Re: [sipx-users] deployed new sipx server....3 phone extensions, only 1 is accessible inbound

Francis, I think that the quickest path to understanding your problem would be to provide either a sip trace as Scott requested or provide a network trace by running tcpdump -n -nn -s 0 -i any -w remote_user_problem.cap directly from your sipXecs. What kind of router is at the remote location and

Re: [sipx-users] deployed new sipx server....3 phone extensions, only 1 is accessible inbound

>> Hi >> >> I deployed another sipx server (server is remote >> located with public IP (firewall is within server, no NAT). >> I provisioned 3 polycom phones. All 3 phones are created via >> phone group, so all have the same settings. All 3 phones >> have successfully regist

Re: [sipx-users] deployed new sipx server....3 phone extensions, only 1 is accessible inbound

the sipx server is plugged in directly to our switch, no firewall in between, also no NAT. that is what puzzles me. there should not be any issues with nat traversal on the server end. I'll try the tcpdump and update you On Mar 9, 2010, at 10:13 AM, R

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

to be sent to gateway configured in your dialplan. You do not need to use your ITSP to do that - just create an unmanaged gateway pointed for 'tf.voipmich.com' and use it as the gateway for your dialplan and 18004664...@tf.voipmich.com when you dial 411. > > > On 3/4/10 10

Re: [sipx-users] Only a limited set of characters is valid in SIP URLs. Spaces and some special characters are not allowed. For example: 321, joedoe, joe!?~*222

Do not put the domain in your resulting call, only digits belong there. Put 18004664411 in the resulting call - this digit string will be sent to whatever gateway you have configured for this dialplan. bob From: sip

Re: [sipx-users] [sipX-dev] sipxbridge and remote users without server behind nat

Re: [sipX-dev] sipxbridge and remote users without server behind nat On Wed, Mar 3, 2010 at 4:20 PM, Robert Joly wrote: > -Original Message- > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net]

Re: [sipx-users] [sipX-dev] sipxbridge and remote users without server behind nat

gt; server behind nat > > > > On Wed, Mar 3, 2010 at 4:01 PM, Robert Joly wrote: > > > > Does sipxbridge have to be behind nat in order for remote > > users or trunking to work? > > > Yes as long as you uncheck the 'Server be

Re: [sipx-users] [sipX-dev] sipxbridge and remote users without server behind nat

> Does sipxbridge have to be behind nat in order for remote > users or trunking to work? Yes as long as you uncheck the 'Server behind NAT' checkbox under System->Internet Calling->NAT Traversal. BTW, just to make sure, sipXbridge and remote NAT traversal are two different beasts - please see qu

Re: [sipx-users] only one phone works in remote location

> I have a SIPX server in a co location @ 100mb internet, I > have around 35 locations all have multiple phones, and all > those phones work. > > One location however, only one phone connects properly, So > let me explain, > > Ext 210 and 211 are in a remote location, If I plug both of > th

Re: [sipx-users] Remote User

make a wiki page :-) > > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979) 361-4676 > > > On 2/26/2010 8:07 AM, Robert Joly wrote: > > I have created a new and improved remote user information > page in the > > new wiki. This

Re: [sipx-users] Remote User

I have created a new and improved remote user information page in the new wiki. This new one is a superset of the old one that Tony kindly pointed you to. http://wiki.sipfoundry.org/display/xecsuserV4r0/Remote+User+NAT+Traversa l > -Original Message- > From: sipx-users-boun...@list.sipf

Re: [sipx-users] List of features for 4.2

> http://sipx-wiki.calivia.com/index.php/Detailed_Roadmap_sipXecs_IP_PBX > > I was wondering how accurate the roadmap is. From what I can > tell DimDim isn't in 4.1.6 and I can't get IM presence based > routing to work either. > > Any info available? Kinda want to do a "technology preview" > f

Re: [sipx-users] Were we hacked?

