Re: [sipx-users] Fax failure(s) and one method to avoid them.

> Call the fax extension and hangup (no negotiation or data transfer). You > will mow get an email that says "Your (0) page fax was received" when a fax > comes in. The same is also true when a 5 of 6 page fax comes in, it will > "send" whatever it receives, and indicate the number of pages receive

[sipx-users] Polycom station delay ring issue

Hey folks, We've been fighting a strange issue where gateway calls (patton 4960 pri) are taking 2-5 seconds before they begin to ring. Display shows the incoming call and it can even be picked up. Internal calls work fine. Oddly, the second call placed will work fine as long as it is shortly aft

[sipx-users] Peak call volume report.

Hey folks, I have been asked several times to help provide a report that will show max concurrent calls both internally and for trunking. Is there a way to generate a report for max concurrent calls? I figure I may be able to get the gateways to provide this info in the TDM environments. Just

Re: [sipx-users] Call park with presence in 4.4

On Tue, Apr 26, 2011 at 10:33 AM, Matthew Kitchin (public/usenet) < mkitchin.pub...@gmail.com> wrote: > On 4/26/2011 10:24 AM, Jim Canfield wrote: > > On Tue, Apr 26, 2011 at 10:17 AM, Matthew Kitchin (public/usenet) < > mkitchin.pub...@gmail.com> wrote: > >>

Re: [sipx-users] Call park with presence in 4.4

On Tue, Apr 26, 2011 at 10:17 AM, Matthew Kitchin (public/usenet) < mkitchin.pub...@gmail.com> wrote: > Thanks. That kind of stinks. I guess I am out of luck on this one. > I could work around it if they weren't treated as one group. > > ya, this is unfortunate. Would be nice if you could slide t

Re: [sipx-users] Call park with presence in 4.4

On Mon, Apr 25, 2011 at 4:47 PM, Matthew Kitchin (public/usenet) < mkitchin.pub...@gmail.com> wrote: > On 4/25/2011 4:31 PM, Jim Canfield wrote: > > On Mon, Apr 25, 2011 at 4:30 PM, Jim Canfield wrote: > >> >> >> Actually, you can turn that off the hold, it&

Re: [sipx-users] Call park with presence in 4.4

On Mon, Apr 25, 2011 at 4:30 PM, Jim Canfield wrote: > > > Actually, you can turn that off the hold, it's redundant. Translated = ..turn off the hold. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.

Re: [sipx-users] Call park with presence in 4.4

On Mon, Apr 25, 2011 at 1:12 PM, Matthew Kitchin (public/usenet) < mkitchin.pub...@gmail.com> wrote: > park extensions (1,2,3). > > Very cool. I see that it gets rid of the "End Call" soft key and you have > to select More to get the additional parking slots. Have your users had any > trouble wit

Re: [sipx-users] Call park with presence in 4.4

On Thu, Apr 21, 2011 at 6:19 PM, Matthew Kitchin (public/usenet) wrote: > I see the examples, but does anyone have a sample or some specifics on > exactly what they use for EFK and parking calls? I don;t want to do this > wrong. FYI, if you don't know already. The custom configuration is now bui

Re: [sipx-users] Call park with presence in 4.4

On Thu, Apr 21, 2011 at 6:19 PM, Matthew Kitchin (public/usenet) wrote: > I see the examples, but does anyone have a sample or some specifics on > exactly what they use for EFK and parking calls? I don;t want to do this > wrong. Hare's a template I use quite a bit. It assumes three single digit

Re: [sipx-users] Basic OpenACD questions on 4.4

On Thu, Apr 21, 2011 at 9:13 AM, Jim Canfield wrote: > I have tried both ways.  The error seems to be directly related to how > I setup the endpoint field in the AGENT_UI. If i set my "VoIP endpoint > data" to 200 I get this... > Okay I got ACD working with

Re: [sipx-users] Basic OpenACD questions on 4.4

On Thu, Apr 21, 2011 at 8:55 AM, Laurentiu Ceausescu wrote: >> >> It's "emstarvoip.com" actually (the .org was a typo on my part).  I >> changed everything to match and I'm still getting this crazy >> underscore in the URI. > > I was wrong earlier .. I wanted to say that '2...@sipx.emstarvoip.com

