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]>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] > > To: [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]> 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] > > 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] > <[email protected]> > To: [email protected] <[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] > Fax: 434.326.5325 > > Email: [email protected] > > LAN/Telephony/Security and Control Systems Helpdesk: > Telephone: 434.984.8426 > sip: [email protected] > > Helpdesk Contract Customers: > http://support.myitdepartment.net > > > _______________________________________________ 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/ > -- ====================== Tony Graziano, Manager Telephone: 434.984.8430 sip: [email protected] Fax: 434.326.5325 Email: [email protected] LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 sip: [email protected] Helpdesk Contract Customers: http://support.myitdepartment.net
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