> Hi again. Yesterday we changed the password for this user > that was compromised, but last nigh around 1 o'clock i see > that there were another tries to call through our system. > Something is really going wrong. As Tony explained this looks > like something related with sipxproxy. The only

Re: [sipx-users] Far End NAT Support

vations. > > Thanks, > Max > > On Mon, Feb 22, 2010 at 1:05 PM, Robert Joly wrote: > >> Robert, > >> > >> I removed the STUN configuration from the ATA and reset the device. > >> The behavior is the same. > > > > What about the L

Re: [sipx-users] Far End NAT Support

s like for your ATA? Basically, if you see a 'sipx-nonat' in there, a SIP ALG in the router or a STUN client in you ATA is at work. > > -Max > > On Mon, Feb 22, 2010 at 12:36 PM, Robert Joly wrote: > >> Hello, > >> > >> I have a sipx server run

Re: [sipx-users] Far End NAT Support

> ensure internet calling is disabled at the sipx server. > ensure the Ip address/subnet is unique and not listed in your > intranet subnets in sipx. Not that it has anything to do with the problem being reported here but remote private networks and the local private network (the a private sipXe

Re: [sipx-users] Far End NAT Support

> Hello, > > I have a sipx server running on a statically assigned public > ip address. I need to support far end NAT for remote users > (ATAs behind a home firewall). I have a Grandstream HT 286 > that we are experimenting with. Initially I connected the ATA > to our office DHCP network, auto

Re: [sipx-users] calls not disconnecting

> I just noticed when looking at the call details records that > I have a call in progress, and it's been running for almost > an hour now. The call was the one I made earlier that was > not answered, and I have already hung up on my end as well as > the other end. So how come it's still runn

Re: [sipx-users] RTP telephony event proxy using internal SBC

> > Does sipxBridge attempt to generate in-audio DTMF tones? It does not. RTP is passed though untouched. 8< ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users Unsubscribe: http://li

Re: [sipx-users] Weird eyeBeam / NAT issue

> There was only one firewall that they did not find any security > vulnerabilities in. :) There is nothing more secure than a down firewall :) ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/arc

Re: [sipx-users] Weird eyeBeam / NAT issue

> -Original Message- > From: Burden, Mike [mailto:m...@lynk.com] > Sent: Thursday, February 11, 2010 12:52 PM > To: Joly, Robert AVAYA (CAR:9D30); sipx-users@list.sipfoundry.org > Subject: RE: [sipx-users] Weird eyeBeam / NAT issue > > > Reboots?! Yikes. If you go back to the old ver

Re: [sipx-users] Weird eyeBeam / NAT issue

> First, a little topology: > > Our sipXecs server is on a DMZ with the address > 192.168.9.1/24. The firewall address on that DMZ (which is > also the default route for the sipXecs server) is 192.168.9.254. > > We have a workstation on the firewall's protected network > with the address 192.

Re: [sipx-users] Remote worker config vs sipxbridge to ITSP

> Unless you specify an SBC other than sipXbridge for Internet > calling, enabling it using sipXbridge breaks remotes users in > some functions (cant receive calls, but can make calls, for example). This is definitely not how it is supposed to work. We use such a configuration every day here w

Re: [sipx-users] Remote worker config vs sipxbridge to ITSP

> > I've seen a bunch of traffic on the list implying that to use > an ITSP you have to turn off 'Internet calling'. Correct, when sipXbridge is used, Internet callign should not be used. > > Should I be able to follow the cheat sheet on the wiki wrt > remote workers and have it not break m

Re: [sipx-users] sipXproxy service Problem

I saw something similar in the past and had raised a tracker for it which since got fixed - please check out http://track.sipfoundry.org/browse/XX-5904 . The fix for this got delivered in revision 15786 but you are running an earlier revision (14059).

Re: [sipx-users] DND Presence and Intercom

> -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > Lawrence, Scott AVAYA (BL60:9D30) > Sent: Wednesday, January 20, 2010 8:44 AM > To: Todd Hodgen > Cc: 'sipx-users'; Fowler, Peter AVAYA (CAR:9D10) > Subj

Re: [sipx-users] Forwarding issues with Polycom handsets

have cases where someone would definitely want to forward their calls and have them go to the other person voicemail as well. Thank you for the info! -Matthew On 1/13/2010 2:31 PM, Robert Joly wrote: I suppose that if none of your dialplan rules r

Re: [sipx-users] Forwarding issues with Polycom handsets

definitive "it can't be done"? I would like to be able to present all the options. There are only a few users I would like to be able to enable it for. I would like to be able to give them all the options, and then explain why they should do what. On 1/13/2010 2

