Re: [sipx-users] New Version Release

FWIW you can't use the newer (cheaper) Polycom 335 phones with 4.0.4 or the current polycom firmware release (3.2.2). I wouldn't deploy it in any sort of critical/mass production mode, but the trunk seems to be kept in pretty good shape. Its good enough to use on a daily basis for my normal des

Re: [sipx-users] New Version Release

It will be ready when it's ready. At the current pace I would predict June/July before you would actually want to use 4.2 as a production system. If it is really critical that it be right, I'd wait until 4.2.1 is available because major changes are afoot. This is one man's opinion (who has been

Re: [sipx-users] New Version Release

I can't just tell you. It is not that simple of an answer. If you read the entire page, you will see. If you don't want to read the entire page, search for the string 32nd and start with that paragraph. I have no input in this. I'm just posting links to information that is already out there. O

Re: [sipx-users] New Version Release

I don't see anything can you just tell me? -Original Message- From: mkitchin.pub...@gmail.com [mailto:mkitchin.pub...@gmail.com] Sent: Wednesday, February 03, 2010 5:26 PM To: Goran Donev Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] New Version Release Follow this link, a

Re: [sipx-users] X-lite outgoing caller ID issues.

If you give them real extensions does it work right? It may be something in the softphone. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jim Canfield Sent: Wednesday, February 03, 2010 7:08 PM To: sipx-users@li

[sipx-users] X-lite outgoing caller ID issues.

Hey guys, I have an interesting situation using X-lite 3.0 soft-phones and the 4.0 Beta. Before I start gathering sip traces, I thought I'd see if this is the expected behavior... 1 - Created soft-phone users. Sation01, Station02... 2 - Registered soft-phones manually (i.e. no phone provision

Re: [sipx-users] Freeswitch processor usage

Welcome to my hell. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk Contract Customers: http://www.myitdepartm

Re: [sipx-users] Freeswitch processor usage

After reading a post today that mentioned size limits in posts, I realized these posts never made it through because it had 2 screenshots. I have removed the pictures and am resending. Any help would be greatly appreciated! On 2/3/2010 12:05 PM, mkitchin.pub...@gmail.com wrote: I rebooted last

Re: [sipx-users] New Version Release

Follow this link, and start about 3/4 of the way down the page: http://forum.sipfoundry.org/index.php?t=msg&goto=40902&S=3207f6889462c7a977fe255cd8f76280#msg_40902 On 2/3/2010 5:18 PM, Goran Donev wrote: > I was wondering if the development team knows the time frame of having the > new version re

[sipx-users] New Version Release

I was wondering if the development team knows the time frame of having the new version released. I am looking to implement this for a business but want to have the latest version, if it's feasible to wait. Thanks. ___ sipx-users mailing list sipx-user

Re: [sipx-users] require one to one NAT(port 5060 -5080) mapping

If it has a filter for SIP it must be turned off (SPF or ALG). Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Telephone: 434.984.8426 Fax: 434.984.8427 Helpdesk

Re: [sipx-users] require one to one NAT(port 5060 -5080) mapping

We are using sonicwall NSA 2400, it always keep said my External port != internal port, How to check my port is be symmetric? isit any log will show? Tony Graziano wrote: On Wed, Feb 3, 2010 at 3:18 PM, Winson (Elabram) wrote: Dear all Expert ! Hi i stuck in

Re: [sipx-users] New CDR database design

Reference helps. A lot. If a call get transferred/put on hold/whatever it is crucial to know how long the call is at each destination (for billing purposes) and be able to link it back to the original call. While it would be nice to have "universal call identifier" and "call sequence number" da

Re: [sipx-users] require one to one NAT(port 5060 -5080) mapping

On Wed, Feb 3, 2010 at 3:18 PM, Winson (Elabram) wrote: > Dear all Expert ! > Hi i stuck in the firewall NAT mapping for my Sipbridge - ISTP . > > I try direct use the modem connected to sipXecs and open the port > 5060,5080, 3-31000,16384-16482(ISTP) is wan work. > > after i plug in the firew

[sipx-users] require one to one NAT(port 5060 -5080) mapping

Dear all Expert ! Hi i stuck in the firewall NAT mapping for my Sipbridge - ISTP . I try direct use the modem connected to sipXecs and open the port 5060,5080, 3-31000,16384-16482(ISTP) is wan work. after i plug in the firewall, the softphone show me is error 408,time out. Then i check the s

Re: [sipx-users] New CDR database design

Why not install 4.1.x and get raw CDR and inspect it? The ISO was taken down a few weeks ago it seems while they finalize some instability issues. I have a copy of the last ISO as I installed it on a lab system. If you need a copy, let me know. -Original Message- From: sipx-users-boun..

Re: [sipx-users] New CDR database design

Scott wrote: >Subject: Re: [sipx-users] New CDR database design > >On Wed, 2010-02-03 at 13:43 -0600, Josh Patten wrote: >> I am curious what the new CDR database design is going to look >> like/consist of. I am meeting with my programming team today to >> discuss writing a CDR parser and I would

Re: [sipx-users] New CDR database design

There is no way from the generated CDR's to follow the call flow. There is also no way to know how long a call actually lasted if it was transferred or put on hold. This information exists in the Call State Events table but doesn't transfer over to the cdrs table. I posted about the ideas I had

Re: [sipx-users] New CDR database design

On Wed, 2010-02-03 at 13:43 -0600, Josh Patten wrote: > I am curious what the new CDR database design is going to look > like/consist of. I am meeting with my programming team today to discuss > writing a CDR parser and I would like to show them examples of what kind > of data they can expect to