Re: [sipx-users] Basic OpenACD questions on 4.4

On Thu, Apr 21, 2011 at 4:24 AM, Laurentiu Ceausescu wrote: > I think the correct value should be 2...@sipx.emstarvoip.org > What is your Domain name ( from System / Domain page)? > In freeswitch.log I have ... It's "emstarvoip.com" actually (the .org was a typo on my part). I changed everythin

Re: [sipx-users] Basic OpenACD questions on 4.4

On Wed, Apr 20, 2011 at 1:17 PM, Jim Canfield wrote: > Even with those issues fixed,  still cannot to a RING_TEST.  Looking > at the log, this line seems very strange to me: > > Test',origination_caller_id_number=000,hangup_after_bridge=true,origination_uuid=4b816a

Re: [sipx-users] Basic OpenACD questions on 4.4

On Wed, Apr 20, 2011 at 12:04 PM, Jim Canfield wrote: > On Wed, Apr 20, 2011 at 10:50 AM, Laurentiu Ceausescu > wrote: > >> Did you enable the FreeSWITCH module using openACD interface? >> (please see http://wiki.sipfoundry.org/display/sipXecs/OpenACD+Setup >>

Re: [sipx-users] Basic OpenACD questions on 4.4

On Wed, Apr 20, 2011 at 10:50 AM, Laurentiu Ceausescu wrote: > Did you enable the FreeSWITCH module using openACD interface? > (please see http://wiki.sipfoundry.org/display/sipXecs/OpenACD+Setup > 'OpenACD to FreeSWITCH SIP communication' paragraph) > What is the value for C Node and Dial String

Re: [sipx-users] Basic OpenACD questions on 4.4

On Wed, Apr 20, 2011 at 8:34 AM, Douglas Hubler wrote: > > - Can you elaborate with more technical details on "cannot answer". > - Can you review logs for errors. >  /var/log/sipxpbx/openacd/*.log >  /var/log/sipxpbx/freeswitch.log > - Once you log into agent console on port 5050, try a ring test.

Re: [sipx-users] Basic OpenACD questions on 4.4

On Tue, Apr 19, 2011 at 3:21 PM, Douglas Hubler wrote: > > sorry, this is not clear at all !  Let me restate this: >  - Change an agent's Security level to "SUPERVISOR" as found in the > Call Center Agent's Edit page.  Do *not* make the user superadmin > privledges in the User Edit page. Ha. I

[sipx-users] Basic OpenACD questions on 4.4

Hi folks, Wanted to start getting familiar with openACD this morning and I have a couple basic questions: 1) Where do you see queue status? 2) When writing commands, what is the proper format?  For example the wiki has the following for "Login": agent_dialplan_listener testme@fedorabox age

Re: [sipx-users] Yumming from 4.2.1 to 4.4 beta

On Mon, Apr 18, 2011 at 5:53 AM, Tony Graziano wrote: > This happened to more than one person. Jim also posted about it. > > I figured out after that fact that I was missing an update on 4.2.1 that caused this. I think as long as users fully upgrade the 4.2.1 install before changing repos, this

Re: [sipx-users] Polycom custom config issues in 4.4?

On Sun, Apr 17, 2011 at 9:33 PM, Douglas Hubler wrote: > On Fri, Apr 15, 2011 at 3:15 AM, Nathaniel Watkins > wrote: > > Is anyone having trouble using Eric & Josh's Custom polycom config - > http://sites.google.com/site/sipxecstipsandtricks/polycom-phones > > This is how 4.4 implemented the cus

Re: [sipx-users] 4.4 Upgrade dependency issues.

On Fri, Apr 15, 2011 at 12:23 PM, Jim Canfield wrote: > Hey folks, > > Trying to figure out dependency issues with the 4.2.1-->4.4.0 upgrade. > Just a follow up: #yum remove sipx-freeswitch-codec-passthru-g729 #yum upgrade This fixed the dependenc

[sipx-users] 4.4 Upgrade dependency issues.