Re: [sipx-users] Forwarding issues with Polycom handsets

This is another case of stale information on the wiki. This statement no longer applies. Also, the use of set-based call forwarding features is strongly discouraged as its usage brings about the permission problems that you are seeing. Instead, sipXconfig or the user portal should be used to co

Re: [sipx-users] How to set up a non-NAT'd sipXecs IP PBXwithoutgiving public addresses to the phones

Generally, you need to enable the remote NAT traversal feature when at least one of these conditions is true: 1- Your sipXecs has a private IP address and is deployed behind a NAT; AND/OR 2- At least one phone is deployed behind a remote NAT (e.g. home user, remote branch office, ...) The remote

Re: [sipx-users] FXS Registration

Did you 'burn' the configuration? Changes made to AC via its web GUI will not be persisted across power cycles unless you 'burn' it. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Derek Bartolo

Re: [sipx-users] Remote user

Run the following command on sipXecs: tcpdump -n -nn -s 0 -i any -w failedcall.cap then reproduce the problem. Once this is done, ctrl-c out of tcpdump and send e-mail me directly the capture file along with a snapshot. Cheers, bob From: sipx-users-bou

Re: [sipx-users] INVITE Request from SIPX

The x-sipX-nonat URL parameter is added by the NAT traversal subsystem of sipXecs and SIP endpoints should be capable of simply ignoring those. I have tested this with many, many SIP endpoints and they all are capable of handling this properly. Are you finding that these are causing problems with

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] 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] 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] MWI (message waiting indicator)

http://track.sipfoundry.org/browse/XTRN-892 From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of jermaine pinder Sent: Wednesday, November 04, 2009 12:18 PM To: Lawrence, Scott (BL60:9D3

Re: [sipx-users] Server behind Two NAT

Hi, As you have discovered, you can only configure a single publicly reachable IP address per sipXecs. If you like to try new things, you could set up a high-availability (HA) system composed of 2 sipXecs behind your router and configure each sipXecs with its own public IP address. On paper, th

Re: [sipx-users] Dial Plan Bug?

If you specify the number of digits in your FXO dialplan rule, you should be able to lift the ambiguity. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Derek Bartolo Sent: Friday, October

Re: [sipx-users] Server Behind NAT

Does Internet Calling need to be enabled? I'm still unclear about its functionality. Thanks, Ken From: Robert Joly [mailto:rj...@nortel.com] Sent: Monday, October 26, 2009 11:53 AM To: Ken Fulmer; sipx-

Re: [sipx-users] Server Behind NAT

The local phones can continue to access the sipXecs using the private IP address. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Fulmer Sent: Monday, October 26, 2009 12:27 PM

Re: [sipx-users] SIPX remote worker setup

hanks Ashwin From: Robert Joly [mailto:rj...@nortel.com] Sent: Monday, October 19, 2009 6:45 PM To: Ashwin Kumar; sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] SIPX remote worker

Re: [sipx-users] SIPX remote worker setup

You can indeed use your LAN as a way to simulate the 'public Internet'. All you really need to test basic remote worker is a 2 NATs. 2 phones and an sipXecs. Connect the WANs of the two NATs together by connecting them to the same network (LAN, public internet, x-over cable). Put one phone behin

Re: [sipx-users] SIPx Questions

Hi Ken, Please see my responses inline. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ken Fulmer Sent: Monday, October 12, 2009 12:09 PM To: 'sipx-users' Subject: [sipx-us

Re: [sipx-users] Problem with communication between remote workers

Pavel, I received your SIP traces and took a look at them. It appears to me that the problem is caused by a bug in your C475 phone. According to section 12.1.1 of RFC3261, " When a UAS responds to a request with a response that establishes a dialog (such as a 2xx to INVITE), the UAS MUST copy

Re: [sipx-users] Problem with communication between remote workers

Hi Pavel, The configuration you mention is supposed to work. Please double-check that you have disabled any kind of SIP ALG in all your routers (e.g. your linksys). If that is the case and you are still facing issues, please run the following tcpdump command on the sipXecs box then replicate t

Re: [sipx-users] Nat Traversal

Ken, What version of sipXecs are you using? When you say that the phones can't call eachother, do you mean that the phone does not ring when being called or that you are having media issues? bob From: sipx-users-boun...@list.sipfoundry.org [mailto:sip

