[sipx-users] Is a user able to diable VM himself? - Feature request for this purpose.

Hello List members, I often get the Request of enabling/disabling VM for users. As far as I can see the only way to disable VM for a user is to remove the permission as administrator. Or did I miss another way to achieve this setting as a user? It would be a great feature to control VM within use

Re: [sipx-users] One way audio

On Fri, Dec 18, 2009 at 11:52 AM, Burden, Mike wrote: > Twice in the past few weeks we have had calls where the audio suddenly > went “one way.” > > > > The first time it happened, it was about 30 minutes into the call. It > happened again today, about 5 minutes into the call. > > > > The sipX

[sipx-users] Sipx trunk ID for mediant

How do I specify the mediant 2K Trunk ID in sipx? ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/listinfo/sipx-users sipXecs IP PBX --

Re: [sipx-users] Nortel Enterprise Sale announcement

Great news!! I hope Avaya keeps the torch lit on this project and, better still, puts more resources into it. As far as I know Avaya has nothing quite like sipX so this project is a great addition to their portfolio. I do have one suggestion for anyone who can talk to the team that determines

[sipx-users] Caller ID based forwarding

I'd like to forward a phone based on the calling party ID. Can this be done? Basically it's a follow-me forward feature but based on the who is calling. For example, if a very important customer is calling the support desk, I'd like to forward the call to the service manager before going to v

[sipx-users] Hunt Goups

All, As part of my call flow, I have all calls come in to ext. 120, which is a phantom that immediately forwards the call to ext 200. After 20 seconds of no answer, I have the call dumped into a hunt group, ext 501. For the most part, it works great. My hunt group is set to initially call ext

Re: [sipx-users] Faxing in sipXecs

We have had great success using asterisk with nvdetect and hylafax even over sip trunks. The beauty of this type of setup is that it is very flexible and eliminates costly additional licenses. It has worked well for us over sip trunks but best over PRI trunks if you need high reliability. On t

Re: [sipx-users] Rejecting calls on Polycom 650

Sorry for incomplete information. Yes, the call comes in to the SPA3102 'fxo1' at 192.172.252.21 and it is programmed to dial 0...@... to get the auto attendant. I enter '200#' to transfer to extension 200 "Kitchen", this is 192.172.252.42. The sipX code is running on 192.172.252.163. Here's a

[sipx-users] Avaya IP 406

I see a lot of Avaya folks on this list so thought I'd ask here. Can the Avaya IP 406 be used with sipx in any way? I have one, along with some expansion chassis sitting here doing nothing. Would be nicer to see it doing, something :). ___ sipx-users

Re: [sipx-users] Incoming Calls

Sipxbridge has many fixes in 4.0.4. Please update. 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 Cu

[sipx-users] Incoming Calls

Hey guys Ive setup sipx at another location and have setup an Unlimitel SIP Trunk. Whats weird is when I go to call the DID the phone will ring once and then hang up then I'll get a busy on the other line. The DID is set under a user alias as I was having the same problem using sipxbridge...any ide

Re: [sipx-users] [sipX-dev] Nortel Enterprise Sale announcement

On Fri, 2009-12-18 at 15:31 -0500, Scott Lawrence wrote: > I'm told that this transaction is completing on schedule today, and want > to update the sipXecs community with what we know. Here is the Nortel press release announcing that the transaction closed: http://www2.nortel.com/go/news_detail.j

Re: [sipx-users] Nortel Enterprise Sale announcement

On Tue, 2009-07-21, Scott Lawrence wrote: > > When there's some definitive news, you'll hear it pretty quickly here. On Mon, 2009-09-14 at 13:09 -0400, Scott Lawrence wrote: > Nortel Selects Avaya as Successful Bidder for Enterprise > Solutions Business I'm told that this trans

Re: [sipx-users] Rejecting calls on Polycom 650

On Fri, 2009-12-18 at 14:40 -0500, Dale Worley wrote: > On Fri, 2009-12-18 at 10:07 -0600, Eric Varsanyi wrote: > > Attached to this email is an xml merged trace, if that's the wrong > > thing to send please let me know. > > > > It looks like things go wrong when it tries to send it back to the >