Hey folks, Trying to figure out dependency issues with the 4.2.1-->4.4.0 upgrade. --> Finished Dependency Resolution sipxecs-4.2.1-18591.9.1.i386 from installed has depsolving problems --> Missing Dependency: sipx-freeswitch-codec-passthru-g729 is needed by package sipxecs-4.2.1-18591.9.1.i386

Re: [sipx-users] Openfire admin console on 4.2.1

On Fri, Apr 8, 2011 at 3:46 PM, Douglas Hubler > > do you have any details on what didn't "work"? > The wiki commands ran fine. Restarted sipXecs and port 9094 and 9095 port both resolve to a blank web page with a couple of odd looking characters.. ___

Re: [sipx-users] Openfire admin console on 4.2.1

My service is running fine, just can't access. Park Running Primary SIP RouterInstant Messaging

[sipx-users] Openfire admin console on 4.2.1

Can anyone confirm that the Openfire admin access is broken on 4.2.1 http://wiki.sipfoundry.org/display/sipXecs/The+Openfire+admin+console This procedure no longer seems to work. Thanks, --Jim ___

Re: [sipx-users] Fax capabilities and Patton Gateway

__ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Jim Canfield Emstar Solutions, LLC Phone: 918-851-2633 Support: http://support.emstar.com Web: http://emstar.com ..IT is

Re: [sipx-users] sipdialing uri's that start with numbers get 403 requires noaccess

On Fri, Mar 18, 2011 at 1:56 PM, Michael Scheidell wrote: > I seem to have problems sip dialing uri's that start with numbers. > (itad dialing works) Works fine on 4.2.1 here at the office. Also on Voip.ms. ___ sipx-users mailing list sipx-users@list.si

Re: [sipx-users] ProCurve LLDP bug

On Mon, Feb 21, 2011 at 6:14 PM, Josh M. Patten wrote: > Just to let everyone know on Procurve switches model 2610 switches there is > an LLDP bug in firmware versions above R.11.54 that prevents phones from > getting proper VLAN configuration information. > > Info found here after pulling my hair

Re: [sipx-users] choose outgoing gateway depending on user id

The code Nate provided will create your sip interconnect, you should now be able to apply your mapping tables and routes which is just a matter of pointing the to and from interfaces or tables. For example: route call dest-table FXS_FAX_PSTN might be... route call dest-interface IF-BRI-LOCATION

Re: [sipx-users] choose outgoing gateway depending on user id

On Tue, Mar 15, 2011 at 3:12 PM, Claas Hilbrecht wrote: > > Has anyone done this before with a patton gateway? I'm somewhat stuck now > how I can route a call from one patton gateway to another. > We do it all the time. Patton routing/mapping tables are amazingly flexible. -Jim ___

[sipx-users] IM status mutes PC audio.

Hey Folks, I'm currently seeking an IM plug-in (Pidgin or Spark) that will mute PC audio when the user is "On the phone". Not sure if this exists, but it seems like it could be a doable solution. If anyone knows of such a plug-in. I'd be very interested. Thanks, --Jim

Re: [sipx-users] Change RTP NAT ports to 10000-20000 for Vitelity SIP trunk conflicts with Java port

On Fri, Feb 25, 2011 at 6:55 PM, Tim Ingalls wrote: > I'm using Vitelity for an ITSP. They require that RTP ports 1-2 be > open. Unfortunately, this conflicts with the Java app port of 15060. The > result is that the sipXrelay service doesn't start because it can't bind to > the proper por

Re: [sipx-users] Polycom 450 PoE issue

On Tue, Feb 22, 2011 at 8:50 AM, Michael Picher wrote: > Yea, I think L is the rev you want on these and rev K doesn't work (or it > was the other way around)... > I have rev "K" on my DOA's so that sounds correct. ___ sipx-users mailing list sipx-users

Re: [sipx-users] Polycom 450 PoE issue

On Mon, Feb 21, 2011 at 6:46 PM, Josh M. Patten wrote: > I've never had this issue before until this recent shipment of 450's I got. > The first two I tried would boot up just fine the first >time you plugged > them in, then if you unplugged them and tried to plug them back in PoE would > cease

