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mike@grounded.net | 10 Feb 23:19
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Off topic: Anyone using appia?

Is anyone on the list using appia? 

They don't yet have a GUI but talk about it being done soon. 
Just changing ports takes having to communicate with support.

So far, every time we come to test something, we're told they need to program that functionality in. We were
trying to test a sip to analog gateway to use with pbx's but nothing worked. They found some problems which
took a week to resolve saying that they didn't cover that functionality. 

Now with sipx, they can't send to me on port 5080 yet I thought someone here mentioned using them and they work fine.

The support person is wonderful to work with but it's going on three weeks I believe now since we contacted
them and have been trying to test.

Just wondering.
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Becker, Jesse | 10 Feb 22:49
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SmartNode 4960 Series

All,

 Given the price-point, I may go with the Patton gateways instead Audiocodes. From previous posts it looks like several have used them with great success. I have experience wtih both Cisco and Extreme devices, so I am not daunted by having to do CLI. I have downloaded the guide and will review over the weekend.

Does someone have an example config file they would be willing to share? Are there any gotchas that I should look out for when using with SipX? 

We will be using one at our main site with 3 PRIs and one at our remote site with 1 PRI.

Thanks, and have a great weekend.

Jes

--


Jesse Becker  |  Technical Manager
Network+ | Linux+ Certified Professional
DATATEL+SGHE <at> SUNY Ulster
491 Cottekill Road, Stone Ridge, NY  12484
Tel 845-687-5064 | Fax 845-687-5105
beckerj <at> sunyulster.edu | www.sunyulster.edu

Check out our knowledge base: http://kb.sunyulster.edu


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Ken Ridley | 10 Feb 22:23
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All phone registrations expiring

I Checked all of the messages I have received from this group, and I found a  thread about this issue, and it says to check the logs

 

Is this the only option?

If it is, what log do I check, and what am I looking for?

 

This has happened twice on this system, rebooting it fixes the problem

 

Thank you,

 

Ken Ridley

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Domenico Chierico | 10 Feb 10:59
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cdr and call forward

someone knows how forwarded call is expected to be into the cdr?

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Tommy Laino | 9 Feb 23:19
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Polycom Remote Phone MWI

I have 3 remote phones running off my lab system. They are all Polycom 550. None of them will show a messaging waiting indicator. I checked the SRV records of the public domain name and they are all pointing at my SipX public IP.

Am I missing a setting for the user somewhere? Reading the user posts in this forum it seems that most of the time its a DNS issue but here I cant seem to find that. All the users are Private and not shared.

Any help is appreciated.

Tommy Laino

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Berthold Rühl | 9 Feb 20:13
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Webinterface not working

Hi,
i just installed sipxecs on a vserver running centos 5. After the Installation i tried to access the
webinterface and got redirected to https port 8443 but the webinterface didn't open because my browser
was unable to connect to that port.
Can you tell me which directory has to be accessed by the webserver?

Thank you
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Tommy Laino | 9 Feb 16:27
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Server Recommendations

I am going to be deploying my first SipX after much work and testing on my lab system. Just curious what most of the experts are using for servers on their deployments.

I am going to have a very simple setup to start. 20 local sets and 2 remote users using NAT and SIP trunking from the cable company.

Tommy Laino

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Mark Roseboom | 9 Feb 16:01
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Re: On Hold Fails for Inbound Calls through SIP Trunk

Unfortunately it appears its not a grandstream issue as transfers handled by the autoattendant are dropped as well.


--
Mark Roseboom
38 Media, Inc. | Head of Operations 

w: 617.674.0828
m: 617.286.2121



On Wed, Feb 8, 2012 at 10:21 PM, Tony Graziano <tgraziano <at> myitdepartment.net> wrote:

You should also indicate if the user is a local or remote.

On Feb 8, 2012 10:12 PM, "Tony Graziano" <tgraziano <at> myitdepartment.net> wrote:

I think this is more a question like "does moh work using gs handsets"?

On Feb 8, 2012 9:28 PM, "Michael Picher" <mpicher <at> ezuce.com> wrote:
and the crowd shudders....

On Wed, Feb 8, 2012 at 11:42 AM, Mark Roseboom <mark.roseboom <at> 38media.net> wrote:
Grandstream GXP2000.

--
Mark Roseboom
38 Media, Inc. | Head of Operations 




On Wed, Feb 8, 2012 at 11:41 AM, Tony Graziano <tgraziano <at> myitdepartment.net> wrote:

What phones are you using?

On Feb 8, 2012 11:37 AM, "Mark Roseboom" <mark.roseboom <at> 38media.net> wrote:
Invites are now being sent to port 5080 however inbound calls that are placed on hold are dropped. Does anyone have any other suggestions?

Thanks.

Mark 
--
Mark Roseboom
38 Media, Inc. | Head of Operations 




On Tue, Feb 7, 2012 at 2:31 PM, Tony Graziano <tgraziano <at> myitdepartment.net> wrote:

Ask them to send invites for inbound calls to port 5080 and make sure the firewall does the same (5080:5080).

On Feb 7, 2012 2:06 PM, "Mark Roseboom" <mark.roseboom <at> 38media.net> wrote:
Currently, the provider is sending calls to 5060, however in our network configuration, the sipxec server is behind a firewall, so I am forwarding port 5060 requests to port 5080 on our sipxec. The provider is IIS Group. You are correct on hold or transfers don't work for inbound calls. Is this port forwarding configuration enough or do I need to have the provider send requests to port 5080?

On Tue, Feb 7, 2012 at 1:59 PM, Tony Graziano <tgraziano <at> myitdepartment.net> wrote:

Is the provider sending calls to port 5080 or 5060? Who is the provider. Calls need to be sent to port 5080. Does this mean transfers don't work? If so check or ask the provider to send invites to port 5080.

On Feb 7, 2012 1:54 PM, "Mark Roseboom" <mark.roseboom <at> 38media.net> wrote:
Using sipXecs 4.4. Inbound and outbound calls work with my ITSP. 

Placing a call on hold works when calling any internal extension. On hold also works when making an outbound call through SIP trunk.

However when an inbound call comes in, and its placed on hold, the call is dropped. Thinking it must be a NAT / firewall issue. 

Not sure what other diagnostics to give, please let me know.

Thanks.

Mark

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Suite 201

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Mark Wood | 9 Feb 00:20

slow transfers

Over the last weekend one of our installations had an outside IT firm add new DHCP & DNS servers. They also changed their internal domain from .com to .local. We added an alias for the .local in System->Domain and the system came back up and calls were processed. Today however, calls transferred to stations are sloooooow to connect and calls from the AudioCodes to IVR’s and stations are experiencing the same problem.

 

Thanks,

Mark W. Wood

office: (760)202-0224   X2010

Direct: (760)459-1981

www.redphonetech.com

 

 

 

 

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Becker, Jesse | 8 Feb 23:56
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PRI Standard for Audiocodes

All,
  What is the best PRI standard to use with AudioCodes M1000s to allow calling number and name?  We currently use ESS5 with our Cisco gear.  Will this standard also work with the M1000s and SIPX or should we be looking into another standard such as NI2?

Thanks,

Jes

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Hernan Robledo | 8 Feb 20:47
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Postgre 9.1-4

Dear,


could you tell me if sipxecs support Postgre 9.1-4 version?

I tried to install this DBMS and my sipexcs doesn't work anymore.

Thanks for your help


Hernan

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Gmane