Re: [sipx-users] Cisco 3945 with Sipxecs

oundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen Sent: Thursday, January 13, 2011 12:24 PM To: jmicc...@redhat.com; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Cisco 3945 with Sipxecs Nortel's implementation of IP is proprietary. In order t

Re: [sipx-users] Cisco 3945 with Sipxecs

Nortel's implementation of IP is proprietary. In order to come to a SIP compliant application on the switch, it requires a very expensive upgrade to the current switch, which is viewed as a sunken investment by most. This is working now. I will be posting the configuration to the wiki once w

Re: [sipx-users] Cisco 3945 with Sipxecs

This was solved. Which means the Cisco 3945 is processing calls between Nortel Meridian network and sipXecs. A new solution for enterprise applications. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen Sent: Wednesday

[sipx-users] Cisco 3945 with Sipxecs

I'm working on a project that looks like this - Nortel PBX --- > PRI ---> Cisco 3945 ---> sipXecs The goal is to have the Cisco do Gateway services between the Nortel PBX and the sipXecs system. Today, I have calls going out of the sipXecs via this setup fine. Two way audio, caller I

Re: [sipx-users] Dimdim bought by Salesforce.com

From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Martin Steinmann Sent: Thursday, January 06, 2011 4:27 PM To: 'Discussion list for users of sipXecs software' Cc: Fred Dixon Subject: Re: [sipx-users] Dimdim bought by Salesforce.com >

Re: [sipx-users] Caller-id settings not working

If you do a simple search of the attached file looking for "6102" you find it multiple times. And not one time does it have a 1 in front of it. At least not from what I could find. It's pretty basic, if sipXecs was putting that one is for any reason, you would find it in your capture from the

Re: [sipx-users] Caller-id settings not working

I don't believe it's a bug. If you go into User / Caller ID there is a section for putting in a unique caller ID on a per user basis. These users are assigned as lines on a phone. "Externally visible user Caller ID. If it is not specified the gateway will create a Caller ID based on the caller's

Re: [sipx-users] Caller-id settings not working

Have you tried doing packet captures to see what is being sent out the PBX, what is being sent out the Gateway, etc. to see just where the number being received by the called party is coming from. Seems without knowing what device is actually developing the incorrect CallerID, it's difficult to k

Re: [sipx-users] New Subscriber Intro

...@list.sipfoundry.org] On Behalf Of bob gailer Sent: Saturday, January 01, 2011 5:15 PM To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] New Subscriber Intro On 1/1/2011 3:16 PM, Todd Hodgen wrote: > sipXecs is an enterprise telephone system. As such, it will generally be > installed on a companies

Re: [sipx-users] New Subscriber Intro

sipXecs is an enterprise telephone system. As such, it will generally be installed on a companies domain. You would give it a host name on that domain. SipXecs.bobgailer.com as an example. On the front page of sipfoundry.org there is a large graphic with a link to a book about the sipXecs syste

Re: [sipx-users] remove voicemail setting from extensions that don't have voicemail

Check User/Permissionuntick the Voicemail option.You can do it through group settings as well if they are in a group. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Andrew Radke Sent: Tuesday, November 30, 2010 9:10 PM To: sipx-

Re: [sipx-users] Disable user to user calls

From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen Sent: Monday, November 22, 2010 8:23 PM To: 'Discussion list for users of sipXecs software' Subject: Re: [sipx-users] Disable user to user calls Go to System/Dialp

Re: [sipx-users] Disable user to user calls

Go to System/Dialplans. Add new "Custom" plan from the dropdown On the template, give it a name, click on enable, then set the dial rule. IF you have 3 digit extensions as an example, that start with 7, then build a rule under Dialed Number of Prefix "7", and "2 digits" Drop to Resultin

Re: [sipx-users] Wiki cleanup day results

What might be helpful for new users is to have the WIKI added to the menu on the current Sipfoundry.org website. It's kind of hidden under "Developers" now. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Martin Steinmann Sent: Sunday,