Re: [sipx-users] Patton 5200 SBC Template

Thanks for doing that Mike. I also have a working broadvox config. Just wish I had more spare time to get all these things documented. On Sat, Feb 19, 2011 at 5:19 AM, Michael Picher wrote: > Added it...  Jim, tweak at will... > ___ sipx-users mailing l

Re: [sipx-users] Polycom custom config files patch

On Fri, Feb 18, 2011 at 2:46 PM, Josh M. Patten wrote: > Hey if anyone uses the custom config files patch written by Eric Varsanyi > that I’ve been posting about lately and would like it to appear in 4.4 > without custom modification, please vote on the issue here: > http://track.sipfoundry.org/br

Re: [sipx-users] dual certs, single domain

On Thu, Feb 17, 2011 at 3:09 PM, m...@grounded.net wrote: > I have a situation where I need to have a sipx server on two different > domains. http://help.godaddy.com/article/3908 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive:

[sipx-users] Patton 5200 SBC Template

Hey folks, Finally have a working Patton 5200 SBC config working on Voip.ms as B2BUA. Here's a rough draft template for anyone who might be interested. -- Jim Canfield Emstar Solutions, LLC Phone: 918-851-2633 Support: http://support.emstar.com Web: http://emstar.com ..IT is about se

Re: [sipx-users] screaming monkeys still work?

On Fri, Feb 11, 2011 at 2:01 PM, Michael Scheidell wrote: > was doing regression testing, and I usually try the screaming monkeys. > > sip:1...@loligo.com > > or > ISN:  1234*256 > Nope, John's Monkeys appear to be dead. ___ sipx-users mailing list sipx

Re: [sipx-users] 25 Second Hunt group forwarding failures via ITSP.

On Tue, Feb 8, 2011 at 12:26 PM, Michael Picher wrote: > is this a login voip.ms account or routed? If by login you mean registered, yes. > > if routed, is the call coming in on 5080? I'm not sure what you mean by routed, but it is registered on port 5080. Thanks! -- Jim C

Re: [sipx-users] 25 Second Hunt group forwarding failures via ITSP.

On Sun, Feb 6, 2011 at 2:39 PM, Tony Graziano wrote: > 8200 will not work. Actually, it does work. I think this was fixed recently. It doesn't matter where I send the fall-back, it's the same result. ___ sipx-users mailing list sipx-users@list.sipfound

[sipx-users] 25 Second Hunt group forwarding failures via ITSP.

Hi Folks, Curious if anyone else is having this issue... ITSP DID rings HG (300) HG 300 Rules - Ring 200 or 16 Seconds - If no answer ring 201, 202, 203 for 16 Seconds - If no answer fall-back to 200 mailbox (8200) It seems, no matter what I do, the sum of the timers cannot exceed 24 seconds

Re: [sipx-users] coonecting to leagacy pbx

Never mind, I should have read the rest of the thread. :) On Sun, Jan 30, 2011 at 10:23 AM, Jim Canfield wrote: > On Wed, Jan 26, 2011 at 7:07 AM, Matt White wrote: >> I have a client that would like to connect sipx to a siemens Highpath pbx at >> a remote branch. >> &g

Re: [sipx-users] coonecting to leagacy pbx

On Wed, Jan 26, 2011 at 7:07 AM, Matt White wrote: > I have a client that would like to connect sipx to a siemens Highpath pbx at > a remote branch. > > Has anybody done any work to integrate these two?  They have open analog > ports but I don't think a patton FXO can really be used to bridge the

Re: [sipx-users] SBC for port translation

On Fri, Jan 28, 2011 at 11:08 AM, Josh M. Patten wrote: > Patton has SBC's for this very purpose. The Smartnode 5200 and 5400 (5400 > does transcoding) yep! We have a 5200 on the bench right now. Will post a working config when it's complete. ___ sipx-

[sipx-users] Creating a sipXecs USB installer in 3 minutes.

anually once booted. - Windows 7 users must "run as administrator" Enjoy! Ref: http://www.pendrivelinux.com/usb-centos-5-live-install-via-windows/ -- Jim Canfield Emstar Solutions, LLC Phone: 918-851-2633 Support: http://support.emstar.com Web: http://emstar.com

