Re: [sipx-users] Under Attack

Nathaniel... I have a how to for this on my blog which will also help limit some other things. I think its the first article. http://blog.myitdepartment.net It talks about limit the number of connections (new, not established), for remote users, which is going to help in addition to the country

Re: [sipx-users] conference bridge transfer unsuccessful with voip.ms [SOLVED]

On 10/13/2010 08:17 PM, Joe Conway wrote: > On 10/13/2010 07:39 PM, Tony Graziano wrote: >> >> Make sure your voip.ms account is set for: >> >> NAT (Network Address Translation) [no] > > I tried that, and it doesn't make any difference. Internal extension > transfers work, confere

Re: [sipx-users] Faxing

I'm very late I know... But I fell asleep very early yesterday. I just tested Faxing but did not get any answer from the gateway on your side. René 2010/10/13 Tony Graziano > 'or > > 4...@voice.phoneutopia.com > > > On Wed, Oct 13, 2010 at 1:59 PM, Tony Graziano < > tgrazi...@myitdepartment.ne

Re: [sipx-users] Under Attack

On Wed 13.Oct.10 23:51, Nathaniel Watkins wrote: >Just checked my cdr records - saw this a few minutes ago. > >sip 0#442075005000 10/13/10 11:12 PM 00:00:00 Failed >sip 1442075005000 10/13/10 11:12 PM 00:00:00 Failed >sip 0001442075005000 10/13/10 11:12 PM 00:00:00 Failed >sip

[sipx-users] Under Attack

Just checked my cdr records - saw this a few minutes ago. sip 0#442075005000 10/13/10 11:12 PM 00:00:00 Failed sip 1442075005000 10/13/10 11:12 PM 00:00:00 Failed sip 0001442075005000 10/13/10 11:12 PM 00:00:00 Failed sip 0011#442075005000 10/13/10 11:12 PM 00:00:00 Failed

Re: [sipx-users] conference bridge transfer unsuccessful with voip.ms

On 10/13/2010 07:39 PM, Tony Graziano wrote: > > Make sure your voip.ms account is set for: > > NAT (Network Address Translation) [no] I tried that, and it doesn't make any difference. Internal extension transfers work, conference bridge no so much (the attendant says "please wa

Re: [sipx-users] conference bridge transfer unsuccessful with voip.ms

Make sure your voip.ms account is set for: NAT (Network Address Translation) yes no route never On Wed, Oct 13, 2010 at 10:36 PM, Joe Conway wrote: > Based on some re

Re: [sipx-users] Remote sipx registrations

On Wed, Oct 13, 2010 at 10:17 PM, R P Herrold wrote: > On Wed, 13 Oct 2010, Tony Graziano wrote: > > > Yes. You need DNS SRV records. > > > > _sip._udp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. > > _sip._tcp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. > > > > As a minimum

[sipx-users] conference bridge transfer unsuccessful with voip.ms

Based on some recent posts to this list, I decided to try out voip.ms. Overall the service seem to work well, and I like their pricing. However, for some reason when a call comes in to the auto attendant through this trunk, it can transfer to a regular extension, but when the conference bridge ext

[sipx-users] Remote sipx registrations

On Wed, 13 Oct 2010, Tony Graziano wrote: > Yes. You need DNS SRV records. > > _sip._udp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. > _sip._tcp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. > > As a minimum for sip. These need to be hosted at the domain > name server for your

Re: [sipx-users] Remote sipx registrations

Don't worry. I'll bounce back. 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.

Re: [sipx-users] Remote sipx registrations

man, you're getting too nice... must be the pain meds. On Wed, Oct 13, 2010 at 7:27 PM, Tony Graziano wrote: > Yes. You need DNS SRV records. > > _sip._udp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. > _sip._tcp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. > > As a minimum

Re: [sipx-users] Remote sipx registrations

Yes. You need DNS SRV records. _sip._udp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. _sip._tcp.SIPDOMAIN.TLD. IN SRV 1 0 5060 hostname.SIPDOMAIN.TLD. As a minimum for sip. These need to be hosted at the domain name server for your primary domain name (as a minimum). On Wed, Oct 13, 2

Re: [sipx-users] Remote sipx registrations

yes, see the wiki re: split dns On Wed, Oct 13, 2010 at 7:11 PM, Roman Gelfand wrote: > Do I need any special dns records, outside of A to get to the server, > to enable remote users to register? > > On Wed, Oct 13, 2010 at 6:53 PM, Tony Graziano > wrote: > > Port 5060 for remote users. tcp and

Re: [sipx-users] Digital Gateway opinions

Tough to go wrong with either the M1000 or the Patton 4960. I think the 4960 is infinitely more configurable though which is a double edge sword because it does make it more complex. Mike On Wed, Oct 13, 2010 at 2:32 PM, Josh M. Patten wrote: > I use Patton for FXO gateways, Audiocodes for ever