Re: [sipx-users] Wiki cleanup day results

Martin, The Sipfoundry.org site is based on a CMS system if I'm not mistaken. I know the old one was Joomla, I'm not sure what the new one is. Maybe some of the content on the wiki is better served being on the web site, within specific pages that are dedicated to things like documentati

Re: [sipx-users] Odd Registration

There was a Pingtel hard phone - xpressa. I've seen them being sold on ebay in the past. I would suspect it had it's own Mac registration when it was being produced. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf

Re: [sipx-users] Odd Registration

That mac address belongs to a Pingtel phone. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net Sent: Sunday, November 21, 2010 7:31 AM To: sipx-users Subject: [sipx-users] Odd Registration I've ne

Re: [sipx-users] Inbound Verizon Callers

I can point to at least one installation that has a hunt group with 9 Polycom's on it that are all in a hunt group with their line 3. Each phone also has their own two extension so it. It works beautifully. The devil is in the details - IF you are provisioning the same line on multiple phones,

Re: [sipx-users] Emergency DNS question - HELP

Nathaniel, First off, get a backup. I may not be the best for this round of advise, but this is what I would try at this late hour. I'd run sipXecs-setup and change the DNS servers. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundr

Re: [sipx-users] Inbound Verizon Callers

Not likely. IMHO, if you look at your configuration between sipXecs and this carrier, it relies on one configuration between the ITSP and your SBC. That configuration doesn't change based on the number of phones that you have. If you think plugging in the last phone created the issue, it woul

Re: [sipx-users] Enable Call Forward

In 3.2.1, if you go to Devices, phones, then select one of the assigned lines, Registration, show advanced Features, you can add a tick mark to "Enable/Disable server based call forwarding" to get a forward button. Unless you are discussing a development release. Since this is the User list, I a

Re: [sipx-users] Problem upgrading 4.0.4 to 4.2.1 - ITSP calls give 'request timeout'.

IS this with dynamic or static IP address? In Diagnostic, Internal SBC statistics, does it show the trunks registered? From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jesse Reynolds Sent: Tuesday, November 16, 2010 4:10 PM To: Dis

Re: [sipx-users] Enable Call Forward

You can have a forward button on the Polycom phones with the current sipXecs provisioning. In fact, I remember having a devil of a time trying to figure out how I enabled it when I didn't want it any longer. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-user

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

+1 for the Kirk. I have a customer that uses the Kirk and is very happy with them. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jim Canfield Sent: Tuesday, November 16, 2010 2:22 PM To: Discussion list for us

Re: [sipx-users] autoattendant wav file

I agree with Mike. Record it into a voicemail box, which will give you excellent quality. Log into the portal for that extension, and copy it as a wav file. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher Sent: Satur

Re: [sipx-users] when to reload Freeswitch dial plans

+1 -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Douglas Hubler Sent: Wednesday, November 10, 2010 9:24 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] when to reload Freeswitch di

Re: [sipx-users] Look what happens when you have ancient PBX equipment

Sounds like an order for more Polycoms! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh M. Patten Sent: Thursday, November 04, 2010 5:55 PM To: Discussion list for users of sipXecs software Subject: [sipx-users] Look what happens wh

Re: [sipx-users] Soft phone

I believe that Voice Operator Panel is working on a provisioning piece with Sipxecs.You might check with them, and they have several versions of their product. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner Sent: Wednes

Re: [sipx-users] Voip.ms

I've been told that Broadvox will be rolling out their new user portal in early 2011, which will have the features you are looking for. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gary Luca Sent: Monday, November 01, 2010 12:01 PM To

Re: [sipx-users] Voip.ms

I am a reseller of Broadvox, and a user of Voip.ms. I don't recommend Voip.ms to customers except for labs, etc. as I find that they are not totally reliable. They are very cost effective, if you have a customer that does not sit at their phone all day long, it works great, but if they are on

Re: [sipx-users] Performance Degradation

Sure - friendly scanner - sipvicious. It is getting a response from you, or it would go away. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Roman Gelfand Sent: Friday, October 29, 2010 11:48 AM To: Discuss