Re: [sipx-users] FYI: Debugging Registration Issues with a New Install and ITSP

On Tue, Dec 28, 2010 at 12:42 PM, Gerald Drouillard wrote: > Have a look at: > > http://malphx.free.fr/dotclear/index.php?post/2010/08/17/Playing-with-SIP-NMAP-and-NSE > > Needs nmap 5.0 or greater: http://nmap.org/download.html > Works with/without sipxecs running. No need to stop it. > > Grea

[sipx-users] Csipsimple android client

Well after downloading several sip clients on my android I finally found a winner: http://code.google.com/p/csipsimple/ SRV - Works! Audio - Works! ISN - Works! Video - not yet. UI - Awesome Makes the new Google SIP stack in 2.3 look stupid actually. Download, donate, and enjoy. -Jim __

Re: [sipx-users] Latest and greatest method to park/distribute a call

On Fri, Dec 10, 2010 at 3:59 PM, Matthew Kitchin (public/usenet) wrote: > With Sipx, Verizon sip trunk, and Polycom 450/550s, we have been parking > the call on 41, 42, 43, etc, then paging "Bob you have a call on *41" > This has not always gone over well. They get used to it, but prefer > their

Re: [sipx-users] Call Forwarding

On Fri, Dec 3, 2010 at 11:37 AM, Michael Picher wrote: > Also, if your phone loses power you lose the forwarding... > For those inclined, it would not take much to create a freeswitch extension script that collects forward digits and calling party and pass it to another script (perl, php, python

[sipx-users] Polycom 3.3.x Testing Notes

Hello sipXers, Started playing with the Polycom 3.3.1 firmware this evening thought I'd share a few notes. It's not much, but I hope it's a start towards getting familiar with the new configuration changes. It's become apparent that testing this against the 4.2.x release would be a waste of resou

[sipx-users] Netselect: Useful tool for picking reliable host.

/netselect/ ...I'll be adding this to the toolbox. -- Jim Canfield Emstar Solutions, LLC ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

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

On Tue, Nov 23, 2010 at 7:41 AM, Martin Steinmann wrote: > +1   Are you organizing this? > --martin Absolutely. I'd like to shoot for next Friday AM. I'd like to hear from community members who will be able to attend. The holiday weekend should provide some time to play with the new firmware.

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

On Fri, Nov 19, 2010 at 2:03 PM, Martin Steinmann wrote: > > If we were to take the lead on this, it would be very helpful to get some > financial contributions as was discussed in this thread.  We can make > available a PayPal account.  The cost for this will be around $ 5.000 - $ > 6.000.  If s

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

[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] Sipx high availability with Mediant

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 pbx2.2ic.ca. > _sip._udp.2ic.ca.   10800   IN

Re: [sipx-users] Nice new SIP Softphone

On Wed, Nov 17, 2010 at 7:05 PM, Gerald Drouillard wrote: > http://icanblink.com/index.phtml > Very nice looking. Can't wait to try it out. ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx

Re: [sipx-users] Enable Call Forward

On Tue, Nov 16, 2010 at 4:36 PM, Tony Graziano wrote: > You do not need efk to do this. You can enable it in sipxconfig but it only > remembers the "last" entry. This is not limited to EFK. It's ANY custom config options that may not be included in sipXconfig. ___

Re: [sipx-users] Enable Call Forward

On Tue, Nov 16, 2010 at 4:23 PM, Josh M. Patten wrote: > http://sites.google.com/site/sipxecstipsandtricks/polycom-phones > > once you've got that set up you can add the following line in the custom > config: > > divert.fwd.3.enabled="1" divert.fwd.4.enabled="1" divert.fwd.5.enabled="1" > diver

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

On Tue, Nov 16, 2010 at 3:27 PM, Josh M. Patten wrote: > I have some Polycom kirk DECT phones and a small kirk wireless server 300 > that work well with sipX, though there is no auto-provisioning available for > them. They are pretty simple to set up though. They don't have feature parity > wit

Re: [sipx-users] External audio player

On Thu, Nov 11, 2010 at 4:38 AM, George Niculae wrote: > On Thu, Nov 11, 2010 at 12:31 PM, Nikolay Kondratyev wrote: >> I just tried to listen to vm message using Chrome. >> It shows buildin player picture, but nothing is audible. >> Rgds, >> Nikolay. > > Thanks, will provide a fix with http://tr

Re: [sipx-users] Valcom VIP-201 not receiving calls or pages [SOLVED]

>  It's 2.13 firmware...so I'm hoping an upgrade to 2.15 will > fix the "every-other" behavior on the 204 as well.  I'll keep you > posted, maybe we can add this to the "Fully supported" list. > Update: Just got off the phone with Allan at Valcom. He apparently does the in-house SipXecs testing.