Re: [sipx-users] Remote sipx registrations

Do I need any special dns records, outside of A to get to the server, to enable remote users to register? On Wed, Oct 13, 2010 at 6:53 PM, Tony Graziano wrote: > Port 5060 for remote users. tcp and udp. This is what the remote users > register and send calls to sipx on. They need port 3-31000

Re: [sipx-users] Remote sipx registrations

Port 5060 for remote users. tcp and udp. This is what the remote users register and send calls to sipx on. They need port 3-31000 udp for media. Additionally if using siptrunking you will need port 5080 udp to receive calls from itsp's. There is currently no mechanism for adjusting the port, e

[sipx-users] Remote sipx registrations

When registering a device from wan with sipx server. 1. What port(s) has to be open on firewall in front of sipx server. 2. Where do you specify what port on sipx is listening for registration or what port is listening for registrations. Thanks in advance

[sipx-users] merging sipx-trace files?

Is there a way or tool to merge several sipx-trace files made with different tokens to become one file for Sipviewer to open? Regards, Don McIlvin "The information in this electronic mail message is the sender's confidential business and may be legally privileged. It is intended solel

[sipx-users] Hiding Error

Any way of hiding the following error, perhaps even using a redirect? Error 404 - Not Found. No context on this server matched or handled this request. Contexts known to this server are: * WebApplicationContext[/cmcprov,sipXconfig-cmcprov] * WebApplicationContext[/sipxconfig,sipXconfig-we

[sipx-users] Preparing for a 4.0.5 release

eZuce need to support a customer on 4.0.4 with the following fixes - fixes replication time-outs for system with great than 8000 users - fixes proxy core when DNS SRV becomes unavailable - upgrades FS with version that's included w/4.2.1 to fix issue where FS is consuming all the CPU I will drop t

Re: [sipx-users] Digital Gateway opinions

I use Patton for FXO gateways, Audiocodes for everything else. I've not had one lick of trouble out of my Mediant 1000. Audiocodes FXO gateways leave something to be desired though... From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Todd

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] Faxing

'or 4...@voice.phoneutopia.com On Wed, Oct 13, 2010 at 1:59 PM, Tony Graziano wrote: > ~~4...@voice.phoneutopia.com > > > > > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > Email: tgrazi...@myitdepartment.net > > LAN/Telephony/Security and

Re: [sipx-users] Digital Gateway opinions

patton stuff is remarkably configurable... i swear. and i use them in great qty... audiocodes are boat anchors around here... On Wed, Oct 13, 2010 at 2:17 PM, Philippe Laurent wrote: > Looking for opinions, and I know that folks here use a variety of makes > including the AudioCodes and Patton d

[sipx-users] Digital Gateway opinions

Looking for opinions, and I know that folks here use a variety of makes including the AudioCodes and Patton devices. Requirements: 1 T1 interface. Support: Required, 24/7 (a small shop with few people) Redundancy: I have the budget, but do I really need it? Ease of setup would be nice, but reliabi

Re: [sipx-users] Faxing

~~4...@voice.phoneutopia.com 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

Re: [sipx-users] Faxing

Tony, to which uri shall the FAX be sent to? I have a T.38 enabled system with that I should be able to send a fax directly over the internet. 2010/10/13 Tony Graziano > I dont know if that will help. my pri has t.38, no issues with that. > > i was trying to get a test trunk for a day that has

Re: [sipx-users] sipx with netgear WNR3500

That netgear is an abomination. You'll find buffalo and some others have some models you can put an opensource os on and are fully configurable. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security

Re: [sipx-users] Faxing

Ok. That might work well. What I'd want to try is just adding the trunk (not needed if just inbound anyway), and putting a teliax DID on it and having someone send a 10-20 page fax, or several, to see if they are succesful. Additionally, if your fxs supports t.38, and is behind sipx already, doing

Re: [sipx-users] sipx with netgear WNR3500

Er... DLS model = DSL modem Mike ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

Re: [sipx-users] sipx with netgear WNR3500

On Wed, 13 Oct 2010 06:12:45 -0400, Tony Graziano wrote: > if he is using a softhphone make sure it is set for using local address, > ice and stun truned off. make sure your sip alg and spi is off. In this case, a simple PAP2 ATA would fulfill the requirement. Putting it directly on his provider

Re: [sipx-users] Faxing

Hi Tony, I might be able to help. I know I've talked about Teliax in the past. They have a T.38 service via their wholesale service and they have set up a PAYG account for me on that. I could temporarily have them add your IP to the ACL for the account if it helps. I haven't tested sipX yet di

Re: [sipx-users] Faxing

I dont know if that will help. my pri has t.38, no issues with that. i was trying to get a test trunk for a day that has t.38 to test that, but I have some things on my schedule i cant avoid, so it will wait until next week. I wanted to see if someone can send me a sip uri only fax (no trunks inv

