Re: [sipx-users] Audiocodes MP-118 FX0 auto dialing to non-operator

2009-01-05 Thread Tony Graziano
Sorry, just read Ryan's full post. This should work, assuming you have manually made the changes and canconfirm they are in there, you should run a log trace from the ACgateway and see why it is complaining. If it looks correct, then a calltrace would be in order. I don't think logging into the

Re: [sipx-users] Audiocodes MP-118 FX0 auto dialing to non-operator

2009-01-05 Thread Tony Graziano
I think the question he is asking is different than what you are offering as an answer. I think what the Quigley needs to know is that a PSTN gateway cannot dial users directly (i.e. Can dial 100, aa's, hunt groups). As. Gateway, it doesn't have that kind of authority. Quigley could also sta

Re: [sipx-users] Audiocodes MP-118 FX0 auto dialing to non-operator

2009-01-05 Thread Gerald Drouillard
On 1/5/2009 1:14 PM, Ryan Quigley wrote: > Hi, > How does one go about routing an incoming PSTN line through an > Audiocodes MP-118 to anything other than the operator Auto Attendant? > I'm trying to route directly to a Hunt Group and bypass the operator, > but setting the Destination Phone Number

Re: [sipx-users] SipX on VMWare ESXi

2009-01-05 Thread Matt White
I should add we have two installation of Sipx on ESXi with "good" success. Here is what we have found. Here are some rules we have found that work most of the time. 1. Use the latest version of ESX...ie 3.5.x 2. Never try to run anything else on the server, I know, seems to be counter to virtua

Re: [sipx-users] SipX on VMWare ESXi

2009-01-05 Thread Tony Graziano
Thanks for following up with him Matt. I think even if you dedicate resources, there's still always a possible issue. It's fine for lab use, but not really for production use I think. Unless you have a really good way to always guarantee all resources without latency in a virtual environment,

Re: [sipx-users] SipX on VMWare ESXi

2009-01-05 Thread laleger
I'm pretty sure if you search: Vmware site:sipfoundry.org On google you will find some info on running sipx on a virtual machine. The quality issues you have reported pertain to virtual machine timing so you can throw all the processor and memory resources possible but it will not make a diffe

Re: [sipx-users] Aastra and presence

2009-01-05 Thread Matt White
Yes, I've tried both URL's with the c and without. The admin manual states the BLF/List feature is complaint with the Broadworks format R11 and up. So that is why I have been focusing on geting the Broadsoft format working. The Aastra phone doesn't seem to provide detailed enough logs for me t

Re: [sipx-users] SipX on VMWare ESXi

2009-01-05 Thread Matt White
How many physical cores are in the ESXi server? It's a common mistake for people to assign 2 vSMP to a host. Generally speaking with ESX, you will get lower performance when using multiple cpu's inside a VM. Thats because there is a performance hit to virtualizing two cpu's. The only time th

[sipx-users] SipX on VMWare ESXi

2009-01-05 Thread Jhony Perez
Hello everyone, I have SipX 3.10.2 with CentOS 5 running on a VMWare ESXi server, I've assign 1GB of RAM and selected 2 CPU during the install, the system seems to run just fine but we started having intermittent issues with voicemail quality, at first it was only on the playback of the messa

Re: [sipx-users] Aastra and presence

2009-01-05 Thread Dale Worley
On Mon, 2009-01-05 at 08:53 -0500, Matt White wrote: > The other option is the BLF/List. This supports the Broadsoft list > type. This is the one I really thought should work. A packet trace > shows the Aastra phone request the sip:~~rl...@example.com and sipx > responds with rls urls. A bunch

Re: [sipx-users] inband DTMF support in sipXecs?

2009-01-05 Thread Dale Worley
On Mon, 2009-01-05 at 14:16 -0600, icy hot wrote: > I'm testing inband DTMF with a Linksys PAP2 ATA against different SIP > servers and noticed it doesn't work with sipXecs. > > > Is there a way to enable this? sipXecs requires that DTMF be sent in the RFC 2833/4733 method. Most devices allow

Re: [sipx-users] inband DTMF support in sipXecs?

2009-01-05 Thread icy hot
Hi Tony, The DTMF Tx Method is set to inband on the ATA; OOB works fine. Where is the signaling media type you mentioned configured? Is this a setting on sipXecs or the ATA? The Linksys PAP2 is the only ATA to which I have access. Thanks for your help! On Mon, Jan 5, 2009 at 2:25 PM, Tony Grazi

Re: [sipx-users] inband DTMF support in sipXecs?

2009-01-05 Thread Tony Graziano
Your signaling media type needs to be "101". Try checking your signalling methods and change t from AUTO to maybe INBAND, and perhaps check your Linksys manual. sipXecs does not have a plugin to autoconfigure this, so saying that it doesn't work is perhaps a statement that is a little overbroad

[sipx-users] inband DTMF support in sipXecs?

2009-01-05 Thread icy hot
Hello, I'm testing inband DTMF with a Linksys PAP2 ATA against different SIP servers and noticed it doesn't work with sipXecs. Is there a way to enable this? Thanks! Joe ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://lis

[sipx-users] Audiocodes MP-118 FX0 auto dialing to non-operator

2009-01-05 Thread Ryan Quigley
Hi, How does one go about routing an incoming PSTN line through an Audiocodes MP-118 to anything other than the operator Auto Attendant? I'm trying to route directly to a Hunt Group and bypass the operator, but setting the Destination Phone Number to anything other than "operator" or "100"

[sipx-users] Aastra and presence

2009-01-05 Thread Matt White
I know I have asked this question before, so I don't want to sound like a pest but I really need to figure this out if I can and could use the help. I have a ticket open with Aastra but other than take the ticket info, they have not gotten back to me. So, if you have Aastra phone with workin