Re: [sipx-users] Valcom VIP-201 not receiving calls or pages [SOLVED]

On Thu, Oct 28, 2010 at 1:10 PM, Tony Graziano wrote: > Solved but not answered -- does this answer every time in either mode > (direct dialing or page group now). > If so, a wiki page on configuring might be a good idea with a link or notice > with the firmware version that ought to be applied to

Re: [sipx-users] Valcom VIP-201 not receiving calls or pages [SOLVED]

-users-boun...@list.sipfoundry.org] On Behalf Of Jim Canfield > Sent: Thursday, October 28, 2010 8:54 AM > To: Discussion list for users of sipXecs software > Subject: Re: [sipx-users] Valcom VIP-201 not receiving calls or pages > > On Thu, Oct 28, 2010 at 8:31 AM, Danny Shay wrot

Re: [sipx-users] Garrett County Fax

On Thu, Oct 28, 2010 at 9:22 AM, Michael Picher wrote: > Dittos...  Take the NEC out of the picture. Nate...I think that is a big fat "Told you so!" ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/arch

Re: [sipx-users] Valcom VIP-201 not receiving calls or pages

On Thu, Oct 28, 2010 at 8:31 AM, Danny Shay wrote: > I have a VIP-201 that is properly registering with the server, but it does > not receive pages as part of the page group.  I made another page group with > just the valcom in the group and it does not play the page either.  When I > called the b

[sipx-users] Park can't be picked up on BLF after 60 seconds.

Just confirmed with a customer that Parks cannot be picked up on a BLF after 60 seconds. You can however pick up the park extension with *4### or *78### manually. I've also confirmed this 60 second behavior spanning ringing hunt groups. Can anyone shed a little light on what system timer is at pla

Re: [sipx-users] t.38 testers please...

On Sat, Oct 23, 2010 at 8:20 AM, Tony Graziano wrote: > And that is my point. > > The receiving t.38 sends a reinvite to switch from g711u to t.38. > > When it is sent through a PRI (pstn>pri>sipx) it negotiates t.38 without an > issue because the patton (pri) offers on t.38 first. > > So I think

Re: [sipx-users] SIP Door Phone or Entry Phone

Valcom VIP-172L ...but you get what you pay for. IMHO your time can end up costing much more when fighting cheap equipment. http://www.valcom.com/Home_links/sipdoorintercom.htm On Oct 22, 2010 2:42 PM, "Morgan, Michael" wrote: ___ sipx-users mailing l

Re: [sipx-users] Best options for Video from a VVX to a softphone (hooked to a projector)

On Wed, Oct 20, 2010 at 4:29 PM, Jim Canfield wrote: > On Wed, Oct 20, 2010 at 1:10 PM, Todd Hodgen wrote: > > Check out this one...I have not tried it yet though. > > http://www.mirial.com/products/Mirial_Softphone_HD.html > Okay installed the mirial demo...first impressions

Re: [sipx-users] Best options for Video from a VVX to a softphone (hooked to a projector)

On Wed, Oct 20, 2010 at 1:10 PM, Todd Hodgen wrote: > I’ve used the VVX to Counterpath Xlite for demonstrations successfully.   It > worked without any hitch. > Ya, but video is U-G-L-Y. I'd be looking for good H.264 support. This is new for the Bria Pro, but still doesn't looks anything like a

Re: [sipx-users] t.38 testers please...

On Tue, Oct 12, 2010 at 5:07 AM, Rene Pankratz wrote: > http://www.faxback.com/ > I use theirt free T.38 Fax plugin for windows. It's really nice as it > installs a FAX Printer driver... > > It works fine with our patton BRI Gateways. > > René > > Very nice René. With 4.3.x and faxworks we should

Re: [sipx-users] VOP Panel monitored park orbits.

