Re: [sipx-users] sipX Bridge

Jordan, Great news. I think you have the "ITSP cheatsheet", if you don't just google it, you will find it on the sipXecs site. Follow it step by step, literally. Your ITSP will come up and work for you. The ITSP is no mystery, and Ranga has written up some nice tips that will help you get it

Re: [sipx-users] sipX Bridge

You are correct Scott, thank you. After re-reading your comments, I tried differently. I know that my main issue is not knowing the exact order of configuration. I messed up the remote workers simply because I chose to tie the default SBC to Internet calling when it had no reason to be - disc

Re: [sipx-users] sipX Bridge

Gentlemen, First sorry for the wrong/strong use of the word lecture! I am determined to get this to work. I am certain that my lack of how the system works and my overall lack of experience in VoIP is the primary deterent. The terminology and the interconnection between the technology and mo

Re: [sipx-users] sipX Bridge

Jordan, The emails you called lectures are not that, but good sage advice, not from users, but the actual developers of the components you are having trouble with. If you listen to them, and describe the issue you are having in the manner they are requesting, you will get the best possible suppor

Re: [sipx-users] Connecting to Verizon DS3

I read your guide, and I don't see anything that sticks out at me that I'm doing wrong. Verizon says they are sending 10 and expecting 10 digits. I have enabled and tweaked the default dialing plans. I'm using Cisco 7940s, because I had a few laying around. I am home now, so I just tried Xlite

Re: [sipx-users] sipX Bridge

On Fri, Nov 20, 2009 at 11:24 PM, Jordan Turner wrote: > With a non-configured brand new system except for users and SIP Trunk being > enabled, my remote workers cannot call each other.  It connects to sipX but > the message says it cannot reach the extension.  If my softphones are inside > the

Re: [sipx-users] sipX Bridge

With a non-configured brand new system except for users and SIP Trunk being enabled, my remote workers cannot call each other. It connects to sipX but the message says it cannot reach the extension. If my softphones are inside the office, then I can call each softphones by extension. The on

Re: [sipx-users] sipX Bridge

I know that I did not understand you =) --- On Sat, 11/21/09, Scott Lawrence wrote: > From: Scott Lawrence > Subject: Re: [sipx-users] sipX Bridge > To: "Jordan Turner" > Cc: sipx-users@list.sipfoundry.org > Date: Saturday, November 21, 2009, 4:04 AM > On Fri, 2009-11-20 at 19:39 -0800, > Jord

Re: [sipx-users] sipX Bridge

On Fri, Nov 20, 2009 at 10:39 PM, Jordan Turner wrote: > When I say "test" bridge, I mean the default SBC.  I removed the standard > sipXbride-1 SBC and just entered a new SBC called "test."  That's it.   > Internet calling would use this "test" SBC and calling between remote workers > WORK just

Re: [sipx-users] sipX Bridge

On Fri, 2009-11-20 at 19:39 -0800, Jordan Turner wrote: > When I say "test" bridge, I mean the default SBC. I removed the > standard sipXbride-1 SBC and just entered a new SBC called "test." > That's it. Internet calling would use this "test" SBC and calling > between remote workers WORK just fi

Re: [sipx-users] sipX Bridge

When I say "test" bridge, I mean the default SBC. I removed the standard sipXbride-1 SBC and just entered a new SBC called "test." That's it. Internet calling would use this "test" SBC and calling between remote workers WORK just fine. If I use the default sipXbridge-1, the calls between rem

Re: [sipx-users] sipX Bridge

Hi! I am not sure what a "test bridge" refers to in your mail below. I am not sure what a "direct extension call" is either. Are you attempting to do "Internet calling" - i.e. calling directly via a softphone using a SIP URL? If so you would need to direct that call through an ITSP account which

Re: [sipx-users] sipX Bridge

This sipXecs install is brand new. We are just testing sipXecs out to see if it fits our needs. ALL users are using eyeBeam 1.5.19.x softphones from CounterPath. We are not purchasing any hard phones with this system until we see it work with current setup. The ITSP we are using is Flowroute

Re: [sipx-users] sipX Bridge

What phones are the remote workers using? Are they dialing the itsp did number or the internal line number? Can you provide a call trace? I'm also curious why this is happening with an itsp. I am going to assume the itsp is not configured for port 5060 also is it? Tony

Re: [sipx-users] sipX Bridge

Yes, system informed of the update and automatically installed when I approved. --- On Sat, 11/21/09, Tony Graziano wrote: > From: Tony Graziano > Subject: Re: [sipx-users] sipX Bridge > To: jordan.turner1...@yahoo.com, sipx-users@list.sipfoundry.org > Date: Saturday, November 21, 2009, 2:24 AM

Re: [sipx-users] sipX Bridge

Have you updated to 4.0.4 yet? 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 Customers: http://www.

