Not likely. IMHO, if you look at your configuration between sipXecs and this carrier, it relies on one configuration between the ITSP and your SBC. That configuration doesn't change based on the number of phones that you have.
If you think plugging in the last phone created the issue, it would be simple enough to unplug it and see if it starts working again. I suspect it won't make a difference. I'd check with your ITSP to see if the resolution they created is still working, as it may have been removed by the resetting of something, somewhere. From: [email protected] [mailto:[email protected]] On Behalf Of Bryan Anderson Sent: Thursday, November 18, 2010 12:52 PM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Inbound Verizon Callers uhm.... I discovered that not all 8 phones were connected. After plugging in the 8t phone, the calls stopped working from verizon again. Any ideas? Could there be a setting on the phone system? Bryan On Thu, Nov 18, 2010 at 11:44 AM, Bryan Anderson <[email protected]> wrote: Okay, NexVortex "looked" into it, and even "opened" a ticket with their underlying carrier. Both of them said "It is with Verizon". I opened a ticket with Verizon just for a paper trail and they opened at ticket with Level Three. NexVortex was not responding with a 200 OK within the 40ms time required. They have now resolved the issues and calls are going through. Just thought I would give you all an update. Bryan On Thu, Oct 28, 2010 at 10:21 AM, Bryan Anderson <[email protected]> wrote: Ok, so YAY my server is up to date now. Although still having Verizon wireless issues, and still can't get the instant messenger service to stay running, although right now that is not a concern for us. Bryan On Thu, Oct 28, 2010 at 9:02 AM, Bryan Anderson <[email protected]> wrote: @Michael - What do you have for phone service? nexvortex or some one/thing else? On Thu, Oct 28, 2010 at 8:22 AM, Tony Graziano <[email protected]> wrote: disable role, restart svc, enable role restart svc. On Thu, Oct 28, 2010 at 11:18 AM, Bryan Anderson <[email protected]> wrote: Clean All worked. Now Instant messanger is failing to start. It reports: Standard output * Starting openfire: * Starting openfire On Thu, Oct 28, 2010 at 4:13 AM, Michael Picher <[email protected]> wrote: Try a 'yum clean all' before the 'yum update' On Thu, Oct 28, 2010 at 1:51 AM, Bryan Anderson <[email protected]> wrote: So with the above mentioned URL I got 404 not found, removed the "i386/" at the end and got this: Not using downloaded repomd.xml because it is older than what we have: Current : Thu Sep 2 08:55:07 2010 Downloaded: Thu Aug 26 07:43:26 2010 Setting up Update Process No Packages marked for Update >.< On Wed, Oct 27, 2010 at 10:05 PM, Bryan Anderson <[email protected]> wrote: Should I change the Base URL to that? if so, still no packages. >.< _____ Date: Thu, 28 Oct 2010 00:02:38 -0400 From: [email protected] To: [email protected] Subject: Re: [sipx-users] Inbound Verizon Callers By golly you might be right. Try this one: http://download.sipfoundry.org/pub/sipXecs/4.2.1/CentOS_5/i386/ On Wed, Oct 27, 2010 at 11:45 PM, Bryan Anderson <[email protected]> wrote: ok, so this is what i did: 1 ) I renamed the repo under /etc/yum.repos.d to CentOS-Base.repo 2) removed the SipX info at the bottom of that repo 3) wget http://download.sipfoundry.org/pub/sipxecs/sipxecs-4.2.0-centos-i386.repo 4)ran "yum update", that updated 242 packages. 5) logged into sipx and didn't see a difference, so I restarted. 6) verifyed everything works. /etc/redhat-release now reads "CentOS release 5.5 (Final)" 7) changed base url to 4.2.1 8) ran "yum update" again and it says "No packages marked for update" Did I do something wrong? Thanks Bryan On Wed, Oct 27, 2010 at 12:05 PM, Tony Graziano <[email protected]> wrote: 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 <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 <http://support.myitdepartment.net/> _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- -Bryan Anderson _______________________________________________ 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 <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/ -- -Bryan Anderson _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- There are 10 kinds of people in this world, those who understand binary and those who don't. [email protected] blog: http://www.sipxecs.info <http://www.sipxecs.info/> call: sip:[email protected] <mailto:sip%[email protected]> _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- -Bryan Anderson _______________________________________________ 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 <http://support.myitdepartment.net/> _______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/ -- -Bryan Anderson -- -Bryan Anderson -- -Bryan Anderson -- -Bryan Anderson
_______________________________________________ sipx-users mailing list [email protected] List Archive: http://list.sipfoundry.org/archive/sipx-users/
