Yes, I do register with ITSP just fine. Attached you will find Flowroute's config per your request. Thank you.
--- On Sat, 11/21/09, M. Ranganathan <mra...@gmail.com> wrote: > From: M. Ranganathan <mra...@gmail.com> > Subject: Re: [sipx-users] sipX Bridge > To: "Jordan Turner" <jordan.turner1...@yahoo.com> > Cc: "Scott Lawrence" <scott.lawre...@nortel.com>, "Todd Hodgen" > <thod...@verizon.net>, sipx-users@list.sipfoundry.org > Date: Saturday, November 21, 2009, 5:59 PM > On Sat, Nov 21, 2009 at 12:51 PM, > Jordan Turner > <jordan.turner1...@yahoo.com> > wrote: > > Thank you Mr. Lawrence and everyone for their > patience. I have successfully got all of the > communications to work. Ihe issue the entire time was > having the Internet Calling enabled. Maybe an > oversight on my part but more information and even examples > given here should be added to the WIKI. As mentioned > last time, my impression of that button or feature was not > what I thought it was. All is well with remote workers > and ITSP calls via Flowroute. For any interested with > flowroute, method to keep SIP alive should be EMPTY SIP > message and method to keep RTP alive should be REPLAY LAST > SENT Packet, and all should work. It's time for me to move > further into the sipXecs feature! > > > Do you REGISTER with the ITSP? > > Please edit out any sensitive information ( passwords for > example ) in > > /etc/sipxpbx/sipxbridge.xml > > and please mail it to me. I can update the wiki. > > Ranga. > > > > > > > > --- On Sat, 11/21/09, Scott Lawrence <scott.lawre...@nortel.com> > wrote: > > > >> From: Scott Lawrence <scott.lawre...@nortel.com> > >> Subject: Re: [sipx-users] sipX Bridge > >> To: "Jordan Turner" <jordan.turner1...@yahoo.com> > >> Cc: "'M. Ranganathan'" <mra...@gmail.com>, > "Todd Hodgen" <thod...@verizon.net>, > sipx-users@list.sipfoundry.org > >> Date: Saturday, November 21, 2009, 1:11 PM > >> On Fri, 2009-11-20 at 23:29 -0800, > >> Jordan Turner wrote: > >> > >> > After re-reading the advices, I redid my > configs > >> differently. I > >> > started with vanilla install again and this > time I did > >> not choose > >> > Internet Calling. And the extension > calling > >> between remote workers > >> > worked! somehow, I had in my mind that > Internet > >> Calling is for ANYONE > >> > outside the Intranet. Thus now, the > default SBC > >> works - if I do not > >> > choose Internet Calling and using the default > SBC. > >> > >> Internet Calling refers to a capability you may or > may not > >> need. Most > >> people make phone calls with phone numbers, but in > SIP you > >> can also use > >> a SIP URL, which looks like an email address: > sip:u...@example.com > >> > >> If your system is not directly on the Internet > (most are > >> not), then > >> routing a call to an arbitrary domain name > requires that it > >> go through > >> an SBC. This routing to the SBC is _all_ > that the > >> Internet Calling > >> does. If you decide to enable this > capability, you > >> can use the same > >> sipXbridge instance for this that you use for the > >> connection to your > >> ITSP. > >> > >> > At this point, the BIG question is, what is > the next > >> step for me to > >> > take to add ITSP into the mix (i.e. add > another SBC or > >> Gateway - > >> > example next step would be great)? I > need more > >> of the exact steps > >> > since I already gotten Flowroute ITSP working > but > >> messed up > >> > SBC/Internet calling mixed up earlier with > internal > >> communications. > >> > >> http://sipx-wiki.calivia.com/index.php/SIP_Trunking_with_sipXecs:_Overview_and_Configuration#How_to_configure_sipXbridge > >> > >> Since you appear to be the first person to try > using > >> Flowroute with > >> sipXecs (if someone else out there has done this, > please > >> speak up), > >> you'll have a little pioneering to > do. If > >> you follow the directions on > >> that wiki page carefully one step at a time, you > should be > >> able to get > >> it working. > >> > >> If you're not sure what to do with a setting, > leave it at > >> the default or > >> stop and ask on this list. > >> > >> You do _not_ need to change any settings on your > phones - > >> leave them > >> alone and they should continue to work. > >> > >> > >> > > > > > > > > > > > > -- > M. Ranganathan >
<?xml version="1.0" ?> <sipxbridge-config xmlns="http://www.sipfoundry.org/sipX/schema/xml/sipxbridge-00-00";> <bridge-configuration> <global-address>EXTERNALIPADDRESSOFSIPXFROMOUTSIDE</global-address> <global-port>5060</global-port> <external-address>LOCALIPADDRESSOFSIPXONNETWORK</external-address> <external-port>5080</external-port> <local-address>LOCALIPADDRESSOFSIPXONNETWORK</local-address> <local-port>5090</local-port> <sipx-proxy-domain>YOURSIPDOMAIN</sipx-proxy-domain> <sipx-supervisor-host>HOSTNAMEOFDNSSERVERFROMOUTSIDE</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>false</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.flowroute.com</itsp-proxy-domain> <user-name>ITSPUSERNAME</user-name> <password>ITSPPASSWORD</password> <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>REPLAY-LAST-SENT-PACKET</rtp-keepalive-method> </itsp-account> </sipxbridge-config>
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