[sipx-users] sipX Bridge

Do I have to use the default sipxBridge-1 SBC that gets created when you enable the SIP Trunk role?  This works fine when using a SIP Trunk provider, but I can not use it properly for  external clients trying to reach each other that are outside the Intranet unless I delete the default sipXBridg

Re: [sipx-users] smooth 4.0.4 upgrade

Worked great for me too, except I had to modify the yum repo file and remove the .2 off the 5.2 portions of the URLs inside of the file. Other than that, everything went smooth. Oh, and HOORAY FOR NO LONGER HAVING TO MODIFY fallbackrules.xml MANUALLY FOR LOCATION BASED 911 I te

Re: [sipx-users] Feature Codes

sipx-dev, generate some discussion and then file a feature request at track.sipfoundry.org. What you should do is provide a full reason of what you are looking for and why. Some of your functions might be easily addressed without a feature code. Example, dialling "999" to do a feature code for an

Re: [sipx-users] Feature Codes

Not a request. You create it and donate it to the project. The beauty of open source is you can customize it to do whatever you want it to do, you just have to donate what you create back to the community. When you ask for features changes, you are asking someone to donate their time to create i

Re: [sipx-users] Feature Codes

> residential Centrex type service with the Features that are available in a > SIP PBX. Surely, one can’t expect them to be the same. Message retrieval, account controls such as *72, *72, etc. And, adding some testing tools would be great too, echo test, etc. > However, if someone is trying to

Re: [sipx-users] Feature Codes

Precisely what he was looking for. I can see where a superadmin definable table would be really helpful. Heck, I still remember my VM integration days when we could assign any digit we wanted for feature codes in voicemail. "D" for DELETE, "S" for SAVE. Those days are gone, keypads got too small,

Re: [sipx-users] Feature Codes

Are we mixing up the feature codes one would use with a Cell phone or residential Centrex type service with the Features that are available in a SIP PBX. Surely, one can't expect them to be the same. However, if someone is trying to create a commercial offering with sipXecs that includes those

Re: [sipx-users] Connecting to Verizon DS3

I did enable the dial plan, but I definitely need to go through his document and see if I did something wrong. Melting Pot Technologies GMail wrote: > I would check your dial plans here like Tony said. It looks like the > server is telling itself the call is denied. Did you enable the dial >

Re: [sipx-users] Connecting to Verizon DS3

But there is a simpler way to do the dial plan first. Ask Verizon what format your numbers should be. I would assume they would accept the full "+12025551212" (all +1 dialing). If so, the guide I have for bandwidth.comhas all the rules laid out. It should work the same, only the gateway name would

Re: [sipx-users] Connecting to Verizon DS3

Yes. 10.87.20.5 is me. I assumed that is what that meant. I will discuss with them on Monday. Tony Graziano wrote: > I see two things that make me go hmmm > > "2009-11-20T23:11:36.845000Z": > 19:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17::sipXbridge:"Sent > SIP Message :\nRemote Ho

Re: [sipx-users] Connecting to Verizon DS3

I see two things that make me go hmmm "2009-11-20T23:11:36.845000Z": 19:OUTGOING:INFO:nshpbx1.sipx.voip:Thread-17::sipXbridge:"Sent SIP Message :\nRemote Host:10.87.20.5 Port: 38526\nSIP/2.0 403 Forbidden\r\nVia: SIP/2.0/TCP 10.87.20.5;branch=z9hG4bK-sipXecs-0016a7a00a17b94

Re: [sipx-users] Connecting to Verizon DS3

I would check your dial plans here like Tony said. It looks like the server is telling itself the call is denied. Did you enable the dial plan? You can see the different local components communicating in the logs. On Nov 20, 2009, at 6:17 PM, "mkitchin.pub...@gmail.com" wrote: > Not su

