One thing to note is that when changing from 4.0.x to 4.2.x is that if you have any location based routing set up you will have to recreate those routing groups as branches. Other than that the transition should be relatively painless. A word of caution, do NOT use the web portal to perform upgrades, it is broken in 4.0.x. Use the command line as Tony suggests.
From: [email protected] [mailto:[email protected]] On Behalf Of Tony Graziano Sent: Wednesday, October 27, 2010 2:06 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Inbound Verizon Callers 2 things need to happen. 1. Rename the repo that is there as CentOS-Base.repo edit it and remove the sipx information at the bottom, that's hurting you right now, and save it. 2. get the 'intermediate" 4.2 repo using cd /etc/yum.repos.d wget http://download.sipfoundry.org/pub/sipXecs/sipxecs-4.2.0-centos-i386.repo (assuming its a 32 bit install) Then if you do an update (which you should), get a full backup first, and do the update from CLI (yum update). If that goes well, consider making sure everything is ok (that version is buggy with registrations), and you need to change the baseurl from 4.2.0 to 4.2.1 in the sipx repo file, save, and update again. Tony On Wed, Oct 27, 2010 at 2:33 PM, Bryan Anderson <[email protected]<mailto:[email protected]>> wrote: cat /etc/redhat-release = CentOS release 5 (sipXecs 4.0.1) I have performed an update once, using the web interface and it crash a couple of services and i couldn't bring them back up so i reinstalled and left it at that. The plan was to in a controled environment figure out the update process then update it. This is the only issue we have had with NexVortex, and I am not sure its them, but haven't ruled that out. They have been quite helpful. So far the issues have been something with my configuration. I'm not a Linux noob but not exactly an expert. I am how ever a noob when it comes to VOIP. Alot of ITSP's we talked to wanted to bring in a T1 and running the SIP trunk over it. A T1 is not available nor feasible for that office. Thanks Bryan ________________________________ Date: Wed, 27 Oct 2010 14:22:18 -0400 From: [email protected]<mailto:[email protected]> To: [email protected]<mailto:[email protected]> Subject: Re: [sipx-users] Inbound Verizon Callers Separately, you are on an "older" version of sipx. What does: cat /etc/redhat-release output say? It is likely you need a repo change and two inline updates in order to get current. On Wed, Oct 27, 2010 at 2:20 PM, Tony Graziano <[email protected]<mailto:[email protected]>> wrote: Firstly. There were many itsp's I have used with sipx (using ingate session border controllers). There were several ITSP's I found too difficult to use at all and nexvortex was one of them. Nice guys, just up the road here, but I had continual issues with them and had to pull the plug. That was a couple of years agao and maybe things have changed for them. I recall some upstream issues with certain carriers always being an issue. I did lots of traces and pcaps. I just started using itsp's that had better answers for what I needed. http://blog.myitdepartment.net/?p=127 Continued... ============================ Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: [email protected]<mailto:[email protected]> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartment.net/gethelp/ ----- Original Message ----- From: [email protected]<mailto:[email protected]> <[email protected]<mailto:[email protected]>> To: [email protected]<mailto:[email protected]> <[email protected]<mailto:[email protected]>> Sent: Wed Oct 27 13:50:35 2010 Subject: [sipx-users] Inbound Verizon Callers I work for a small non profit and we have begun using sipX in an office in Everett,WA. We have 8, 2 line polycom soundpoint 331 phone. We use NexVortex as our service provider with 4 DID's. Just after moving to this system we noticed some calls would ring once and then disconnect. We did some testing and discovered it was isolated to Verizon Wireless callers. I have yet to figure it out. I have had 3 T- mobile customers call, 2 AT&T, 1 Sprint, and 5 Verizon. I have tried 7 land lines. Aside from those about 20-40 calls a day in and out of the system work. Callers that have show as "failed" or "Abandoned" in the CDR when called back say they are Verizon and report the same issue, "It rang once then hung up". Two of the DID's go to hunt groups that simultaneously ring all phones. If no answer the Hunt group falls to x200 and hits a general mailbox. One number is an Alias that is Assigned to a user, which is the user on line 2 of all 8 phones. The fourth number is really just for me to test with. Yesterday I did some testing on routing the call different ways in our system using one number. 1) I sent callers to a hunt group = Verizon callers got disconnected. 2) I sent callers to auto-attendant = Verizon callers got through, entered an extension and calls were answerable by our users. (if no answer it went to voicemail) 3) I sent callers to a user directly = Verizon callers got through to that extension. (if no answer it went to voicemail) So if I send them to a hunt group, the call doesn't work, If I send them to a user assigned to all the phones, it doesn't go through. If I send them to 1 user or to auto attendant, the call goes through. Any ideas or advice? sipXconfig (4.2.0-018575 2010-04-15T00:40:11 ecs-centos5) I Installed from an ISO. When I last tried to upgrade various services stopped functioning. Bryan Anderson -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected]<mailto:[email protected]> Fax: 434.326.5325 Email: [email protected]<mailto:[email protected]> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected]<mailto:[email protected]> Helpdesk Contract Customers: http://support.myitdepartment.net _______________________________________________ sipx-users mailing list [email protected]<mailto:[email protected]> List Archive: http://list.sipfoundry.org/archive/sipx-users/ _______________________________________________ sipx-users mailing list [email protected]<mailto:[email protected]> List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected]<mailto:[email protected]> Fax: 434.326.5325 Email: [email protected]<mailto:[email protected]> LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected]<mailto:[email protected]> Helpdesk Contract Customers: http://support.myitdepartment.net
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