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

Danny, a link to that firmware on the wiki would be helpful for others that experience that same issue. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Danny Shay Sent: Thursday, October 28, 2010 7:30 AM To: Discu

Re: [sipx-users] Incoming DID routing not working properly

With Broadvox, you can't use Dynamic addressing from them, and you need to be on their legacy platform. Once you do this, it works perfect. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Gary Luca Sent: Wednesday, October 27, 2010 6:12

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

I'd be careful. If you have door unlocks, you need one that will filter out incoming DTMF, or someone will use a pocket dialer to break into the place. Viking designs around that issue only allowing one way DTMF once speaker is enabled. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx

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

I've used Viking Doorphones, with the Viking Controller and connect it to an FXO port - works great. Careful though - the Viking Controller has an issue with Caller ID working with Audio Codes. It's a Viking issue. If you go with this, email me separately, I have a new model of the Viking to

Re: [sipx-users] HD MoH

Sounds like a great JIRA change request. Nice Job Josh! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh M. Patten Sent: Thursday, October 21, 2010 1:51 PM To: sipx-users@list.sipfoundry.org Subject: [sipx-users] HD MoH I was ti

Re: [sipx-users] Registration issue

Configure the domain name and not the IP address of your server. SipXecs will work much better when using domain name, as it has access to domain registration information that way - srv records, sips, etc. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry

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

r.com; Discussion list for users of sipXecs software Subject: 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

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

I've used the VVX to Counterpath Xlite for demonstrations successfully. It worked without any hitch. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins Sent: Wednesday, October 20, 2010 10:33 AM To: Discussion list for u

Re: [sipx-users] INSTALLATION +DNS

Install it with DNS, and use the server's DHCP, or your router. The DHCP is less fussy, but I would highly recommend you use the DNS in the sipXecs server, and it will work with your ISP's DNS. For a domain name, make one up -Jaramillo.local as an example. -Original Message- From: si

Re: [sipx-users] mediant 2K ring delay or sipx?

The ring has very little meaning, it's a wav file played by the phone, much like in the analog world - the number of rings you hear in your ear is not the same as you hear on the other end either. I would recommend looking at a trace on both ends to see what your SIP signaling is saying. I'm gues

Re: [sipx-users] Help with SUBSCRIBE for MWI

turns "Unauthorized" from a SUBSCRIBE. Shouldn't the reason appear in some log somewhere? If so, where? If I had a Polycom phone and it gave me a SUBSCRIBE error, where would you direct me to look to troubleshoot it? Jeff On Oct 18, 2010, at 1:33 PM, Todd Hodgen wrote:

Re: [sipx-users] Help with SUBSCRIBE for MWI

I have to agree. I ordered one once, it's around here somewhere, in my junk pile. Maybe they work for some, but geez, what a lot of work for little or nothing. They are low cost, but the cost in your time to get them to the level of others is immense. From: sipx-users-boun...@list.sipfoundr

Re: [sipx-users] Help with SUBSCRIBE for MWI

I think the issues you will find is that not a lot of people on this list use them for whatever reasons. No takers yet, eh? Bummer. Can anyone at least tell me which log would show the results of subscription requests. I see them in the sipxproxy log, but they never appear in sipxregistrar.

Re: [sipx-users] Mailing lists harvested for sip attacks

Additionally, if you start modifying the content of emails, you will be liable for any content that is there, there are plenty of legal precedents on that. If someone puts something libelous in an email, the owners of the site can be legally responsible. If you have a policy of never modifying c

Re: [sipx-users] Under Attack

In any event, my firewall stopped a huge amount and at really no bandwidth or resource loss during the attempt at the firewall, because I implemented my own formulas BEFORE I posted the blog entry... On Thu, Oct 14, 2010 at 3:04 AM, Todd Hodgen wrote: Welcome to Sipvicious! It really sux. -

Re: [sipx-users] Under Attack

Welcome to Sipvicious! It really sux. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dan McDaniel Sent: Wednesday, October 13, 2010 9:54 PM To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Under Atta

