They new (not really new anymore) is called Fusion, and it does not support invites without SDP. Until they accept that, they aren't going to work with SIPXecs. So, ensure you are on their legacy system. Additionally, they are making some changes to their network by end of month. I haven't made the changes yet, but hopefully, no issues when I start migrating about 12 customers.
>From the main menu, go to Diagnostics, Configuration Test, and select the option at the bottom of the page to Run the tests. It will check your SRV records, DNS, DHCP, etc. IF there are errors, report back so someone can help interpret them. Broadvox works well with Polycom Phones. I'd be curious if you have QOS issues though. I would recommend setting up your router port on a mirrored port on your switch so you can see all traffic coming onto the network. Then run a packet capture on that port and look to see what kind of Jitter and lost packets you have. Running Ping Plotter sometimes gives you some idea of what is going on with the network between your end stations and Broadvox. Are you using Covad circuits to Broadvox by chance? From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Saturday, August 20, 2011 8:28 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Attempt To Make Callls But Never Successully Invite Sent Over and Over you need to ask broadvox if you are on their new product or the "legacy" platform. im not aware that anyone using sipx has their new platform working as of yet. On Aug 20, 2011 11:25 PM, <supp...@itnc.biz> wrote: > All, > > I've included a packet capture that was taken from the InGate SBC. I > would have attached on earlier, but the supp...@itnc.biz is a distro list > and I was unable to send as it from Outlook, had to make changes to my > Exchange server. Anyway the ITSB is BroadVOX. This is a system that has > been in production for several years and we took over their IT. BroadVOX > has been their SIP provider the entire time. Sorry for the lack of the > phones etc. I'd included them on a previous email and thought I'd included > them. They are using PolyCom IP 650 and 330 phones. We'll try some of the > suggestions given by Todd. We are still new to the SipXecs and I'm not the > tech looking into this at the time, so not to sound like newbie, but just > in cass, since I'm not logged into the SipXecs right now, how do we run a > diagnostic configuration test. > > > > Brian Buckles > > IT Manager > > IT Network Consultants > > (859)963-1911 > > 877-888-ITNC (4862) > > brian.buck...@itnetworkconsultants.com > > > > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen > > Sent: Saturday, August 20, 2011 7:24 PM > > To: 'Discussion list for users of sipXecs software' > > Subject: Re: [sipx-users] Attempt To Make Callls But Never Successully > Invite Sent Over and Over > > > > Look up sipviewer on the wicki - it will be your friend - downloand and > install on a computer. > > Make sure you have Winscp on your computer as well. > > Make sure you have Putty. > > > > Go to /var/log/sipxpbx on the server and do a logrotate. Instructions on > the wiki. > > Try a call, and then run merge-logs from putty in /var/log/sipxpbx > > This will create a merged.xml file - open it with sipviewer to see what your > calls are doing. > > > > Name the ITSP, and how you have it setup - Like the name of the company - we > need to see if it is one that others have used successfully. If not, we can > recommend some to try to see if it is the system or ITSP that is at fault. > > > > Time is off - generally means it can't get to the NTP server you have > defined. Makes me wonder what you have for a router? > > > > What type of phone are you using? Can you call phone to phone? Can you > call VP or auto attendant - 100 or 101? > > > > Have you ran Diagnostics Configuration Test - what were the results? > > Have you downloaded Preflight and ran it from another machine on the network > - what were the results? > > > > These are the types of things that will help people see just how bad things > are, or how close you might be to having things working. > > > > Now, get going............ > > > > > > > > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > supp...@itnc.biz > > Sent: Saturday, August 20, 2011 2:59 PM > > To: sipx-users@list.sipfoundry.org > > Subject: [sipx-users] Attempt To Make Callls But Never Successully Invite > Sent Over and Over > > > > > > --Please enter your response above this line-- > > We have a client using SipXecs 4.0.1-015823 They've had several phone > issues over the last few weeks with call quality and a few other things. We > resolved the issue for several weeks by replacing a switch and cabling. On > Thursday they had some transfer issues and the date/time were wrong after we > rebooted the SipXecs because it froze while we were in it look for the cause > of poor quality which had just returned. We restored a recent backup to the > SipXecs thinking that the config may have been corrupted during the system > hang. Note prior to restoring the config, calls to/from outside worked, > only the date/timer were wrong and internal transfers did not work. Since > the restore they cannot send or receive calls to the outside. A packet > capture on the SBC (InGate Siparator) and at the ITSP reveal the following > error. > > "Traversal licenses limit has been reached. Please contact your reseller to > upgrade with more SIP Traversal Licenses." Upon talking with support of the > SBC and ITSP, they analyzed packet captures and noticed that if a call is > placed...the ITSP sends us an invite, then we send an invite back to them > and this happens multiple times over and over, for just one VOIP call. They > say that something in the phone system is causing a loop and that during > normal call the following should happen...The ITSP sends an invite>>we send > a 100 trying>>a ringing 180 or 183 is sent>>a 200 OK is sent >> then an ACK > is sent and normal call progresses. We are working on setting up SipXecs > 4.4 on a virtual machine and restore the most resent backup. However just > in case this doesn't work, what logs do we need to look at for the above > issues and does anyone have any suggestions on this? This client has > multiple companies working off this SIP Trunk and we really need it fixed by > Monday. Any suggestion would be appreciated. Thanks in advance. > > > > ________________________________________ > > > > > > ________________________________________ > > No virus found in this message. > > Checked by AVG - www.avg.com > > Version: 10.0.1392 / Virus Database: 1520/3846 - Release Date: 08/20/11 > _____ No virus found in this message. Checked by AVG - www.avg.com Version: 10.0.1392 / Virus Database: 1520/3847 - Release Date: 08/20/11
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