[sipx-users] Eyebeam intergrate with SIPX

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <54512> Message-ID: I am using Eyebeam to register with SIPX. My Eyebeam's Presence is set to "Peer to Peer" Currently i am using two Eyebeams to

Re: [sipx-users] Emergency DNS question - HELP

Nathaniel, First off, get a backup. I may not be the best for this round of advise, but this is what I would try at this late hour. I'd run sipXecs-setup and change the DNS servers. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundr

[sipx-users] Emergency DNS question - HELP

Our 911 guys just called and... My network guy apparently moved some servers around - which includes my DNS server for sipx (it is not being managed on it's box - might change that moving forward) - it is being managed by a windows box. Well - the IP I was using for that is no longer being used

Re: [sipx-users] wireless SIP phones - opinions

i have had an enormous amount of problems with the snom m3's. the hardware is poor and the stack is abysmal. if you want an idea of what you'll be dealing with, take a look at the firmware bugfix inventory. i moved to kirk handsets with kws-300 servers and haven't looked back. it's more than 2x

[sipx-users] Inbound Verizon Callers

On Thu, 18 Nov 2010, Bryan Anderson wrote: > All phones get their IP's from the DHCP server distributed > on the SipXecs ISO. It is not unheard of for a MAC address, which is used to try to ensure uniqueness of IP assigmment to be duplicated by a hardware vendor [I have some specialty hardware

Re: [sipx-users] Sipx high availability with Mediant

On 11/18/2010 2:34 AM, Michael Picher wrote: Well, he stated the following: I am testing with is routed to dest IP address "2ic.ca " which is the domain sipx is configured for. So I would go with Firmware or gateway settings do not have lookup via SRV configured proper

Re: [sipx-users] Sipx high availability with Mediant

On 11/17/2010 6:38 PM, Jim Canfield wrote: > On Wed, Nov 17, 2010 at 7:15 PM, Chris St Denis > wrote: > >> My DNS records are setup as what the DNS Advisor provided, and a dig -t SRV >> _sip._udp.2ic.ca appears to give correct values >> >> _sip._udp.2ic.ca. 10800 IN SRV 1 0 5060

Re: [sipx-users] Transferring offsite disconnects

Thanks Both. Wireshark was on my list to try. I just need to wait until a maintenance window so I can transfer the DID and re-test. I will create a SPAN on on my Core switch to gather the traffic. I will give a URI point to the Cisco system a try as well, however, I believe it will only accept the

Re: [sipx-users] Inbound Verizon Callers

All phones get there IP's from the DHCP server distributed on the SipXecs ISO. Bryan On Thu, Nov 18, 2010 at 1:36 PM, Tony Graziano wrote: > The phone is getting its ip from dhcp (#8)? > > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Emai

Re: [sipx-users] Inbound Verizon Callers

The phone is getting its ip from dhcp (#8)? 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

Re: [sipx-users] Inbound Verizon Callers

It does make a difference. Unplugging phone number 8 calls start working again. If we deley one or more phones to ring "if no answer", the calls also work. Bryan On Thu, Nov 18, 2010 at 1:03 PM, Todd Hodgen wrote: > Not likely. IMHO, if you look at your configuration between sipXecs and > t

Re: [sipx-users] Transferring offsite disconnects

This sounds like it could be a permissions issue. What you might try doing is instead of just entering a standard number such as 8461234 is to append the ip or dns name of the CUCM such as 8461...@10.1.10.10 and see if that gets you anywhere. From: sipx-users-boun...@

Re: [sipx-users] Inbound Verizon Callers

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 woul

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 wrote: > Okay, > > NexVortex "looked" into it

Re: [sipx-users] Polycom 501s and DST

I think they adjust on their own. You may just need to set the dates, but i think those are set properly in the configs... take a look at the config file generated for the 501 and compare it to a working 6XX. Mike On Thu, Nov 18, 2010 at 2:52 PM, Becker, Jesse wrote: > Mike, >Thank you for

Re: [sipx-users] Transferring offsite disconnects

I think the place you'll have to start is capturing the event in a wireshark (I would assume at the router would be the best place). You'll spend a lot of time speculating until you do that. Mike On Thu, Nov 18, 2010 at 2:49 PM, Becker, Jesse wrote: > All, >Our college is currently using Ci

Re: [sipx-users] Polycom 501s and DST

Mike, Thank you for your reply. I will test this mod tonight and will re-send profile to the phones. Do I need to remove the DST 1 when DST has finished or will the phones adjust on their own? Thanks, Jes On Thu, Nov 18, 2010 at 2:35 PM, Michael Picher wrote: > I had noticed a bug in an ear

[sipx-users] Transferring offsite disconnects

All, Our college is currently using Cisco Unified Communications (V8) for all phones. I was recently granted approval to pilot SipX as an alternative in our IT department by the college administration. I am currently trying to replicate our Helpdesk auto attendant (currently house on Cisco Unity