Re: [sipx-users] Digital Gateway opinions

I followed Mike, Tony and Jim's input and used a Patton SmartNode with one customer - it's been up for about 6 months - no issues yet once it was setup correctly. Seems to be very stable and reliable thus far. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoun

Re: [sipx-users] Multiple SIP Trunking

nt where do you setup call routing based on destination phone number. On Sun, Oct 10, 2010 at 4:13 PM, Roman Gelfand wrote: > Where do you configure routing based on caller id? > > Thanks for all the help > > On Sun, Oct 10, 2010 at 4:06 PM, Todd Hodgen wrote: >> Sorry, meant c

Re: [sipx-users] Multiple SIP Trunking

[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd Hodgen Sent: Sunday, October 10, 2010 1:02 PM To: 'Discussion list for users of sipXecs software' Subject: Re: [sipx-users] Multiple SIP Trunking With regards to sipXecs your impression is correct. If you have an ITSP tha

Re: [sipx-users] Multiple SIP Trunking

With regards to sipXecs your impression is correct. If you have an ITSP that supports the SIP standards, you will be able to multiple DID numbers (many many) that termintate at an auto attendant, or multiple auto attendants, your choice, or as DID to specific stations, or specific hunt groups, yo

Re: [sipx-users] Problem changing active greeting from phone

I think you will have to explain the issue in more detail. I for one have no idea what you are reporting here. Is this a problem where you can't change the voicemail active greeting via their telephone extensions? IS there an error tone, is there a hangup? Please explain the steps you are takin

Re: [sipx-users] 3:00 PM EST - Conference Call Capacity Test

Per voip.ms [Peter] If the did is in per minute, we can assure it will come with at least 10 incoming channels. If your traffic requires it we can increase, also some dids come with more thatn 10 by default [Peter] If your DID is in per minute plan do not have to worry, as we can increase channels

Re: [sipx-users] Is there a response code that keeps SipX from trying the next gateway in the list in dialplan?

-Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Worley, Dale R (Dale) Sent: Friday, October 08, 2010 11:20 AM To: Discussion list for users of sipXecs software Subject: Re: [sipx-users] Is there a response code tha

Re: [sipx-users] Welcome to George Niculae

Welcome to the wild side George! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Martin Steinmann Sent: Thursday, October 07, 2010 6:58 AM To: 'Discussion list for users of sipXecs software' Subject: [sipx-users] Welcome to George Nicula

Re: [sipx-users] SNOM 370 Problem Registering with sipx 4.2.1

When you specify the phone, it loads a preconfigured template that is designed for that phone. I'm not convinced the snom templates are updated much, but maybe someone else can chime in. SipXecs registers the end point, which is the snom phone. When you go into Diagnostics, Registrations, do yo

Re: [sipx-users] frequent restarts needed

Joe, rest assured, that definitely is not a normal attribute of this system. I've got customer systems that have never had to be reset - running consistently for 10-12 months without issues. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.s

Re: [sipx-users] Anyone in a position to test authorization code on 4.3 install?

If you are going to expose a security vulnerability like this, I'd recommend only allowing access to a trunk that has a finite ld value. For instance, a prepaid like VOIP.ms, so that when they have exceeded the $100 of long distance on the account in 15 minutes, calls don't go through any longer.

Re: [sipx-users] Anyone in a position to test authorization code on 4.3 install?

They seemed to work for the Road Runner. Maybe they are a different tone? -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Kyle Haefner Sent: Wednesday, September 29, 2010 11:23 AM To: Discussio

Re: [sipx-users] Random SipxProxy Problem

Fail2ban is what the developer of sipvicious states is used by most to stop sipvicious and the several variants of it that are active now. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Picher Sent: Monday, September 27, 2010 4:

Re: [sipx-users] Random SipxProxy Problem

I've seen this attack at customer sites that were blocked at the firewall. The only way I was able to detect them was when I ran TraceBuster between the firewall and the router - they jumped out pretty easily. Tracebuster is my new best friend. Can't go wrong for $99. In my particular case, s

Re: [sipx-users] Voice Operator Panel / SipXecs Optimized Version

He announced on a previous message that their next step is to make it configurable from sipXecs just for that purpose. also... How about a wiki page for configuring the phone? I see VOP uses an XML config file. It would be nice if sipXconfig can generate the config for so the user/admin

Re: [sipx-users] Anyway to call barred internal extension?