Yes, but I've done some updates since I last ran the RPM. I may apply it again. On Mon, Sep 27, 2010 at 1:40 PM, Tony Graziano wrote: > Are you running the patched version of rls? > > ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Arc

[sipx-users] VOP Panel monitored park orbits.

I just finished doing some preliminary testing of the VOP park monitoring and it looks very good. The only odd behavior I found was on the part of the Polycom stations that monitor the park. They never release their status in the RLS server so they remain in a BUSY state in VOP ever after the call

Re: [sipx-users] Improvement Request :: Make ITSP templates a hosted service

On Tue, Sep 21, 2010 at 8:26 AM, Tony Graziano wrote: > Today I added a JIRA improvement request, and expect with the current > roadmap it could be a while before this type of functionality can be > scheduled and added, but at leasta discussion can take place. While it may > not be a model that

Re: [sipx-users] VOIP Panel...

On Mon, Sep 20, 2010 at 11:58 PM, Rene Pankratz wrote: > Jim, > yes we Tested the current version of VOP and those features are working > fine with SipX 4.2.1. > > The only problem we see with VOP is that the CallerID is shown on the > monitor but not on the phone that is ringing (You can use a te

[sipx-users] VOIP Panel...

This is huge...anyone tested it yet? * VOP now supports sipXecs parking extension with monitoring.* You can now transfer a call to a parking extension by a simple drag and drop. You can retrieve a parked call by dialing *4X (where X is the parking extension number). The caller/ee ID will be

[sipx-users] Valcom paging identities only work every other page using sipXpage.

Here's a fun one. When I add a Valcom sip identity to a sipXpage group paging works every other time. Snapshot is attached to the jira. http://track.sipfoundry.org/browse/XX-8905 -Jim ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Arc

Re: [sipx-users] Session timer help needed.

For those interested, here's a quick hack script to set your session timer to -1 (NO-TIMEOUT). It works nicely. #/bin/bash mkdir ~/tmp mkdir ~/sipxbackups cp /usr/share/java/sipXecs/sipXconfig/sipxconfig.war ~/sipxbackups cd ~/tmp jar xvf /usr/share/java/sipXecs/sipXconfig/sipxconfig.war perl -e

Re: [sipx-users] Session timer help needed.

> On Mon, Sep 13, 2010 at 8:00 PM, Tony Graziano >> wrote: >> > Well, that file would have produced the >> > sipxconfig.properties file in /etc/sipxpbx >> > and the parameter you are seeking (which may have changed with post 4.0 >> > builds was... >> > tapestry.adminSessionTimeout=60 >> > tapestry

Re: [sipx-users] Session timer help needed.

On Mon, Sep 13, 2010 at 7:19 PM, Douglas Hubler wrote: > On Mon, Sep 13, 2010 at 8:00 PM, Tony Graziano > wrote: > > Well, that file would have produced the > > sipxconfig.properties file in /etc/sipxpbx > > and the parameter you are seeking (which may have changed with post 4.0 > > builds was..

Re: [sipx-users] Session timer help needed.

es.in" file. I'm not sure if this was ever implemented or perhaps the config schema was changed, but I don't see this in current releases. Any ideas? On Thu, Sep 9, 2010 at 3:14 PM, Jim Canfield wrote: > Hi Folks, > > Looking for the session timer for sipXconfig. Everythi

[sipx-users] Session timer help needed.

Hi Folks, Looking for the session timer for sipXconfig. Everything I read says it should be in the web.xml like so... 60 ... I simply don't know my way around the jar and war files well enough to find the right web.xml file so I can change the session timer. Ideally I would like

Re: [sipx-users] Strange, but good feature request for Polycom phones

_ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://list.sipfoundry.org/archive/sipx-users/ > -- Jim Canfield Emstar Solutions, LLC Phone: 918-851-2633 http://emstar.com ..IT is about service.

Re: [sipx-users] IE8 Voicemail Player - Only external player is supported.