Re: [sipx-users] Rejecting calls on Polycom 650

On Fri, 2009-12-18 at 10:07 -0600, Eric Varsanyi wrote: > Attached to this email is an xml merged trace, if that's the wrong > thing to send please let me know. > > It looks like things go wrong when it tries to send it back to the > proxy ("481 Call Does Not Exist"). There are three things I not

Re: [sipx-users] Rejecting calls on Polycom 650

On 12/18/09 11:44 AM, Scott Lawrence wrote: > > We've discussed the problem of simultaneously supporting multiple > > versions many times... I just looked and couldn't find an issue to track > > it, so I just created it: > > > > http://track.sipfoundry.org/browse/XX-7308 On Fri, 2009-12-18 at 12

Re: [sipx-users] Rejecting calls on Polycom 650

On 12/18/09 11:44 AM, Scott Lawrence wrote: > On Fri, 2009-12-18 at 09:46 -0600, Eric Varsanyi wrote: >> >> If I try to downgrade the 650's to 3.1.3RevC to test the 'reject' key >> there is there some way to downgrade the sipxecs templates for just >> some phones but not others? Polycom warns of di

Re: [sipx-users] DTMF between SPA-3102 and Polycom 650s/335s

Eric, Look to the Patton Smartnode's... 4114/jo (4 fxo ports) and a 4112/jo (2 fxo ports) if you need 6 lines. From: Eric Varsanyi [mailto:sip...@eljv.com] Sent: Friday, December 18, 2009 8:47 AM To: sipx-users@list.sipfoundry.org Cc: Picher, Michael Subject: Re: [sipx-users] DTMF between

Re: [sipx-users] One way audio

I bet the first question someone will ask is "which way"? The inbound audio kept working correctly (we could hear the other party), but the outbound audio failed (the other party could not hear us.) Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone

[sipx-users] Auto dialer add on?

I have a requirement for an auto dialing system for some marketing work, sending updated information to some 1000 customers or so. Is there anything which works with sipx which might help us send the info out using say a full PRI along with a little intel to let us know what happened, msg's left

[sipx-users] One way audio

Twice in the past few weeks we have had calls where the audio suddenly went "one way." The first time it happened, it was about 30 minutes into the call. It happened again today, about 5 minutes into the call. The sipXtrace looks normal: Time: 2009-12-18T15:51:44.184000Z sipXbridge ->

Re: [sipx-users] Rejecting calls on Polycom 650

On Fri, 2009-12-18 at 09:46 -0600, Eric Varsanyi wrote: > > If I try to downgrade the 650's to 3.1.3RevC to test the 'reject' key > there is there some way to downgrade the sipxecs templates for just > some phones but not others? Polycom warns of dire trouble if you try > to use the wrong base con

Re: [sipx-users] Rejecting calls on Polycom 650

The firmware itself isn't the issue, its the configuration templates that sipxecs uses to make the tree of xml config files for the phones. They're supposed to 'match' the firmware. I guess I can look back in subversion for the commit that took the templates from 3.1.3Rev to 3.2.x and check out

Re: [sipx-users] Rejecting calls on Polycom 650

You would activate the 3.1 firware, reboot one 650, load the firmware, then deactivate the 3.1 firmware again. If you have NO firmware activated they will all stay at whatever versions they are running. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 E

Re: [sipx-users] Rejecting calls on Polycom 650

3.1.3RevC doesn't support the SoundPoint IP 335's that I have (they are post 3.2.1RevB only). If I try to downgrade the 650's to 3.1.3RevC to test the 'reject' key there is there some way to downgrade the sipxecs templates for just some phones but not others? Polycom warns of dire trouble if yo

Re: [sipx-users] Rejecting calls on Polycom 650

On Fri, 2009-12-18 at 09:29 -0600, Eric Varsanyi wrote: > With 4.2-17527 and the 3.2.2.0477 release of sip.ld a call from an FXO > gateway (SPA3102) to an internal 650 extension rings fine and if I let > it ring it goes to voicemail. If I hit the 'reject' softkey it > terminates the session and han