Re: [sipx-users] Inbound Verizon Callers

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 re

Re: [sipx-users] Polycom 501s and DST

I had noticed a bug in an earlier build with the 550 phones (wonder if the same one is there for 501's)... I created a group for the 550's and set the DST flag to 1 to fix it. The templates for the other phones was correct... This is corrected in the 4.3 dev branch. Mike On Thu, Nov 18, 2010 a

Re: [sipx-users] Messages button for Polycom 335/331/321

Thank you both for your replies. Would it be worth having development add the sip.cfg mod to the phone group configuration or should one proceed with simply setting up the message number as the first speed dial. I avoid editing any of the tftp config files and they get overwritten the next time you

[sipx-users] Polycom 501s and DST

All, At another site I helped setup, they are using some older Polycom 501s. Currently their phones are all displaying a hours ahead of what they are suppose to be. I checked the config for the phone group and it is setup the sames as my lab and production environment here. The server has the corr

Re: [sipx-users] Messages button for Polycom 335/331/321

On Thu, Nov 18, 2010 at 11:44 AM, Becker, Jesse wrote: > All, >    I am currently experimenting with the Polycom 335s, 331s, and 321s. They > have the ability to either have a second line button or it can be replaced > with an included Messages button cap. Is there a configuration options in > sip

Re: [sipx-users] [sipx-dev] Rally for Polycom 3.3.0 support.

On Thu, Nov 18, 2010 at 10:03 AM, Douglas Hubler wrote: > > 1.) Test manually configured phone for compatibility w/sipxecs, >  required skill of TESTER = 5 >  size of effort for TESTER = 3 > Is the this scale graded 1-5? This part would be fairly straight forward using the Polycom conversion to

Re: [sipx-users] Messages button for Polycom 335/331/321

My Polycom 320 works as you describe. I believe all I had to do was put the voicemail number as the first speed dial, and ensure that the "Subscribe to Presence" box was NOT checked. Jeff On Nov 18, 2010, at 12:44 PM, Becker, Jesse wrote: > All, >I am currently experimenting with the Polyc

Re: [sipx-users] Custom Polycom Ring Tones

Depends on the phone. The HD models support standard wav (slin, araw) files @ 16kHz, the non-HD models only support 8kHz I have some samples and instructions here: http://sites.google.com/site/sipxecstipsandtricks/polycom-phones They are all in 16 kHz so they'll only work on the HD models (650,

[sipx-users] Custom Polycom Ring Tones

All, Does anyone have a solid methods for creating custom ring tones for the Polycom phones (650,550,430,335,321)? I have the specs from Polycom's site, however, I have inconsistent results. I have used both Audacity and sox with mixed results. Some work some do not. Thanks, Jes -- *Jesse

[sipx-users] Messages button for Polycom 335/331/321

All, I am currently experimenting with the Polycom 335s, 331s, and 321s. They have the ability to either have a second line button or it can be replaced with an included Messages button cap. Is there a configuration options in sipxecs to enable the messages button or should one simply make the c

Re: [sipx-users] choose codec on network

yea, sorry, was thinking one thing (park) and wrote another... On Thu, Nov 18, 2010 at 11:41 AM, Josh M. Patten wrote: > http://wiki.sipfoundry.org/display/sipXecs/HD+Music+on+Hold I beg to > differ about MoH not being in FreeSWITCH. > > > > *From:* sipx-users-boun...@list.sipfoundry.org [mailto:

Re: [sipx-users] Nice new SIP Softphone

Seems to work OK on my XP machine. Other things I miss: - Video (they are working on it) - XMPP - Desktop sharing for Windows (working on it) I will certainly keep an eye on them. Paul Scheepens Jan Fricke wrote > Works out-of-the-box on my Win7 32 bit. Easy to configure. Bonjour > integratio

Re: [sipx-users] choose codec on network

http://wiki.sipfoundry.org/display/sipXecs/HD+Music+on+Hold I beg to differ about MoH not being in FreeSWITCH. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher Sent: Thursday, November 18, 2010 9:35 AM To: Discussion list fo

Re: [sipx-users] [sipx-dev] Rally for Polycom 3.3.0 support.

On Thu, Nov 18, 2010 at 7:32 AM, Jim Canfield wrote: > I'm not sure how much work is actively being done (tracker seems > vacant), but I'd like to give things a push in the right direction. > How many community members (and Ezuce) are willing to collaborate on a > focused effort to support the lat

Re: [sipx-users] Rally for Polycom 3.3.0 support.