It fallows the dial plans in order on the screen. Make sure your dial plan to block is at the top. It follows the first one it finds a match with. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of heebun Low Sent: Friday, September 24, 2

[sipx-users] Conference with fxs ports

Has anyone gotten sipXecs to create conference calls on fxs ports on a gateway? Specifically on Audiocodes. I believe I have the Audiocodes set correctly, but there appears to be only one audio path it can setup.I'd be curious to know if anyone has ever had this feature work on a single li

[sipx-users] Viking Door Controllers

While we are talking about interoperability with other products, Viking Door Controllers are an issue, but they can be worked around. Caller ID out of the Viking has a very loose adherence to the Bell 202 standards for Caller is, they use 1200hz and 2400 hz rather than the 1200hz and 2300hz fo

Re: [sipx-users] Mail server

No, it does not have an integrated mail server. It uses sendmail to send to your smtp gateway. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of mattias Sent: Saturday, September 11, 2010 11:22 AM To: sipx-users@list.sipfoundry.org Subjec

Re: [sipx-users] [sipx-dev] ITSPs that work with sipXecs

M, Matthew Kitchin (Public) wrote: > No. Sorry. I was just saying when I had an ITSP say they wouldn't let me use sipx with their service, I told them it was a Nortel SCS and everything worked fine. > -Original Message- > From: "Todd Hodgen" > Sender: sipx-users-bo

Re: [sipx-users] [sipx-dev] ITSPs that work with sipXecs

t, Sep 11, 2010 at 8:12 AM, Matthew Kitchin (Public) wrote: > No. Sorry. I was just saying when I had an ITSP say they wouldn't let me use sipx with their service, I told them it was a Nortel SCS and everything worked fine. > -Original Message- > From: "

Re: [sipx-users] OT RTCP-XR monitor

nt: Monday, August 23, 2010 6:08 PM To: Martin Steinmann Cc: Todd Hodgen; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] OT RTCP-XR monitor Yes... *** McIlvin, Don to me show details 6:29 PM (0 minutes ago) Polycom Productivity suite is what we have for the station level. B

Re: [sipx-users] Call drops after 1 min & 29 secs

There have been some discussions about this ITSP on the list in the past. I did find this one. http://forum.sipfoundry.org/index.php?t=msg &goto=44468&S=9a2fe924342a700db212b8481e97cc22#msg_4

Re: [sipx-users] NTP server - or lack of

Dale - thanks for all the details, I appreciate it. -Original Message- From: Worley, Dale R (Dale) [mailto:dwor...@avaya.com] Sent: Thursday, September 02, 2010 11:10 AM To: Todd Hodgen; 'sipx-users' Subject: RE: [sipx-users] NTP server -

Re: [sipx-users] NTP server - or lack of

urst # server 1.pool.ntp.org iburst # server 2.pool.ntp.org iburst On Thu, Sep 2, 2010 at 3:59 AM, Todd Hodgen wrote: > Actually, I am just wanting the sipXecs server to be the clock source for > the entire network. I thought that the phones would clock off of the > server, and if t

Re: [sipx-users] NTP server - or lack of

Thanks Mike. I’ll give this a try this evening! From: Michael Picher [mailto:mpic...@gmail.com] Sent: Thursday, September 02, 2010 4:00 AM To: Todd Hodgen Cc: Paul Scheepens; Tony Graziano; sipx-users-boun...@list.sipfoundry.org; sipx-users Subject: Re: [sipx-users] NTP server - or lack of