Vote for it if you are interested. http://track.sipfoundry.org/browse/XX-8837 On Thu, Aug 26, 2010 at 12:19 PM, Jim Canfield wrote: > I was afraid that might be the answer. > > > On Thu, Aug 26, 2010 at 11:21 AM, Tony Graziano < > tgrazi...@myitdepartment.net> wrote: >

Re: [sipx-users] IE8 Voicemail Player - Only external player is supported.

I was afraid that might be the answer. On Thu, Aug 26, 2010 at 11:21 AM, Tony Graziano < tgrazi...@myitdepartment.net> wrote: > Use firefox. Is there a user agent switcher for ie8. > > I think its the ability for ie8 (it lacks) to properly interpret this > version of a flash/wav player. > > (Mp3

[sipx-users] IE8 Voicemail Player - Only external player is supported.

Okay what magic needs to happen for IE8 to play messages in the browser? I've tried all the security settings and compatibility mode. -Jim ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-u

Re: [sipx-users] Continued park issues.

On Thu, Aug 26, 2010 at 8:30 AM, Josh M. Patten wrote: > Even at 50 users the current RLS implementation has problems when > changing/adding any user attribute. This encompasses everything from setting > a call forward to changing an email address to adding a new user. Each time > one of these act

Re: [sipx-users] Continued park issues.

I wish I could intelligently describe the issue, but I simply can't duplicate it reliably. Here's my best description: Park Extension = 1 Park 1 Monitored via speed dial with presence checked Call is placed on park (Transfer-->Park BLF-->Hang-up) Call is picked up by pressing the now blinking P

Re: [sipx-users] Continued park issues.

kes. I'm about to do my first 4.2.1 with park. Is it really worse than >> 4.0.4? >> -Original Message- >> From: Jim Canfield >> Sender: sipx-users-boun...@list.sipfoundry.org >> Date: Wed, 25 Aug 2010 15:14:05 >> To: >> Subject: [sipx-users

Re: [sipx-users] Continued park issues.

On Wed, Aug 25, 2010 at 3:23 PM, Tony Graziano wrote: > > I might suggest reaching out to Josh because, he has done VALET and he has > a patch for BLF that he says "might" fix the scalability of the RLS server. > Neither of these does NOTHING for stuck calls in park, but maybe he can show > you a

[sipx-users] Continued park issues.

I guess I'm a gluten for punishment since I added a monitored park at another customer when I should have known better. Of course, it's not going well, in fact, the 4.2 release seems to get stuck more than the 4.0 releases. At any rate, I figured it was time to reach out I see if there is a bette

Re: [sipx-users] [sipx-dev] Welcome to Joegen Baclor

On Tue, Aug 24, 2010 at 10:07 AM, Tony Graziano < tgrazi...@myitdepartment.net> wrote: > A pleasant bit of news for a Monday! > > Ever see the episode of "The office" where Dwight thinks it's Friday and it's actually Thursday? ___ sipx-users mailing lis

Re: [sipx-users] [sipx-dev] VVX-1500 Broken BLA fix.

On Mon, Aug 23, 2010 at 12:20 PM, Tony Graziano < tgrazi...@myitdepartment.net> wrote: > Is there a jira on this? > > http://track.sipfoundry.org/browse/XX-8484 ___ sipx-users mailing list sipx-users@list.sipfoundry.org List

[sipx-users] VVX-1500 Broken BLA fix.

I haven't seen this mentioned yet, but I've been dealing with it recently so I thought I'd pass it along. Music on hold is broken by default for the VVX-1500 for the 4.2.x releases, but the fix is fairly simple. All you have to do is add sip: to the MOH uri under line settings. Doing so will als

Re: [sipx-users] Polycom 3.3.0 Firmware

On Mon, Aug 23, 2010 at 10:55 AM, Tony Graziano < tgrazi...@myitdepartment.net> wrote: > > >> I'm very interested in the some of the new features like dynamic updates >> w/o rebooting the phones. >> >> hopefully! > > i also think distinctive ring could help with parked calls, but once the > park s

[sipx-users] Polycom 3.3.0 Firmware

I've been reading up on the 3.3.0 firmware changes and it brings up several questions: What version(s) of SipXecs will support the new configs? Is the current development version being tested against the new configs? Is there a migration plan? I noticed there is also a configuration file migrat

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