Re: [sipx-users] Connecting to Verizon DS3

Not sure if there was a delay of some sort, but I got lots of activity now. It is below. I will dig through them and see what I can find. They announced the routing update (I'm not the network guy) and the equipment did learn it. I can definitely ping their .1 address. "2009-11-20T23:11:36.831

Re: [sipx-users] Connecting to Verizon DS3

How are you/they learning the route to your/their network? Can you ping their .1 address they gave you from the IPPBX? On Nov 20, 2009, at 6:03 PM, "mkitchin.pub...@gmail.com" wrote: > It didn't put anything in a new log file. I've obviously got some > work to do on my end. > In a document

Re: [sipx-users] Connecting to Verizon DS3

It didn't put anything in a new log file. I've obviously got some work to do on my end. In a document I gave them several weeks ago, I did provide them, the IP of my server. Melting Pot Technologies GMail wrote: > Can you run: > > cd /var/log/sipxpbx > > rm -f ./sipxbridge.log > > Make a test ca

Re: [sipx-users] Connecting to Verizon DS3

Can you run: cd /var/log/sipxpbx rm -f ./sipxbridge.log Make a test call, and post the results from sipxbridge.log If your a static configuration they are more than likely pointing to a specific address on your end. Did they say anything about that? On Nov 20, 2009, at 5:25 PM, "mkitchin.pu

Re: [sipx-users] Connecting to Verizon DS3

Yes, only inbound calls. You need to understand what they will send you for did (7/10/11/+11 digits) so you can assign aliases. You need to know what they expect you to send (10/11/+11 digits). This is where you would adjust the dialplan and gateway settings. I'd suggest you look at the section

Re: [sipx-users] Connecting to Verizon DS3

Yes. I just found that under advanced settings. that seems to have gotten rid of that error. Thank you! I still can't make any outbound calls, but hopefully I will be able to find some more logs showing why. The last entries in my sipxbridge log are below. My inbound calls from Verizon are stil

Re: [sipx-users] Connecting to Verizon DS3

Put your main TN in the username, no password. Uncheck register. 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 Helpde

Re: [sipx-users] Connecting to Verizon DS3

Did you uncheck register on initialization? On Nov 20, 2009, at 4:24 PM, "mkitchin.pub...@gmail.com" wrote: > In case you didn't have enough emails from me, here is a little more > info. I put in 123 for the username, so that is obviously where the > 'sip:1...@pcelbcn0001' entry is coming from.

Re: [sipx-users] smooth 4.0.4 upgrade

On Fri, 20 Nov 2009 15:51:53 -0500, Burden, Mike wrote: > Nevermind, I tried it a third time and it worked (maybe someone was > updating the repo?) The update isn't working from the gui either. It goes through the whole process, comes back, same version and I keep getting notices about updates b

Re: [sipx-users] Connecting to Verizon DS3

In case you didn't have enough emails from me, here is a little more info. I put in 123 for the username, so that is obviously where the 'sip:1...@pcelbcn0001' entry is coming from. The service errors out if I don't put in a username and password, but Verizon isn't requesting we use one. mkitc

Re: [sipx-users] Connecting to Verizon DS3

On Fri, 2009-11-20 at 16:06 -0500, Tony Graziano wrote: > Two things: if you plan on having remote workers you ought to try to keep > 5060 for them. Verizon should be able to send to you on 5080. > > I think you need to consider how you route from your 10.x network to your > 172.x provider. Do you

Re: [sipx-users] Connecting to Verizon DS3

I wanted to make sure I understood everything regarding port 5060, but that is what I thought. I will ask them to change that. Our 10.87.20.X network is only separated from the 172.30.209.X network by a router (no NAT or firewall). I was hoping this would simplify things instead of complicating

Re: [sipx-users] Connecting to Verizon DS3

Two things: if you plan on having remote workers you ought to try to keep 5060 for them. Verizon should be able to send to you on 5080. I think you need to consider how you route from your 10.x network to your 172.x provider. Do you have any other services on your private network that natively rou

Re: [sipx-users] Connecting to Verizon DS3

Oh yeah, I'm running 4.0.4 mkitchin.pub...@gmail.com wrote: > Robert Joly wrote: >>> That's weird. The ip they gave you to use is on the same subnet sipx >>> is on? >>> >>> If so, sipxbridge is not needed. Creating an unmanaged gateway would >>> be simpler. >>> >> >> That is true only if Ve