[sipx-users] New CDR database design

I am curious what the new CDR database design is going to look like/consist of. I am meeting with my programming team today to discuss writing a CDR parser and I would like to show them examples of what kind of data they can expect to work with. As it currently stands it looks like they will ha

Re: [sipx-users] "media release" support

I had my discussion with Verizon, and they are not able to offer me anything that looks appealing in any way. I think I'm only left with a few questions. 1) If I want to centralize as much as possible, what components are required at a remote site to force the call traffic to go directly out tha

Re: [sipx-users] sipXbridge and REFER

On Wed, Feb 3, 2010 at 12:13 PM, Scott Lawrence wrote: > On Wed, 2010-02-03 at 11:54 -0500, Dale Worley wrote: >> On Wed, 2010-02-03 at 10:18 +, Sven Evensen wrote: >> >> > The scenario is as follows: >> > >> > A calls B, B answers >> > >> > Our appl. sends a REFER to transfer from A to C >> >

Re: [sipx-users] sipXbridge and REFER

On Wed, 2010-02-03 at 11:54 -0500, Dale Worley wrote: > On Wed, 2010-02-03 at 10:18 +, Sven Evensen wrote: > > > The scenario is as follows: > > > > A calls B, B answers > > > > Our appl. sends a REFER to transfer from A to C > > > > The REFER has From : A, To : B, Refer-To : C, Referred-B

Re: [sipx-users] sipXbridge and REFER

On Wed, 2010-02-03 at 10:18 +, Sven Evensen wrote: > The scenario is as follows: > > A calls B, B answers > > Our appl. sends a REFER to transfer from A to C > > The REFER has From : A, To : B, Refer-To : C, Referred-By : A > > When using unmanaged GW (Cisco IOS GW), it replies with > >

Re: [sipx-users] best replacement phone for cisco7960

Or 670's if you desire a color display. The vvx is supported, but its way to pricey and doesn't do much more without connecting to a polycom server for a webfeed. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/

Re: [sipx-users] best replacement phone for cisco7960

Either Polycom IP 550's or 650's. 550's have 4 line buttons, 650's have 6 line buttons plus the ability to add an expansion console. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 2/3/2010 9:44 AM, Michael Scheidell wrote: yes, cisco 7960 isn't sip complia

[sipx-users] best replacement phone for cisco7960

yes, cisco 7960 isn't sip compliant, the old ones didn't even do POE right, and cisco hates sip. so, whats the best phone. #1, executive phone (at least 3 lines and at least 3 local speed dials, good soft buttons for most features, supports headsets) #2, assistant phones (at least 2 lines, can

Re: [sipx-users] sipXBridge being bypassed

Yes, 5080... you will experience the problems you are experiencing if calls are coming in on 5060. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Sven Evensen Sent: Wednesday, February 03, 2010 8:26 AM To: Scott

[sipx-users] skype sip won't register anymore

no updates, no changes to anything, all of a sudden, 3:51pm yesterday, skype sip stopped registering. can't find ANY support contact info, except for their forums. did get an email from someone at skype warning us that they may have made mistakes in billing, but said they were fixed. (I emailed

Re: [sipx-users] sipXBridge being bypassed

OK. So on outbound calls, the calls go via sipXBridge if the dial plan directs it to, i.e. to go via a GW with sipXBridge enabled. To get inbound calls to go via sipXBridge, we must make sure the initial INVITE comes in on 5080, either the ITSP can do it, translation in firewall or even using ipta

Re: [sipx-users] sipXBridge being bypassed

On Wed, 2010-02-03 at 11:43 +, Sven Evensen wrote: > We are using a SIP provider in UK which operates with two different IP > addresses. > > In the GW/ITSP setup in sipxConfig, we enter their main sip domain, > username, password etc. > > So INVITEs going to this SIP trunk, go out this way. >

Re: [sipx-users] sipXbridge and REFER

On Wed, 2010-02-03 at 10:18 +, Sven Evensen wrote: > (Is there a mximum size of attachments? ) Yes - 60K > Our application uses REFER to transfer an external call (mobile) from > one internal user to another internal user > > When the REFER is sent to the Cisco GW, it responds with an INVITE

Re: [sipx-users] sipXBridge being bypassed

Create an unmanaged gateway for the inbound calls with the IP address, with no dialing rules. I'm assuming the way the firewall is configured that sipx can route to it, and it can route to sipx. Only use the other one for sipxbridge. On Wed, Feb 3, 2010 at 6:43 AM, Sven Evensen wrote: > We are

[sipx-users] sipXBridge being bypassed

We are using a SIP provider in UK which operates with two different IP addresses. In the GW/ITSP setup in sipxConfig, we enter their main sip domain, username, password etc. So INVITEs going to this SIP trunk, go out this way. But for incoming calls, the INVITEs come from a different IP addre

[sipx-users] Time format and entering a time period over midnight in 24hr format

Hi! I am seeing both 12hr and 24hr time format in sipxconfig and that is a bit confusing sometimes but no major problem. (User schedules 24hr format and autoattendant schedules 12hr format) Using 24hr format i can not: -Set up a schedule/working time from 19:00 to 07:00. "The start time of o

Re: [sipx-users] SipXbridge 5060 and sipXpbx 5060 how?

FWIW to turn off the sip rewriting code (if its loaded): rmmod nf_nat_sip rmmod nf_conntrack_sip You can check if its loaded with 'lsmod | grep sip'. Depending on what platform you're on and what software is installed wrt the platform being a 'firewall' the instructions for disabling the SIP na