Re: [sipx-users] NAT set up questions

> Hi all, > > I'm running the following: > > > sipXconfig (4.1.0-016351 2009-09-02T16:01:46 ecs-centos5.x64) > > > I'm having an issue where I can't call my registered UA's. > The INVITE just never hits the phone. Outbound calls work fine. > > > Here is an example of my registered Polyco

Re: [sipx-users] Get registration status of phones via SOAP/REST oranything else

> Hello list members, > as we want to observe registration status of phones by an > external application I just wanted to ask if there is a > (simple) way to poll registration status of all phones on > SipX Registrar. sipXecs supports rfc3680 and thus allows a client to SUBSCRIBE for registrati

Re: [sipx-users] Remote users and Firewalls

> After testing for a solid week, we're floored to come to a > realization... MOST hot spot providers are either blocking > voip or don't have ALG or other problems which prevent voip > from working properly. The main culprit was one way audio. > > So, after visiting at least a dozen sites and

Re: [sipx-users] ACD sends call to agent on wrong IP

James, I would like to help you but in order to do so, I'll need more precise info. Please see my comments inline. > Sent: Tuesday, September 29, 2009 9:32 PM > > Our agents pass through a box that has 2 physical Ethernet > interfaces before getting to sipX. > 1 DMZ and 1 inside Wouldn't th

Re: [sipx-users] Call_Park problem and Call_Pickup problem

What kind of phone is 255? It looks like this phone does not support dialog event packages which are required by the call pickup feature. Make sure you are running the latest and greatest firmware available for this phone. From: sipx-users-boun...@list

Re: [sipx-users] Dialing Extensions Internally

James, Try disabling GRUU on your phones to see if that helps. sipXecs seems to have difficulty routing the ACK because of it. We will have to look more closely at this problem. bob From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun.

Re: [sipx-users] SIP Trunk with Teleworker problem

Michael, This is an odd problem. What happens if you call the auto-attendant from a local extension and then dial the remote user extension on the AA? Let me know, bob From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoun

Re: [sipx-users] Source Call Routing being ignored in Emergency dialrule

> Situation: 2 gateways set up for two different locations. One > user group is set to send calls (local and emergency) out of > one gateway, the other group out the other gateway. sipX will > obey the source call routing rules on the local dial plan > rule and send the call out the appropriate

Re: [sipx-users] Source Call Routing being ignored in Emergency dialrule

> Situation: 2 gateways set up for two different locations. One > user group is set to send calls (local and emergency) out of > one gateway, the other group out the other gateway. sipX will > obey the source call routing rules on the local dial plan > rule and send the call out the appropriate

Re: [sipx-users] Provisioning for Remote Workers

> hi, > > I am running a SipX 4.0.1 and we have around 20 Remote > Workers with a Polycom 550. Is there any possibility to run a > provisioning service for the phones, even they are behind a NAT? By default, polycom phones are provisionned via FTP so sipXecs runs an FTP server for that purpose.

Re: [sipx-users] Experiences with SipX, Grandstream GXW4104 FXO gateway, Snom 320 phones, a couple of firewalls, and remote worker setup.

ssage- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of > Robert Joly > Sent: Monday, August 24, 2009 9:43 AM > To: Dennis Wallen; sipx-users@list.sipfoundry.org > Subject: Re: [sipx-users] Experiences with SipX,G

Re: [sipx-users] Experiences with SipX, Grandstream GXW4104 FXO gateway, Snom 320 phones, a couple of firewalls, and remote worker setup.

> > Over the past few weeks I've been considering SipX as a > replacement for our existing phone system. My greatest > interest is in the remote worker configuration to help > support "virtual office" functionality. I don't really have > any questions, but thought I'd share my experience in

Re: [sipx-users] architecture decisions for new install

> Hey folks, > > I want to set up a SipX server for my new employer, and I > have a few decisions to make on where to put it. > > Most of our employees are here at the main office in Ottawa - > about 15 or 16 of us. > > We also have a half dozen remote workers in the US and Asia. > > Coming i

Re: [sipx-users] Cisco 7971 Behind NAT

I personally never tried the sipXecs remote worker feature with Cisco phones behind NAT and I don't think I ever heard of anyone on this list attempting it therefore I cannot provide specific hints. I can however give you the usual general advice that any SIP ALGs, STUN or ICE must be disabled in

Re: [sipx-users] Phones connect, ring, but no voice.