Re: [sipx-users] Connecting to Verizon DS3

Robert Joly wrote: >> That's weird. The ip they gave you to use is on the same >> subnet sipx is on? >> >> If so, sipxbridge is not needed. Creating an unmanaged >> gateway would be simpler. >> > > That is true only if Verizon is capable of handling things like REFERs. > If that is not the c

Re: [sipx-users] smooth 4.0.4 upgrade

Nevermind, I tried it a third time and it worked (maybe someone was updating the repo?) Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone: 616-532-4985 From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Burden,

Re: [sipx-users] smooth 4.0.4 upgrade

Is there an issue with one package in the "Stable" repo, or did I do something wrong? [sipxecs-stable] name=SIPfoundry sipXecs pbx - latest stable version baseurl=http://sipxecs.sipfoundry.org/pub/sipXecs/LatestStable/CentOS/5/ $basearch/RPM gpgcheck=0 # yum --disablerepo=* --en

Re: [sipx-users] Connecting to Verizon DS3

My sipx server is at 10.87.20.5 and the IP they gave me is 172.30.209.62. We get to 172.30.209.X with no firewalls or NAT in place. Would I still go the route of unmanaged gateway vs sipxbridge? Sorry for my ignorance on this. Tony Graziano wrote: > That's weird. The ip they gave you to use is

Re: [sipx-users] Connecting to Verizon DS3

> That's weird. The ip they gave you to use is on the same > subnet sipx is on? > > If so, sipxbridge is not needed. Creating an unmanaged > gateway would be simpler. That is true only if Verizon is capable of handling things like REFERs. If that is not the case then sipXbridge is required to m

Re: [sipx-users] Connecting to Verizon DS3

That's weird. The ip they gave you to use is on the same subnet sipx is on? If so, sipxbridge is not needed. Creating an unmanaged gateway would be simpler. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telep

Re: [sipx-users] Connecting to Verizon DS3

Here are some log file entries that appear relevant to me: "2009-11-20T20:08:02.577000Z":5:OUTGOING:INFO:nshpbx1.sipx.voip:main::sipXbridge:"Sent SIP Message :\nRemote Host:172.30.209.62 Port: 5070\nREGISTER sip:pcelbcn0001.munged.munged.com:5070 SIP/2.0\r\nCall-ID: 8d16619

[sipx-users] Connecting to Verizon DS3

This was an inevitable question from me. I need some help connecting to Verizon SIP over a private DS3. There is no firewall or NAT involved. The information they gave me is below. From Verizon: Inbound calls will route from the 172.30.9.0/24 port 5060 network and you should be able to ping 17

[sipx-users] Reject anonymous incoming calls

Does anyone have a solution for blocking incoming calls that pass no digits but caller ID shows up as "anonymous" or similar? Thank You, Austin ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/arch

Re: [sipx-users] Siemens Openstage Hold/Retrieve Problem

Hi, G711 did not solve my problem, DNS was setup properly, checked via sipx page. I was not able to collect traces, but will as soon as possible(maybe 2-3 weeks) Regards Tony Graziano schrieb: > are you registering your phone via an IP address? did you run your > diagnostics tests and make s

Re: [sipx-users] Overlap Sending with SIP

You are right by what you say with digit matching in the phone, but we have an operlapping german numbering plan, so that you have never fixed lenght numbers so you can never say there are exactly 8 digits, could 7 or 9 also. So we start to dial and the other pbx tells you when enough digits

Re: [sipx-users] am I here?

Cool, I finally exist again. Thanks. On Fri, 20 Nov 2009 11:35:25 -0500, Scott Lawrence wrote: > On Fri, 2009-11-20 at 10:03 -0600, m...@grounded.net wrote: > >>> but showing as lists@ again... >>> >> Silly mailer... better? >> > Arrived as mike... ___

Re: [sipx-users] am I here?

