Re: [sipx-users] Help with Patton gateway

You know I'll be one of the first ones to test it. Just let me know when it's available. On 07/18/2010 07:20 PM, Douglas Hubler wrote: > On Sun, Jul 18, 2010 at 5:10 PM, Michal Bielicki > wrote: > >> The fix is for a far newer freeswitch version than the one in sipX, so I >> guess in the cu

Re: [sipx-users] Help with Patton gateway

There used to be an issue in the tracker XX-7892: When performing consultative transfer of a call to VM and AA, IVR session is not always restarted after transfer completion When performing a consultative transfer of a caller to voicemail, onc

Re: [sipx-users] Help with Patton gateway

On Sun, Jul 18, 2010 at 5:10 PM, Michal Bielicki wrote: > The fix is for a far newer freeswitch version than the one in sipX, so I > guess in the current sipXecs versions it is still in. Idea: I have to update FS in the unstable build for an OpenACD fix anyway, once I do someone can test this on

Re: [sipx-users] Help with Patton gateway

Stop trying to calm Todd down. He gets so cute and purplish when his veins stick out. You just ruined by beer:30! Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk:

Re: [sipx-users] Help with Patton gateway

The fix is for a far newer freeswitch version than the one in sipX, so I guess in the current sipXecs versions it is still in. Am 18.07.2010 um 23:03 schrieb Tony Graziano: > Merry Christams (in July). This was a known issue back in August 2008 and was > fixed. Something must have changed. I wi

Re: [sipx-users] Help with Patton gateway

Your analogy is flawed anyway. The conference DID workarounds are invisible to the end user in that they don't have to worry or think about how they dial the conference bridge. The attended transfer "workaround" (if you want to call it that) requires end users to make a conscious decision about

Re: [sipx-users] Help with Patton gateway

Merry Christams (in July). This was a known issue back in August 2008 and was fixed. Something must have changed. I will post this relevant FS thread in the JIRA you opened. http://jira.freeswitch.org/browse/SFSIP-86 On Fri, Jul 16, 2010 at 8:49 PM, Josh Patten wrote: > The reason why this is an

Re: [sipx-users] Help with Patton gateway

Mr. Hodgen, I take GREAT issue with only one thing: "Set expectations. You can’t do a supervised transfer to Auto Attendant or Voicemail in the system today. It’s true, it’s accurate, it’s an expectation you need to set for the end users. You can dance around this all you want, but it doesn’t

Re: [sipx-users] Help with Patton gateway

Tony, I realize that you think your opinion on these things is the only important one, however, the analogy that was used is accurate. The feature is not there for direct DID to a conference bridge. Yes, there are workarounds, and when I discuss this with an end user, I set the expectation that I

Re: [sipx-users] "Internal Error" trying to activate in Devicefiles 3.1.3revC for Polycom phones

On a lighter note, if you are restoring a system the bootrom and frimware that you probably needed was already on the phones, so having this fail trying to manipulate them is a generic cleanup issue (it should be fixed) and "I would not" consider it something that stopped me from getting a system r

Re: [sipx-users] "Internal Error" trying to activate in Devicefiles 3.1.3revC for Polycom phones

Yep. Happened to me yesterday. -Original Message- From: Eric Varsanyi Sender: sipx-users-boun...@list.sipfoundry.org Date: Sun, 18 Jul 2010 15:04:16 To: Tony Graziano Cc: Subject: Re: [sipx-users] "Internal Error" trying to activate in Device files 3.1.3revC for Polycom phones

Re: [sipx-users] Help with Patton gateway

Sure, but where is it written or set by the project one can no longer do an attended transfer to a media based service? It's not, so the anology should be thrown out. There, I just did it. Besides the fact there are at least two workarounds for getting a DID number to the conferencing system also

Re: [sipx-users] Help with Patton gateway

Analogy Tony. A simple Analogy. From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Sunday, July 18, 2010 12:56 PM To: Todd Hodgen Cc: Josh Patten; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Help with Patton gateway I disagree. Doing an attended transfer to AA wo

Re: [sipx-users] "Internal Error" trying to activate in Device files 3.1.3revC for Polycom phones

I've seen this when restoring from a backup on a freshly installed system. The backup proess doesn't (as of 4.2 anyway) save all the managed files, but they remain referenced in the database so you get this weird error trying to work with them. If you delete them then re upload it will clear its

Re: [sipx-users] Help with Patton gateway

I disagree. Doing an attended transfer to AA worked in 3.10.3, there is nothing in the notes that this is no longer supported going forward. The project also needs to set these expectations. It was never stated it was no longer supported. So in this case, setting an expectation would still result i

Re: [sipx-users] Help with Patton gateway

Set expectations. You can't do a supervised transfer to Auto Attendant or Voicemail in the system today. It's true, it's accurate, it's an expectation you need to set for the end users. You can dance around this all you want, but it doesn't change the fact that it isn't supported. Yes, it sho

Re: [sipx-users] Help with Patton gateway

The thing that we should be pushing for is an answer from the FS team as to whether this is fixed in 1.07. I'm somewhat eager to get 1.07 since is can handle t.38 faxing too, which is a big deal too. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Ema

Re: [sipx-users] Help with Patton gateway

I only run 550's and 650's. I really don't think speculating about ways for end users to "work around" the problem is very constructive. As Tony and I have both said this is a major flaw that should be fixed and I have a feeling it's going to take a while due to having to wait for FreeSWITCH u

Re: [sipx-users] let's strip (a digit that is...)

You need to state two things in that regard: routing-table called-e164 {context} route .% dest-interface IF_NAME_HERE STRIP_MAP route default dest-interface IF_NAME_HERE #strip the first digit from outgoing calls mapping-table called-e164 to

Re: [sipx-users] Help with Patton gateway

On Sat, Jul 17, 2010 at 7:09 AM, Michael Scheidell wrote: > > I suppose we could just train everyone to push that extra 'more' button and > hit blind transfer. > In fact, an attended transfer to AA or VM doesn't make all that much sense > anyway. > > > For your Polycom 320/335's you can make blind

Re: [sipx-users] Restore questions - mainly voicemail

Yes unless you remove the MOH uri from the phone or group setting. 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 Help

Re: [sipx-users] Restore questions - mainly voicemail

Thanks. What a night. I accidentally upgraded to CentOS 5.5. Then, I ended up having all sorts of issues trying to get the latest HP management software installed. It caused the server to panic, fried the NIC drivers, etc. All this because I had no idea that a caller hears music instead of ring

Re: [sipx-users] Restore questions - mainly voicemail

I know its late to say this, but restore from CLI is an option 9for future reference). On Sun, Jul 18, 2010 at 1:06 AM, Matthew Kitchin (public/usenet) < mkitchin.pub...@gmail.com> wrote: > False alarm maybe. It looks like I may have picked 5 users that only > listen to them as email attachments.