> Ok, which router would you recommend then? > I did the bulk of my testing using a Nortel BSR222. Others on this list will probably be able to propose other alternatives. > > > -Original Message- > From: Robert Joly [mailto:rj...@nortel.com] > Sent: Monday, Au

Re: [sipx-users] Phones connect, ring, but no voice.

> > Apparently, you cannot disable these settings on the routers, > Just spoke with Netgear, any suggestions of which > routers/switches I should use that are moderately priced? That is too bad because I know some prosafe models that have the option to disable the SPI under advanced WAN setting

Re: [sipx-users] Phones connect, ring, but no voice.

> Well, > > > > I have 2 phones (Polycom 430) connected and extensions work, > However, when I call from extension to extension, there is a > ring, you pick up but no voice. > > > > My scenario is as follows > > COLO SIte > > > > Server: dual core with 2 gb ram > > Router: Netgear P

Re: [sipx-users] IMDB database query.

> M. Ranganathan wrote: > > On Fri, Aug 7, 2009 at 3:54 PM, Damian > Krzeminski wrote: > >> Dale Worley wrote: > >>> On Fri, 2009-08-07 at 11:38 -0400, M. Ranganathan wrote: > I would like to get some pointers on how I can access > the In Memory > Database that holds the SIP passwords

Re: [sipx-users] remote worker can't register

> On Tue, 2009-08-04 at 23:59 +0200, Peter Selc wrote: > > Hi, > > > > attached is the sip log from registrar. The extension > behind NAT was 202. > > >From that log file: > > "2009-08-04T21:46:53.644866Z":1194:AUTH:ERR:sipx.company.local > :SipRegistrarServer:B6CB5B90:SipRegistrar:"Unable to g

Re: [sipx-users] Remote worker cannot be terminated

> > OK, let me try to elaborate my testing network configuration > related to the feature of remote worker calling of sipx. > > I put sipx behind my lab Firewall/NAT box which has an ADSL > internet access with a public IP. My sipx stays in intranet 1 > of 192.168.2.0/24 and I have another in

Re: [sipx-users] remote worker can't register

> > Hello, > > i installed and setup sipxecs and i'm able to register phones > from local subnet and make local calls + calls to siptrunk > (alcatel oxe). > > I would like to use remote workers which are behind NAT. I > followed the "configuring remote workers cheatsheet". The > public IP is

Re: [sipx-users] Remote worker cannot call to second intranet subnets

or these two subnets since sip phone inside them can > make call to each other. Both of them are disabled SIP ALG. > > -----Original Message- > From: Robert Joly [mailto:rj...@nortel.com] > Sent: Tuesday, July 21, 2009 10:06 PM > To: jun,wen; sipx-users@list.sipfoundry.org >

Re: [sipx-users] Remote worker cannot call to second intranet subnets

gt; > -----Original Message- > From: Robert Joly [mailto:rj...@nortel.com] > Sent: Tuesday, July 21, 2009 9:16 PM > To: jun,wen; sipx-users@list.sipfoundry.org > Subject: RE: [sipx-users] Remote worker cannot call to second > intranet subnets > > After making

Re: [sipx-users] Remote worker cannot call to second intranet subnets

nside the intranet subnet of SIPX > (192.168.2.0/24). The issue only happens to the call to other > intranet subnets (192.168.8.0/24 and 192.168.10.0/24). > > -Original Message- > From: Robert Joly [mailto:rj...@nortel.com] > Sent: Monday, July 20, 2009 11:00 PM &g

Re: [sipx-users] Remote worker cannot call to second intranet subnets

the sipXecs NAT traversal feature usually breaks things. So, the morale of the story is that if you choose to rely on sipXecs solving your NAT traversal issues, please disable any other NAT traversal features in your deployment. > > -Original Message- > From: Robert Joly

Re: [sipx-users] Remote worker cannot call to second intranet subnets

> intranet subnets > > Yes all of my three intranet subnets, 192.168.2.0/24, > 192.168.8.0/24 and 192.168.10.0/24, were already set in the > "Intranet Calling". > > -Original Message- > From: Robert Joly [mailto:rj...@nortel.com] > Sent: Monda