>but showing as lists@ again... Silly mailer... better? ___ 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 I

Re: [sipx-users] sipXecs and Polycom "Buddies"

OK. Sounds like some patience is in order on my part. :) Thanks! Mike Burden Lynk Systems, Inc e-mail: m...@lynk.com Phone: 616-532-4985 -Original Message- From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Friday, November 20, 2009 9:39 AM To: Burden, Mike Cc:

Re: [sipx-users] sipXecs and Polycom "Buddies"

Only if your users have a softphone also. Again, this would work in 4.2 with the bria phone. In this current stable release there are still bugs being worked out that prevent the workgroup feature from working properly on the bria. So, to answer your question: right now there is not a way to subsc

[sipx-users] smooth 4.0.4 upgrade

Just did the upgrade to 4.0.4 without any issues. Thanks for all the good work! Dan ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://list.sipfoundry.org/mailman/

Re: [sipx-users] sipXecs and Polycom "Buddies"

OK. That's not one of my "must have" features, so I can wait for 4.2 stable release and use an app on the PC. Are there any Windows-based (sorry) programs that simply register with the SIP server (like a soft-phone) and monitor BLF without doing too much else (ie, they have a very simple user

Re: [sipx-users] Question regarding resource usage...

To a certain extent, within a few hours or days, it is not at all unusual for sipx to assume use of 100% of your available RAM and keep it there. I remember asking this ages ago and got a great explanation from Dale, but I can't easily find that message. I think it caused so many unnecessary quest

[sipx-users] Question regarding resource usage...

I've been trying out sipXecs and as is common these days I've been doing so in a variety of virtual environments: 1. VMware ESX 3.5 and vSphere 4 2. VMware Workstation 3. KVM 4. OpenVZ (container "virtualization") on CentOS or Proxmox (Debian 5) My question is: Does this *really* need this much

Re: [sipx-users] sipXecs and Polycom "Buddies"

Correct. Upcoming (4.2, development version 4.1 bit not near stable yet), is an IM integration which will allow for an IM client to "watch" the phone. Pick up the handset and make a call, show your watcher that you are on the phone. No real limitations there yet, but surely more than a few dozen I'

Re: [sipx-users] Calls goes through but no audio

> we have installed 4.0.2-016420 on Centos 5. We have applied > latest patch for sipXbridge too (4.0.4). > > We are experience a problem with audio on our calls. > > When we do internal calls from extension to extension phones > are ringing, but we don't have audio when we pickup the > phone

Re: [sipx-users] sipXecs and Polycom "Buddies"

BLF only tracks as many people as I've got buttons for on the phone, right? The Polycom IP550 has 4 buttons, the registration takes up one, so I can only track three people, and I want to track at least five. Sorry if I'm being stupid... I've got a total of about 6 weeks experience in IP telepho

Re: [sipx-users] Openfire

So this means any IM client expecting a SRV record of: _jabber._tcp 20 0 5269 will not work. That is understandable, since it means a very old IM client anyway, and there are several good opensource/paid commercial ones out there that use the standard SRV of "xmpp-client" on port 5222. I was won

Re: [sipx-users] SIPX Recommendations

I also prefer the patton. I prefer factory direct support, the very flexible CLI. I find it very stable too. There are sample configs on the wiki. On Fri, Nov 20, 2009 at 5:40 AM, Picher, Michael wrote: > Maybe… but I use Patton. :-) > > > > I think they might be the second most commonly used

Re: [sipx-users] Openfire

There is only one way for IM clients to connect to openfire and that is through the XMPP protocol on TCP port 5222. The XML-RPC ports are used by the sipXecs applications and are of no use to IM clients. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx

Re: [sipx-users] sipXecs and Polycom "Buddies"

Buddy lists are not supported. Why don't you use BLF instead? 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

[sipx-users] sipXecs and Polycom "Buddies"

Whenever sipXecs reboots one of our Polycom IP550 phones, the phone loses its "Buddies" list. Is there a way to load the Buddies list in sipXecs (maybe under "Device Files"?) so that it will repopulate the Buddies list on the phone after a reboot? It's OK if everyone has the same Buddies list.

Re: [sipx-users] 4.0.4 upgrade and sipxbridge pacth

Sorry. After more coffee and re-reading I understand patch23 is a backwards step after 4.0.4 is installed, so further patching sipxbridge is not necessary. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telepho

[sipx-users] Release 4.0.4

Release 4.0.4 is posted - since we never formally designated 4.0.3 as 'LatestStable', the following list is all the fixes since 4.0.2: Bugs Fixed [XX-6412] - Location based routing does not work for emergency dialplans [XX-6551] - Calls through sipXbridge to scstrial.us get disconnected

Re: [sipx-users] 4.0.4 upgrade and sipxbridge pacth

On Fri, Nov 20, 2009 at 7:24 AM, M. Ranganathan wrote: > On Fri, Nov 20, 2009 at 4:59 AM, Tony Graziano > wrote: >> When installing sipxecs 4.0.4 update, what level of the sipxbridge pacth is >> the equivalent to? patch 23? > > > I fixed two more issues after that having to do with Max-Forwards >