Let me split that out a little more On a scale of 1 to 10 (10 being most) Time - 8 Resources - programming - 1 testing (phones, servers, sip trunk, etc) - 10 Money - 1 On 11/18/2010 6:37 AM, Matthew Kitchin (Public) wrote: > On a scale of 1 to 10 (10 being most) > Time - 8 > Resources - 4

Re: [sipx-users] choose codec on network

If you had a gateway device to gateway device type configuration you could force that gateway to only G.729 and as long as your phones support it you should be fine you'd also need a couple G.729 licenses for freeswitch so users could leave VM / etc. MoH wouldn't work across there because it'

Re: [sipx-users] choose codec on network

Since codec negotiation is a function of the phones this is really a phone question, but in any case I don’t think this is currently possible short of using an external SBC that supports transcoding. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On

Re: [sipx-users] Rally for Polycom 3.3.0 support.

Using the scale (1-least, 10-most): Time - 2 Resources - 2 Money - 7 Philippe On Thu, Nov 18, 2010 at 7:32 AM, Jim Canfield wrote: > Hi folks, > > I'm not sure how much work is actively being done (tracker seems > vacant), but I'd like to give things a push in the right direction. > How many c

Re: [sipx-users] [sipx-dev] Rally for Polycom 3.3.0 support.

We can provide testing on 650 550 450 335 330 Am 18.11.2010 um 13:32 schrieb Jim Canfield: > Hi folks, > > I'm not sure how much work is actively being done (tracker seems > vacant), but I'd like to give things a push in the right direction. > How many community members (and Ezuce) are willing

Re: [sipx-users] Rally for Polycom 3.3.0 support.

On 11/18/2010 7:32 AM, Jim Canfield wrote: > Hi folks, > > I'm not sure how much work is actively being done (tracker seems > vacant), but I'd like to give things a push in the right direction. > How many community members (and Ezuce) are willing to collaborate on a > focused effort to support the

[sipx-users] choose codec on network

Hi all, I’m trying to see if something we would like to do is possible. The case: I have one HA SipXecs setup on network 1. I have multiple networks with phones in them, let’s say network 2 and 3. I don’t use call admission control YET, but I want to control my data due to a possible bandw

Re: [sipx-users] Rally for Polycom 3.3.0 support.

I have a full testbed I can use to test 3.3.x on. I have just about every polycom model (335, 450, 550, 650, and vvx-1500) in the testbed and it's not in production so I can screw it up all I want :-) -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-bou

[sipx-users] Music on Hold - Linksys SPA-3000

[cid:imageadb320.GIF@fa3f530c.448f6fe3] Hi, Can anybody tell me how to configure music on hold on a Linksys SPA-3000 with Sipx 4.2.1? The only option I see in the SPA config is ‘MOH server’ into which I have entered ~~...@btg-supplies.local – but there is no

Re: [sipx-users] Changing login screen

That did the trick, thanks Jan. Vielen dank. Sven From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jan Fricke Sent: 18 November 2010 12:21 To: Discussion list for users of sipXecs software Subject: R

Re: [sipx-users] Rally for Polycom 3.3.0 support.

On a scale of 1 to 10 (10 being most) Time - 8 Resources - 4 Money - 1 --Original Message-- From: Jim Canfield Sender: sipx-users-boun...@list.sipfoundry.org To: sipx-users@list.sipfoundry.org To: sipXecs developers ReplyTo: Discussion list for users of sipXecs software Subject: [sipx-users

[sipx-users] Rally for Polycom 3.3.0 support.

Hi folks, I'm not sure how much work is actively being done (tracker seems vacant), but I'd like to give things a push in the right direction. How many community members (and Ezuce) are willing to collaborate on a focused effort to support the latest firmware? I currently have some very skilled j

Re: [sipx-users] Changing login screen

Hi Sven, have a look at /usr/share/java/sipXecs/sipXconfig/sipxconfig.war It’s a zip file which contains some directorys. WEB-INF/LoginPage.properties should contain what you are looking for. Regards Jan *Von:* sipx-users-boun...@list.sipfoundry.org [mailto: sipx-users-boun...@list.

[sipx-users] Changing login screen

On the Login Screen (sipX 4.2.1) we can see "WELCOME TO SIPXECS". Our customer wants a different text there. Does anyone know how to do that. I have been playing around with color.css and layout.css, but I cannot find where that text string is. Sven Sven Evensen, Technical Support Consul

Re: [sipx-users] Nice new SIP Softphone

Works out-of-the-box on my Win7 32 bit. Easy to configure. Bonjour integration is a nice feature ;-) Things I miss: - MoH - Attended/Blind Transfer Regards Jan -Ursprüngliche Nachricht- Von: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] Im Auft

Re: [sipx-users] Nice new SIP Softphone

On 11/18/2010 1:41 AM, Tony Graziano wrote: > you must be using a mac, always crashes on my windoze boxes. Win7 64bit. No problems at all. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-u

Re: [sipx-users] Sipx high availability with Mediant

Well, he stated the following: I am testing with is routed to dest IP address "2ic.ca" which is the domain > sipx is configured for. > So I would go with Firmware or gateway settings do not have lookup via SRV configured properly, or there's a timeout issue (like what your phones are experiencing