Re: [sipx-users] Remote worker cannot call to second intranet subnets

Please make sure that your 192.168.8.0/24 and 192.168.10.0/24 appear in your intranet subnet list under System->Intranet Calling. The rule of thumb here is that the Intranet Subnets section should contain all the subnets that describe your intranet. > > Hi, I have three intranet subnets by a cen

Re: [sipx-users] InGate/siptrunk module and sipxBridge config issues

I once tried to disable SIP features in InGate to turn it into a non-SIP-aware vanilla Firewall but I found that I couldn't find the recipe that would totally kill everything that is SIP-related inside the box - disabling the SIP module it surely not enough. Perhaps it can be done but I would sugg

Re: [sipx-users] SIP route to SIP_DOMAIN_NAME does not match theconfigured IP address

This can happen if your SIP domain does not resolve to the IP address of your box. Please do a DNS lookup (using 'dig' utility of instance) on your SIP domain and make sure that the answer you get is the IP of your box. If not then either change the DNS record or the IP of your box to resolve the

Re: [sipx-users] Remote Worker registering with "sipX-nonat"

> Are there any special phone settings for a remote polycom > handset to register appropriately? I say this because I see > the registration showing the remote public ip address from > the ISP but not announcing itself as being being NAT. Is this correct? This indicates that some device is tryi

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

> I'm testing a SIP trunk to Covista Communications. I'm able > to receive and send calls but there's two catches: > > 1. Placing a call from internal to my cell phone, I do not > hear ring tones but the cell phone will ring and caller ID is > shown properly. If I answer the cell I do not hear

Re: [sipx-users] wrong refer-to

> Hello all, > > recently my sipxpbx ( v3.8 ) start doing strange thing. > when someone call my main line, he is answered by > autoattendand, then user type extension 200 ( direct > extension ), call blind transferred to phone. > Since yesterday user is back in autoattendand. I was doing > pack

Re: [sipx-users] Scalability figures

Hi Gabor, Back in February I conducted an experiment that aimed at finding how many call attempts per second (CAPS) could be achieved on a single sipXecs. The setup involved a series a SIPp scripts simulating load and was executed on a single Dell PowerEdge R300 server equipped with 2.50GHz Quad-c

Re: [sipx-users] NETGEAR ProSafe VPN Firewall model FVX538v2

Clock Information Systems -Original Message----- From: Andreas Sent: Tuesday, June 23, 2009 5:19 PM To: 'Robert Joly' Subject: RE: [sipx-users] I'm compiling a list of routers. . .

Re: [sipx-users] Thoughts on Silence suppression with SipTrunks

> Just wondering how many people are using silence suppression > over a sip trunk? I had a sip trunk provider tell me today > that silence suppression is never a good idea for sip trunks. > Of course, he couldn't really tell me why with any detail. > > So, I'm just wondering what others are d

Re: [sipx-users] VAD Threshold Adjustment In Conference

> Hi. > > I am using the conference, and have noticed that the voice > detection threshold seems too high (i.e. it's not passign > lower volume audio of some of the "quiet" people talking. > > Does anyone know where this threshold might be adjusted? Can > this change be made on the fly? > K

Re: [sipx-users] [Bulk] Re: Remote worker registration problem

> Thank you, Scott! > > I informed this user to check this router settings and he has > found out some settings for sip on his router. After > disabling these sip settings everything is ok and he is able > to register :) May I ask what kind of router you are using? I'm compiling a list of rou

Re: [sipx-users] Remote worker registration problem

> Hi, > > i have installed 4.0.0-015321 version on our centos 5 test > server and i have a very strange issue with one of our users > that try to register to sipx. > > Everyone is able to register to sipx, except him. What > happens is that he gets unauthorized message when he try to > regist

Re: [sipx-users] NAT Telephony

> My topology is... > > > WAN  = cicso pix 501 firewall  === DMZ > (subnet) Microsoft ISA 2006 firewall/proxy > === Internal Network > > DMZ network spans 2 subnets.  They are 192.168.4 and > 192.168.5.  The sipx server's address is 192.168.4.25. > > I have su

Re: [sipx-users] Interconnection 2 sipXecs

> Hi All, > > I have several questions about interconnection between 2 > sipXecs server > > > > Here are the scenarios: > > 1. sipX server A configured for the HQ in the Asia and > providing VoIP services for the local users. > > 2. sipX server B configured for the Branch offic

