In the voip.ms template you would put in atlanta in two places. beyond that, your first description was wrong on the call flow.
how sipx uses ports on itsp calls... FROM port 5080 (sipx) to ITSP on port 5060 (ITSP) for outbound calls. FROM port 5060 (ITSP) to sipx on port 5080 (sipx) for calls coming TO sipx. You are literally having a bunch of people trying to help you, but you really need to be accurate in your descriptions, chaos ensues. Do us all a favor and consider a backup, yum update and reboot so you have something new-ish and describe your call flow a little more accurately. Besides that, the little snippets from your firewall tell us little other than what IP's are talking to what ports. So you should ensure your sonicwall has current supportable code in it and consider sending a siptrace. If you confirm the music on hold is being heard AND it is sipx MOH, the problem could be any number of things, the least of which is phone firmware. SIPTRACE http://wiki.sipfoundry.org/display/sipXecs/Display+SIP+message+flow+using+Sipviewer The problem is MOH and resume. Your firewall isn't giving us diddly to go on for that. Either get a pcap at the firewall looking at the ITSP's address or get a siptrace (preferred) so we can see what sipx is thinking before it does it. We've seen this done a hundred times with sonicwall, without issue, while we suspect phone firmware and the fact you are woefully behind (as in sipx and phone firmware is wayyyy out of date)... hard to keep asking folks to get in wayback machine multiple times for you. You may also consider updating ONE phone to 3.2.6 and bootrom 4.3.1 and test just it. On Tue, Mar 27, 2012 at 4:20 PM, Stiles Watson <[email protected]>wrote: > Todd, > > No, I've not changed any port info. In the SBC SIP settings "Public Port" > is empty, and "External Port" is the default of 5080. Also, in the > voip.msgateway, the only thing I changed from the default template is > Username, > Authentication Username, Password, and ITSP server address (changed from > sip.ca2.voip.ms to atlanta.voip.ms). I set the Dial Plan to the built-in > Long Distance dial plan and added the DID as an alias to the built-in Auto > Attendant. > > Stiles > > > On 03/27/2012 04:00 PM, Todd Hodgen wrote: > > Stiles, are you by chance putting port numbers in fields that are > blank? There are usually some fields that call for a port number that are > blank, as they default to 5060, sometimes people fill those fields out. > If you have changed a field from its default to something different, you > will notice a dotted line around it to denote it is changed from the > default. You would want to remove those and keep the defaults.**** > > ** ** > > *From:* [email protected] [ > mailto:[email protected]<[email protected]>] > *On Behalf Of *Tony Graziano > *Sent:* Tuesday, March 27, 2012 12:49 PM > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] voip.ms config**** > > ** ** > > That looks wrong. The destination port for an outbound call should be 5060 > not 5080. If you used the stock template the firewall is he issue here.*** > * > > On Mar 27, 2012 3:38 PM, "Stiles Watson" <[email protected]> wrote:** > ** > > I have a Sonicwall NSA 240. I've gone back and deleted all the firewall > rules and NAT policies related to sipx and only created new ones for 5080 > UDP and 30000-31000 UDP, leaving 5060 blocked at this point. Sip > transformations is turned off and Consistent NAT is turned on. When I make > a call I see the following in the Sonicwall's Connections Monitor:**** > > #**** > > Src IP**** > > Src Port**** > > Dst IP**** > > Dst Port**** > > Protocol**** > > Src Iface**** > > Dst Iface**** > > Flow Type**** > > IPS Category**** > > Expiry (sec)**** > > Tx Bytes**** > > Rx Bytes**** > > Tx Pkts**** > > Rx Pkts**** > > 1**** > > 174.34.146.162 (atlanta.voip.ms)**** > > 18530**** > > xxx.xxx.xxx.xxx (sipx server)**** > > 30000**** > > UDP**** > > X1**** > > X3**** > > N/A**** > > 29**** > > 837786**** > > 754800**** > > 4259**** > > 3774**** > > 2**** > > 174.34.146.162**** > > 5060**** > > xxx.xxx.xxx.xxx**** > > 5080**** > > UDP**** > > X1**** > > X3**** > > SIP Control**** > > N/A**** > > 21**** > > 6335**** > > 5586**** > > 10**** > > 12**** > > > Connection #2, the SIP Control, is the registration connection and stays > live all the time. #1 above only appears when there is an active call. > > Stiles > > > On 03/27/2012 12:25 PM, Gerald Drouillard wrote: **** > > On 3/27/2012 12:03 PM, Stiles Watson wrote: **** > > This is where one swallows one's pride.... The way I was entering data > caused the drop-down to not be displayed. > > To keep this short:**** > > 1. When you first select Add new gateway>Sip Trunk, the template drop > down is not visible. I was not aware this was the case until yesterday. I > just thought it was not there.**** > 2. The template drop-down is only displayed after you enter a name for > the gateway and then select the default SBC.**** > 3. If you ever click the Apply button before both the name and SBC are > entered, the drop down is never displayed. **** > > This is why I never saw the template drop-down. > > Now, having said all of that, I deleted my existing voip.ms gateway and > created a new one using the template drop-down. However, this did not fix > my problem and everything is as it was before. I still can not retrieve a > call from hold or cancel a transfer. I have verified in my voip.msaccount > that it is registered with the public IP and port 5080. > > So it looks like we are back to a firewall problem, correct?**** > > Yes. What kind of firewall do you have? > > **** > > -- **** > > Regards**** > > --------------------------------------**** > > Gerald Drouillard**** > > Technology Architect**** > > Drouillard & Associates, Inc.**** > > http://www.Drouillard.biz**** > > ** ** > > _______________________________________________**** > > sipx-users mailing list**** > > [email protected]**** > > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/**** > > ** ** > > LAN/Telephony/Security and Control Systems Helpdesk:**** > > Telephone: 434.984.8426**** > > sip: [email protected]**** > > ** ** > > Helpdesk Customers: http://myhelp.myitdepartment.net**** > > Blog: http://blog.myitdepartment.net**** > > > _______________________________________________ > sipx-users mailing [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > > > > _______________________________________________ > sipx-users mailing list > [email protected] > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- ~~~~~~~~~~~~~~~~~~ Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.465.6833 ~~~~~~~~~~~~~~~~~~ Linked-In Profile: http://www.linkedin.com/pub/tony-graziano/14/4a6/7a4 Ask about our Internet Fax services! ~~~~~~~~~~~~~~~~~~ -- LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Customers: http://myhelp.myitdepartment.net Blog: http://blog.myitdepartment.net
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