Re: [sipx-users] Caller ID

On Thu, 2009-11-19 at 17:14 -0500, Tony Graziano wrote: > I have a need for a specific group of people to call out without displaying > a caller-id. Until I determine whether the carrier is blocking the ability > to send "anonymous" (when I send anonymous from the PRI gateway it shows the > full BT

Re: [sipx-users] Overlap Sending with SIP

On Thu, 2009-11-19 at 22:09 +0100, Kalle Tetto wrote: > Hi, > > are there any plans to implement overlap sending and receiving with > sipx? That would be a nice feature for germany, austria and denmark. > > I know that asterisk hast support and patton inalp. I have spoken to a > Cisco router de

Re: [sipx-users] 4.0.4 upgrade and sipxbridge pacth

On Fri, Nov 20, 2009 at 4:59 AM, Tony Graziano wrote: > When installing sipxecs 4.0.4 update, what level of the sipxbridge pacth is > the equivalent to? patch 23? I fixed two more issues after that having to do with Max-Forwards header missing from an ITSP inbound INVITE and consultative transfe

Re: [sipx-users] In-call status?

Counterpath's response: "We support BLF (RFC 4235) with Bria Professional. We have done extensive testing with the team producing sipXecs so there should be no problem. Note that the feature must be enabled via our provisioning mechanism." AFAIK, that's only one Counterpath product from the 4 payi

Re: [sipx-users] Caller ID

I've not heard of any... -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Thursday, November 19, 2009 5:15 PM To: sipx-users@list.sipfoundry.org Subject: [sipx-users] Caller ID I have a need fo

Re: [sipx-users] SIPX Recommendations

Maybe... but I use Patton. :-) I think they might be the second most commonly used solution! From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dylan Ebner Sent: Thursday, November 19, 2009 12:20 PM To: sipx-users@list.sipfoundry.org

Re: [sipx-users] Feature Codes

True... our feature codes are certainly limited. From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Thursday, November 19, 2009 8:28 PM To: Picher, Michael Cc: li...@grounded.net; sipx-users Subject: Re: [sipx-users] Feature Codes Yes, but... (aw daddy, you said butt), I thin

[sipx-users] 4.0.4 upgrade and sipxbridge pacth

When installing sipxecs 4.0.4 update, what level of the sipxbridge pacth is the equivalent to? patch 23? Thanks, Tony ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: htt

[sipx-users] Calls goes through but no audio

Hi, we have installed 4.0.2-016420 on Centos 5. We have applied latest patch for sipXbridge too (4.0.4). We are experience a problem with audio on our calls. When we do internal calls from extension to extension phones are ringing, but we don't have audio when we pickup the phones. In this cas

Re: [sipx-users] sipXecs bridge or Dial Plan or other?

For kicks, I have a complete how-to for pfsense. http://blog.myitdepartment.net/?p=37 You might want to look it over and compare your config for rules/nat. In eyebeam/xlite you should have discover global address, discover server, ice unchecked. The domain f

Re: [sipx-users] yum update to 4.0.4

4.0.4 has several bug fixes. If you are going to use sipxbridge, I think it's better to update to 4.0.4. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Winson (Elabram) Sent: Friday, November 20, 2009 12:24 PM Cc: sipx-us

Re: [sipx-users] yum update to 4.0.4

Is better update to 4.04? because i just start configure in 4.02.. Any change? Nikolay Kondratyev wrote: Uups… please ignore my previous message. I’m sorry. There was wrong HTTP_PROXY … fat fingers..       From: sipx-users-boun...@list.sipfoundry.org [ma

Re: [sipx-users] yum update to 4.0.4

Uups. please ignore my previous message. I'm sorry. There was wrong HTTP_PROXY . fat fingers.. _ From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nikolay Kondratyev Sent: Friday, November 20, 2009 12:02 PM To: sipx-users@

[sipx-users] yum update to 4.0.4

Hi all, I tried to update from 4.0.2 to 4.0.4 via yum. But I got the following error: [r...@sipx4 yum.repos.d]# yum --disablerepo=* --enablerepo=sipxecs-stable list available Loading "fastestmirror" plugin Loading "basearchonly" plugin Loading "downloadonly" plugin Determining fastest mir