All, I am still having issues here - the ITSP is basically insisting I use 5060, not 5080 (they apparently tried changing the Asterix port to 5080 and it didn't work for them for some reason), and I just want to double check my settings before going back to them. I have SipX installed on a machine with two NICS - one on the ITSP external network and one on our internal network. I was wondering, given this is the case, whether it mattered that they want us to use 5060, we should be able to bind to 5060 on the external address and separately to 5060 on our internal address for the phones, no? Currently I have a gateway set up with our ITSP (Allegro in Australia): <?xml version="1.0" ?> <sipxbridge-config xmlns="http://www.sipfoundry.org/sipX/schema/xml/sipxbridge-00-00";> <bridge-configuration> <external-address>172.32.255.140</external-address> <-- the address the ITSP provided me for the sipx server <external-port>5060</external-port> <local-address>10.10.5.28</local-address> <-- the internal address of the sipx server <local-port>5090</local-port> <sipx-proxy-domain>mionegroup.com.au</sipx-proxy-domain> <sipx-supervisor-host>sipx.mionegroup.com.au</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>172.32.255.138</itsp-proxy-domain> <user-name>USER</user-name> <password>PWD</password> <itsp-proxy-address>172.32.255.138</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> When I try to start the SIP Trunk, I get the following exception. I assume this is related to the 5060/5080 issue: Messages for SIP Trunking Standard error • javax.sip.InvalidArgumentException: Address already in use • at gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:778) • at org.sipfoundry.sipxbridge.Gateway.initializeSipListeningPoints(Gateway.java:471) • at org.sipfoundry.sipxbridge.Gateway.start(Gateway.java:898) • at org.sipfoundry.sipxbridge.Gateway.main(Gateway.java:1233) • Caused by: java.io.IOException: Address already in use • at gov.nist.javax.sip.stack.UDPMessageProcessor.<init>(UDPMessageProcessor.java:141) • at gov.nist.javax.sip.stack.SIPTransactionStack.createMessageProcessor(SIPTransactionStack.java:1742) • at gov.nist.javax.sip.SipStackImpl.createListeningPoint(SipStackImpl.java:760) • ... 3 more • SipXbridge : Exception caught while running I feel like I am trying to make something work that just isn't designed to unless the ITSP can change their port to support 5080. Cheers, David ----- Original Message ----- From: "M. Ranganathan" <[email protected]> To: "David Hobley" <[email protected]> Cc: [email protected] Sent: Tuesday, May 12, 2009 4:28:32 PM GMT +10:00 Brisbane Subject: Re: [sipx-users] Asterix in use by ITSP? On Tue, May 12, 2009 at 2:12 AM, David Hobley <[email protected]> wrote: > Hello, > > I am just trying to set SipX 4.0 up to interoperate with our ITSP. These > guys are using Asterix and are currently claiming that they can't set the > SIP port at their end to chat on 5080 (as I believe, from the documentation, > is required at our end). Has anyone had any success connecting with Asterix > at the ITSP end? Sure! callwithus.com uses asterisk and if I am not mistaken so does vitelity.net. These ITSPs typically expect Registration. Enable Register on intialization in the ITSP configuration screen. You should be able to register with port 5080 unless the ITSP has seriously restricted the behavior of Asterisk to only allow registration from Port 5060 ( in which case you would have a problem ). > > Cheers, > David > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users > Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users > -- M. Ranganathan -- Cheers, David Hobley IT Manager Creators of Miessence, MiVitality and MiEnviron Phone: +61 (7) 5582 7020 Fax: +61 (7) 5539 6719 USA Fax 1800 840 0827 Email : [email protected] Website: www.mionegroup.com
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