusiness * Best Anti-Spam Product 2008, Network Products Guide * King of Spam Filters, SC Magazine 2008 ________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ________________________________ ________________________________ This message and any files transmitted with it are intended only for the individual(s) or entity named. If you are not the intended individual(s) or entity named you are hereby notified that any disclosure, copying, distribution or reliance upon its contents is strictly prohibited. If you have received this in error, please notify the sender, delete the original, and destroy all copies. Email transmissions cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. Garrett County Government therefore does not accept any liability for any errors or omissions in the contents of this message, which arise as a result of email transmission. Garrett County Government, 203 South Fourth Street, Courthouse, Oakland, Maryland 21550 www.garrettcounty.org ______________________________________________________________________ This email has been scanned and certified safe by SpammerTrap(r). For Information please see http://www.secnap.com/products/spammertrap/ ______________________________________________________________________
<?xml version="1.0" ?> <sipxbridge-config xmlns="http://www.sipfoundry.org/sipX/schema/xml/sipxbridge-00-00";> <bridge-configuration> <external-address>192.168.1.252</external-address> <external-port>5080</external-port> <local-address>192.168.1.252</local-address> <local-port>5090</local-port> <sipx-proxy-domain>example.com</sipx-proxy-domain> <sipx-supervisor-host>sample.sample.com</sipx-supervisor-host> <sipx-supervisor-xml-rpc-port>8092</sipx-supervisor-xml-rpc-port> <stun-server-address>stun01.sipphone.com</stun-server-address> <sip-keepalive-seconds>20</sip-keepalive-seconds> <sip-session-timer-interval-seconds>1800</sip-session-timer-interval-seconds> <media-keepalive-seconds>1</media-keepalive-seconds> <xml-rpc-port>8088</xml-rpc-port> <music-on-hold-support-enabled>true</music-on-hold-support-enabled> <music-on-hold-address>~~mh~</music-on-hold-address> <music-on-hold-delay-miliseconds>500</music-on-hold-delay-miliseconds> <music-on-hold-supported-codecs>PCMU,PCMA</music-on-hold-supported-codecs> <route-inbound-calls-to-extension>operator</route-inbound-calls-to-extension> <log-level>INFO</log-level> <log-directory>/var/log/sipxpbx/</log-directory> <location-id>1</location-id> </bridge-configuration> <itsp-account> <itsp-proxy-domain>sip.us3.voip.ms</itsp-proxy-domain> <user-name>SampleUserName</user-name> <password>SamplePassword</password> <itsp-proxy-address>sip.us3.voip.ms</itsp-proxy-address> <itsp-proxy-listening-port>0</itsp-proxy-listening-port> <itsp-transport>UDP</itsp-transport> <use-global-addressing>true</use-global-addressing> <strip-private-headers>false</strip-private-headers> <default-asserted-identity>true</default-asserted-identity> <register-on-initialization>true</register-on-initialization> <registration-interval>600</registration-interval> <sip-keepalive-method>CR-LF</sip-keepalive-method> <rtp-keepalive-method>NONE</rtp-keepalive-method> </itsp-account> </sipxbridge-config>
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