Re: [sipx-users] sipXecs 4.0 Registration status for Remote Workers

> Hi All, > > I have a question about registration status in diagnostic tab > for remote worker. > > > > 1. If a remote user registered to the server with > "3...@114.58.107.77:26116;rinstance=61e4a43ba568d13d;x-sipX-non at" status, the remote user able to call the others extension > a

Re: [sipx-users] Routing calls through an intermediate proxy

> Hi all, > > Quick question: Is it possible to configure SipXecs to force > ALL outgoing SIP transactions through an intermediate proxy? > This without prior knowledge of SIP domains the proxy may > have to interact with? I see that I can configure a SIP > firewall for a SIP trunk, but thi

Re: [sipx-users] Media Relay Service Failed

> Right now, I'm still open all UDP & TCP port for the SIPX server. > I used static ip address for the SIPX server NAT > > > > -Original Message- > > From: Robert Joly [mailto:rj...@nortel.com] > > Sent: Friday, May 15, 2009 8:25 PM > > To: B

Re: [sipx-users] Media Relay Service Failed

> Hi All, > I just found out that, after I put back the NAT port to > default configuration (3-31000), the remote worker > support can't run anymore. > Internal user able to called remote user and vice versa. When > the call established remote user can hear the voice from > internal user bu

Re: [sipx-users] Public IP Requirement is mendatory in ITSP andRemoteRegistration?

> Thanks Tony. > > But i want to know simply that if i configure sipx on local > ip like sip 192.168.x.x and there is a dsl router comes in > picture with dynamic ip given by my isp then sipx will work > for itsp registration in that case. Any time you deploy sipX in a private network behind a

Re: [sipx-users] Bria Pro hangs up after 25 seconds using siptrunk

> The BRIA installation was on a different IP address, not > behind NAT at the far end. That never had an issue. > > Counterpath made us get a new license because ours wouldn't > work, we had to download a new build today. > > When a call is placed audio is heard in both directions. DID > call

Re: [sipx-users] Bria Pro hangs up after 25 seconds using siptrunk

> > I am testing a 4.1 system right now. I provisioned some > numbers from bandwidth.com to come into the system. > > When calling out from a bria professional softphone, as a > remote worker, I get disconnected at 25 seconds. I have a > similar experience calling in. > > I checked the bandwi

Re: [sipx-users] Hunt Groups and Max Forward

> > On Fri, 2009-05-01 at 12:14 -0400, Joly, Robert (CAR:9D30) wrote: > > > > That's interesting. I do all my debugging with tcpdump/wireshark > > > using the 'any' interface and that always gets me the intra-box > > > messages as well... > > Not the ones that go between services on the _other

Re: [sipx-users] sipXbridge 4.0 bandwidth.com template

dig -t srv _sip._udp.ot.bandwidth.com ; <<>> DiG 9.3.4 <<>> -t srv _sip._udp.ot.bandwidth.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 13786 ;; flags: qr rd ra; QUERY: 1, ANSWER: 2, AUTHORITY: 3, ADDITIONAL: 0 ;; QUESTION SECTION: ;_sip._udp.

Re: [sipx-users] Remote Worker Questions

> In testing, I am having a hard time getting NAT Traversal to > function the way I assumed it would for remote workers. > > I assumed if the remote worker had a personal firewall/router > it would traverse and work, but so far have not found this to > be the case. This is the intent and I'm s

Re: [sipx-users] Remote worker DID not working unless local

Tony, This is intriguing. I would need a snapshot of your system (or perhaps a tcpdump of the 'any' interface on sipx would suffice) to really troubleshoot this problem. bob > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfound

Re: [sipx-users] Remote Worker Call Quality

> > > There are two pieces that haven't been fully discussed here, the > > > firewalls > > at > > > either end. Just a suggestion that if you have a way in either > > > firewall > > (the > > > one where sipx sits behind, or the one where the remote > worker sits > > behind) > > > to prioritize t

Re: [sipx-users] Remote Worker Call Quality

> > There are two pieces that haven't been fully discussed here, the > > firewalls > at > > either end. Just a suggestion that if you have a way in either > > firewall > (the > > one where sipx sits behind, or the one where the remote worker sits > behind) > > to prioritize traffic over other tra

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