Re: [sipx-users] Rejecting calls on Polycom 650

The 3.2 firmware release from Polycom is known to have problems. If the same problem manifests itself with 3.1.3RevC it would be of more interest. On Fri, Dec 18, 2009 at 10:29 AM, Eric Varsanyi wrote: > With 4.2-17527 and the 3.2.2.0477 release of sip.ld a call from an FXO > gateway (SPA3102)

[sipx-users] Rejecting calls on Polycom 650

With 4.2-17527 and the 3.2.2.0477 release of sip.ld a call from an FXO gateway (SPA3102) to an internal 650 extension rings fine and if I let it ring it goes to voicemail. If I hit the 'reject' softkey it terminates the session and hangs up on the caller rather than going to voicemail. Is there

[sipx-users] Avaya and sipXecs

Should this project be safe with Avaya? Or should be go ahead and reconsider going to FREESwitch or even 3CX? ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: ht

Re: [sipx-users] Faxing in sipXecs

SipXBridge already works well with T.38 faxing. We have had some success with the Zoiper soft fax client. It's basically a softphone that supports T.38 faxing. You should be able to use any SIP softphone with T.38 support. Your siptrunk provider also needs to support T.38. We have been us

Re: [sipx-users] FW: No CDR Data

On Fri, 2009-12-18 at 01:31 +, Abdul Mayat wrote: > My server is runnning v4.0.3 and was built from ISO. > > I noticed that there is no call data in the CDR screens since a couple > of days. > > I have restarted the database and the server, but still no change. > > There have not been an

Re: [sipx-users] Faxing in sipXecs

The 200 extensions project is just fine. It's setting up the Asterisk+Hylafax+IAXmodem+avantfax that is giving me trouble. Sorry for the confusion, I should have made it clearer. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 12/18/2009 6:04 AM, Robert B w

Re: [sipx-users] DTMF between SPA-3102 and Polycom 650s/335s

I did an installation using a Dialogic 8 port FXO/FXS DMG1008. Quite a lot to configure but it worked i would say flawlessly with sipxecs 3.8. If i remember correctly and i think i do, grouping ports was easy. *Vänliga Hälsningar/Best Regards* /Ola Samuelson / 2009-12-18 14:46, Eric Varsan

Re: [sipx-users] DTMF between SPA-3102 and Polycom 650s/335s

Yeah, I wish I could have stuck with the Audiocodes 4FXO/4FXS I originally bought. I discovered after I tried to get it working that this is the only member of the mediapack family that sipXecs doesn't support. I was willing to figure out its rather complex configuration manually but I ran acro

Re: [sipx-users] Faxing in sipXecs

Something I discovered the other week: I have known that MS Exchange 2007 allowed the receipt of faxes using SIP/T.38. I recently upgraded our mail system from Exchange 2003 to Exchange 2010. After that, I thought I'd take a look at the Exchange fax receiving capability. It turns out that thi

Re: [sipx-users] FW: No CDR Data

I would want to know what UA's you have that could be showing up as "unknown". Also, you should upgrade it to 4.0.4, which is the current stable version first. from the CLI: yum update I think it would be a bad assumption to raise a bug until more is known. If your UA is sending badly formed inf

Re: [sipx-users] FW: No CDR Data

Tony, That was the obvious thing to try, which I did a few times, the service always shows running. I have also restarted the server completely. Do you know of any method to competely re-initialise the service? Also, if the service is being terminated due to a SIP UI using 'unknown' in t

Re: [sipx-users] Analog port Question

If you connect a handset (analog) to the line and dial the same number, how long does it take to start ringing? On Fri, Dec 18, 2009 at 2:43 AM, Picher, Michael wrote: > It really shouldn’t take that long to hear the other side start ringing. > > > > Is your gateway setup for G.711u only? Also,

Re: [sipx-users] FW: No CDR Data

In sipXconfig: System>Servers>(choose yours)>CDR>Restart On Thu, Dec 17, 2009 at 8:31 PM, Abdul Mayat wrote: > Sorry forgot to add, the only error I have come across is: > > "2009-12-17T15:08:54.055156 ":INFO:TERM intercepted. Terminating reader > threads. > "2009-12-17T15:10:24.794261 ":ERR:Ta