Re: [sipx-users] Faxing

Tony, let me know if you still need fax help, we can test what ever you need if we've got what is required. We use flowroute and voip.ms, one of them has T.38 if not both or we can go through the mediant. Mike ___ sipx-users mailing list sipx-users@l

Re: [sipx-users] sipx with netgear WNR3500

> well, there's a cost benefit equation...  if you blow 6 hrs trying to make > this work (and then it still doesn't) and a linksys wrt54g costs $70... > Fight the good fight if you'd like. > At some point Mr. Logical will concede to Mr. Slightly Deranged. Got it, that's what I thought you meant :)

Re: [sipx-users] SIP URI Dialing - Solved - And another question

those messages are normal. On Wed, Oct 13, 2010 at 10:06 AM, Scott Richesson < sriches...@cincinnatifan.com> wrote: > I've solved the problem with SIP URI dialing by switching the server > external IP discovery mechanism from STUN to a fixed IP address. I'm not > sure why this made a difference,

Re: [sipx-users] SIP URI Dialing - Solved - And another question

I've solved the problem with SIP URI dialing by switching the server external IP discovery mechanism from STUN to a fixed IP address. I'm not sure why this made a difference, because the media relay logs were showing the proper external IP before I made the change. Good enough. Also, I noti

Re: [sipx-users] SIP Trunking Registering

Tony, The patch for this is complete and should be available in 4.2 branch in a few days after it has been reviewed by Douglas. See http://track.sipfoundry.org/browse/XX-9012 Joegen On Wednesday, 13 October, 2010 06:47 PM, Tony Graziano wrote: > Content-Type: text/plain; >charset="utf-8

Re: [sipx-users] SIP Trunking Registering

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <53229> Message-ID: CONSIDER SUBSCRIBING TO THE SIPX-USERS LIST FOR A BROADER AUDIENCE... The log really does not show much. It's

Re: [sipx-users] Client behind Local NAT device

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <53228> Message-ID: If it is going through NAT it is NOT local. You need to make sure your NAT'ing router has its sip ALG turned of

Re: [sipx-users] sipx with netgear WNR3500

if he is using a softhphone make sure it is set for using local address, ice and stun truned off. make sure your sip alg and spi is off. If that doesnt work, trash it, or consider burning more time than its worth to load myopenrouter on it and get out of the whole netgear software. http://www.myo

Re: [sipx-users] sipx with netgear WNR3500

well, there's a cost benefit equation... if you blow 6 hrs trying to make this work (and then it still doesn't) and a linksys wrt54g costs $70... Fight the good fight if you'd like. At some point Mr. Logical will concede to Mr. Slightly Deranged. On Tue, Oct 12, 2010 at 2:04 PM, m...@grounded.n

[sipx-users] off-topic: Bria end-device experience

After some extensive testing and troubleshooting with Counterpath, including me sending them a Polycom CX300, I have to conclude that if you consider buying this device then please switch to the Plantronics P540. The devices are identical. The big difference being that AEC works on the Plantroni

Re: [sipx-users] Upgrade to 4.2.1 with separate XMPP server failed

I am asleep... My excuse is that kind of things happens when you use a cellphone to reply to messages. On Wed, Oct 13, 2010 at 4:40 AM, Paul Scheepens wrote: > Replied to the wrong thread.you need a bed in the server room as > well:o) > > Paul > > > Tony Graziano wrote > > > > Enum lookups

Re: [sipx-users] Redundant system - two locations, two groups of users, dedicated SIP trunk for each group of users

The only way I could see to do something like this would be to use a different dialing plan entry by using a different dialing prefix when the ITSP fails. Far from automatic but it would work. Mike On Tue, Oct 12, 2010 at 4:45 AM, Irena Dolovčak wrote: > Hi, > I'm testing SipX in HA environment

Re: [sipx-users] Upgrade to 4.2.1 with separate XMPP server failed

Replied to the wrong thread.you need a bed in the server room as well:o) Paul > Tony Graziano wrote > > Enum lookups can be done in sipregistrar now if desire, as well as > ISN dialing. > > On 10/12/10, Tony Graziano wrote: > > So you sleep in the server room and separate personal and w

Re: [sipx-users] Fw: Fw: Upgrade to 4.2.1 with separate XMPP server failed

Heavier sleeper then you, that's for sure :o) Paul > Tony Graziano wrote > > So you sleep in the server room and separate personal and work space with a > bike. Must be a heavy sleeper. > > Tony Graziano, Manager > Telephone: 434.984.8430 > Fax: 434.984.8431 > > E

Re: [sipx-users] Regarding Time out option in Bria softphone

You need to delete the line profile on the phone, push the profile from sipxconfig and log back into the phone to have it download and use the new profile. If you look in the newly generated profile for the phone and find the parameters have changed, this should be sufficient. On Wed, Oct 13, 201