Re: [sipx-users] NTP server - or lack of

: Paul Scheepens [mailto:pscheep...@epo.org] Sent: Thursday, September 02, 2010 12:28 AM To: Tony Graziano Cc: sipx-users; sipx-users-boun...@list.sipfoundry.org; Todd Hodgen Subject: Re: [sipx-users] NTP server - or lack of If I understand the question correctly then Todd needs an NTP source for

Re: [sipx-users] NTP server - or lack of

sting. -Original Message- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Wednesday, September 01, 2010 12:26 PM To: Todd Hodgen Cc: sipx-users Subject: Re: [sipx-users] NTP server - or lack of why wouldn't you just run ntpd running on the demo sipx system and have the phones p

[sipx-users] NTP server - or lack of

I'm wanting to run a sipXecs system without internet access, or access to an external NTP server for Demonstration purposes. Does anyone know a way to work around the need for an NTP server, and keep the time from flashing on end devices. Without an NTP server, the boot process for Polycom phone

Re: [sipx-users] From Header Info Incomplete on Unmanaged Phones

Is it possible that there is a configuration option that is missing for that phone, and the others you are having the issue with, that would cause it to send what you are needing to sipXecs. Maybe looking through a Config file for another phone that is managed might show what option is missi

Re: [sipx-users] Call out from a database list?

The IVR in sipXecs is based on Freeswitch, but is limited to providing Auto-attendant features. Freeswitch does support much more extensive IVR applications such as what you describe, but it's not a part of sipXecs. You might want to dig around to see if anyone else has developed an IVR with thi

Re: [sipx-users] OT RTCP-XR monitor

Yes, but it's not very cost effective for small business from what I can tell. Can't confirm as they aren't real fast at returning calls. From: Martin Steinmann [mailto:mar...@ezuce.com] Sent: Monday, August 23, 2010 2:36 PM To: 'Todd Hodgen'; sipx-users@list.s

[sipx-users] OT RTCP-XR monitor

Polycom has an application suite that creates RTCP-XR voice quality streams for monitoring in a network management application. Anyone have a low cost, or open source product they have used for monitoring this traffic reliably? Looking for something for on customer sites, as well as for running on

Re: [sipx-users] Itsp's tested :: India/Thailand

Don't know 100% of what you are looking for, but VOIP.ms does offer international DID. I've used some European ones successfully in the past. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: S

Re: [sipx-users] SIMPLE SOLUTION FOR ITSP'S WHO SEND TO PORT 5060 Re: iptables experts: port forwarding.

I've used Voip.ms for over 12 months now with sipXecs, and did the interop testing on it for Ranga. It works fine. And, I have several friends that use it as well. But let's be real - for $ .99 a month, what are you really getting? From: sipx-users-boun...@list.sipfoundry.org [mailt

Re: [sipx-users] SIMPLE SOLUTION FOR ITSP'S WHO SEND TO PORT 5060 Re: iptables experts: port forwarding.

Let's not exaggerate here. Sipxbridge has not been available for a couple of years. From my standpoint, I use it with two different ITSP's. It works flawlessly, no problems, nada, zilch. It amazes me the complaints that people make over products they use for free, but have not taken the man

Re: [sipx-users] loop detected in sipxbridge after adding second ITSP to the internal SBC

I seem to recall early on with sipxbridge that if you had multiple gateways pointed to the same IP address, that it created problems. Not like this, but that they were treated like one trunk group. This may have been cleared up in a later release, I don't recall seeing any more discussion of it,

Re: [sipx-users] Remote office (was: port 5060/ port 5080, proxy why?

-Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Matthew Kitchin (public/usenet) Sent: Thursday, August 19, 2010 8:11 PM To: Martin Steinmann Cc: sipx-users@list.sipfoundry.org Subject: [sipx-users] Remote office

Re: [sipx-users] SIP digest auth and caller ID

Don't you want to check mark the Register on Init? I've always found it doesn't work unless that is marked... From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell Sent: Thursday, August 19, 2010 2:19 PM To: Hearty, John

Re: [sipx-users] call park enhancement discussion :: "ring-back" pattern

Tony, Could you transfer the call to a specific extension that has a long time before it goes to voicemail, and then have it ring back to another extension on the console when it isn't answered. Have the party that it is parked for do a call pickup instead of an unpark? Call comes into

Re: [sipx-users] Polycom won't auto discover or even boot on 4.2.1 (was 4.2.1 - Polycom error - Config file error 0x0)

Seems odd the message - "2010-08-17T20:52:16.984000Z":19::INFO:local:main::Servlet:"Generate d Nortel IP 12x0 SIPdefault.xml." if you are generating for a Polycom. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Be

Re: [sipx-users] 4.2.1 - Polycom error - Config file error 0x0

I would agree that it is an issue. The Polycom 450 was supported in 4.0.2. Auto discovery of the Polycom phones was introduced in 4.2.You should be able to get a phone up and running, without a line, and you can actually make calls with it prior to assigning a line to it. Time to write up a

Re: [sipx-users] 4.2.1 - Polycom error - Config file error 0x0

Polycom 450's work great in 3.1.3c. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh M. Patten Sent: Tuesday, August 17, 2010 1:41 PM To: Matthew Kitchin (public/usenet); Tony Graziano Cc: sipx-users@list.sipfoundry.org Subject: Re:

Re: [sipx-users] anyone using a broadsoft call manager based itsp?

I got caught on that one as well. I would swear that the default setting for that changed from 4.0.4 to 4.2.x.I haven't confirmed that fact, but I had a similar issue on an update, and that fixed it. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.o

[sipx-users] FW: failure notice

he SIP side. Once I turned it on with the SIP side, it started behaving like a good little sipXecs. Thanks for the help. From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Tuesday, August 10, 2010 10:56 AM To: Todd Hodgen Cc: sipx-users@list.sipfoundry.org Subject: Re: [

Re: [sipx-users] Question regarding Audio Codes and Freeswitch media

. Once I turned it on with the SIP side, it started behaving like a good little sipXecs. Thanks for the help. From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Tuesday, August 10, 2010 10:56 AM To: Todd Hodgen Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Question

[sipx-users] Question regarding Audio Codes and Freeswitch media

I have an installation that doesn't use a receptionist, and has all calls from one gateway going through the auto attendant. On the auto attendant, straightforward setup, dial one, extension this, dial two, extension this, or dial by name or extension number. If a call comes into the auto a

Re: [sipx-users] Enable and configure conference bridge between SipX and DimDim Webconference Service

Dim Dim integration is already in the 4.2 release. It's not a real tight integration, more of a link between the two system. Enter your dim dim account information in your conference bridge is basically all you do. It creates links you can send to your conference invitee's, you click on your lin

Re: [sipx-users] SIP Trunk Setup questions

IF you can't get the sipviewer working, how about a Wireshark trace of the call coming into your network, or into the server. You can look at the capture from the VOIP call settings, it's very similar to sipviewer. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.si

Re: [sipx-users] SIP Trunk Setup questions

Not sure what configurations have changed, so I would at least check the following - You state that you have sipxbridge set to operator for incoming calls. Do you have operator in the auto attendant? It defaults to being in there along with 0. In the sipxbridge, you can put any extensi

Re: [sipx-users] Awesome - We received a check today

SIPfoundry and the commercial versions that are available from eZuce. Regards, Todd Hodgen From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Martin Steinmann Sent: Monday, August 09, 2010 10:12 AM To: sipx-users@list.sipfoundry.org; sipx

Re: [sipx-users] SIP Trunk Setup questions

You can also use a different Auto Attendant as a conference room Attendant, allowing them to dial the conference room they need to join, and giving them options to transfer to specific conference rooms. With this you can provide a specific voice recording with instructions pertaining to conference

Re: [sipx-users] SIP Trunk Setup questions

I think you are missing a few options on the page. When you open Device/gateway you will get a menu on the left hand side of the screen. Click on ITSP account. You can enter your username and password on that screen. Don't miss the Advanced Options on the upper right hand corner while in that sc

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