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] Enhanced 911

Thanks Mike for the comments. When Using CAMA trunks, you have a direct interface from Your PBX to the PSALI database, so the PBX passes data to PSALI and they get the direct dispatch information. In recent years, that was changed from dedicated CAMA trunks to trunks that you designate within

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

Does sipxBridge attempt to generate in-audio DTMF tones? If the outside ITSP claims it won't support RFC2833 RTP telephony event messages and the inside session (with the phone) has negotiated the RFC2833 inband event style DTMF what's supposed to happen? This morning outbound calls via voip.ms

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

This is a Global Technologies Associates, but do you really think any other firewall DOESN'T have issues? At least it failed closed. Granted, this was some years ago, but we first started looking at firewalls, we found that Counterpane (not Counterpath!) had done a head-to-head lab test of abou

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

So I know what firewalls to stay away from, what are you running? 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 Helpd

Re: [sipx-users] Canned Prompts

Check the archives. I think "Karen" (as in my name is Plankton and this is my computer wife "Karen"), is the talent, scott mentioned her here a few months ago. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Te

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

> Reboots?! Yikes. If you go back to the old version of x-lite do things > go back to normal? Yes, it does. The really bizarre part is that Counterpath says that both are based on the same codestack. It's possible that there's something that's correct in my X-Lite config that's wrong in my ey

Re: [sipx-users] Canned Prompts

Disclaimer: I'm running RPM's of 4.2 I built from SVN The standard prompts appear to be in /usr/share/www/doc/stdprompts . I think the voice talent is 'Callie' from www.gmvoices.com (this is what the freeswitch project uses and I think these prompts came from there, but I could be wrong; to my

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] Canned Prompts

4.2 is also supposed to have better dial by name functionality: http://wiki.sipfoundry.org/display/xecsuser/Detailed+sipXecs+Roadmap Tony Graziano wrote: 2010/2/11 Bryan Simmons I need to change the to say enter the first few digits and then p

Re: [sipx-users] Canned Prompts

2010/2/11 Bryan Simmons > I need to change the to say enter the first few digits and then press > pound callers are complaining it takes too long to wait for system to narrow > down the search. It does take about 6 seconds if they don’t press pound. > Where are the standard English canned prompt

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

> Did you really expect anyone to try to use that? Wasn't expecting anyone to interpret the packet... I was mostly checking to see if anyone could tell me whether or not what I described is expected behavior on sipXec's part (if sipXecs is doing the right thing, then there's nothing to debug!)

[sipx-users] Canned Prompts

I need to change the to say enter the first few digits and then press pound callers are complaining it takes too long to wait for system to narrow down the search. It does take about 6 seconds if they don't press pound. Where are the standard English canned prompts located and can I just rerecord o

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

On Thu, 2010-02-11 at 11:47 -0500, Burden, Mike wrote: > > 00 16 35 a2 f8 22 00 90 fb 0d c0 fc 08 00 45 00 ..5..".. ..E. > > 0010 05 5a 82 c8 00 00 3f 11 4a 86 c0 a8 09 01 c0 6f .Z?. J..o > > 0020 1f 2c 13 c4 dd bc 05 46 51 74 53 55 42 53 43 52 .,.F QtSUBSCR > >

[sipx-users] Weird eyeBeam / NAT issue

Good morning, 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 addr

Re: [sipx-users] voip.ms: tries to connect to random port

On 2/11/10 9:26 AM, Scott Lawrence wrote: Very likely they are attempting to do NAT compensation for you. Ask them to disable that. its disabled on their side. I think i'll just either upgrade the firmware, or put it in the DMZ behind a different type of firewall. -- Michael Scheidell,

Re: [sipx-users] voip.ms: tries to connect to random port

On Thu, 2010-02-11 at 09:13 -0500, Michael Scheidell wrote: > and one more thing: > voip.ms seems to be the only one to signal back on the 'actual source > port'. everyone else assumes you are behind a firewall and looks at > the actual registration string that contains the public ip and port! Ve

Re: [sipx-users] voip.ms: tries to connect to random port

and one more thing: voip.ms seems to be the only one to signal back on the 'actual source port'. everyone else assumes you are behind a firewall and looks at the actual registration string that contains the public ip and port! -- Michael Scheidell, CTO Phone: 561-999-5000, x 1259 > *| *SECNA

Re: [sipx-users] voip.ms: tries to connect to random port

On 2/11/10 9:08 AM, Tony Graziano wrote: We use pfsense for this. you slap a *BSD firewall on? thats a thought, we originally thought we were going to dual home the box, public ip for sipx, internal network for phones. That won't work :-) you have pfsense r

Re: [sipx-users] voip.ms: tries to connect to random port

We use pfsense for this. 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.myitde

Re: [sipx-users] voip.ms: tries to connect to random port

On 2/11/10 8:49 AM, Tony Graziano wrote: Is the keepalive is trying to reach you on 5080 from them, then it sounds like the voip.ms is setup properly. You must have the sonicwall sip ALG for turned off. Your sonicwall must perform "symmetric NAT". If the NAT was symmetric,

Re: [sipx-users] voip.ms: tries to connect to random port

On Thu, Feb 11, 2010 at 8:08 AM, Michael Scheidell wrote: > > On 2/11/10 6:40 AM, Scott Lawrence wrote: > > On Thu, 2010-02-11 at 01:29 -0500, Michael Scheidell wrote: > > > > If you are using userauth, then their ny sever is stupid. When > register runs on sipx , source port 12699, they bind to

Re: [sipx-users] voip.ms: tries to connect to random port

On 2/11/10 6:40 AM, Scott Lawrence wrote: On Thu, 2010-02-11 at 01:29 -0500, Michael Scheidell wrote: If you are using userauth, then their ny sever is stupid. When register runs on sipx , source port 12699, they bind to that port and try to send signaling there. But, ill try it. When I

Re: [sipx-users] voip.ms: tries to connect to random port

On Thu, 2010-02-11 at 01:29 -0500, Michael Scheidell wrote: > If you are using userauth, then their ny sever is stupid. When > register runs on sipx , source port 12699, they bind to that port and > try to send signaling there. But, ill try it. When I get into > office. What do you have betwee

Re: [sipx-users] voip.ms: tries to connect to random port

On Wed, 2010-02-10 at 23:09 -0500, Nathaniel Watkins wrote: > Sorry - I've been using the blackberry for some of those emails...and > I'm easily distracted :) add that to the fact that you're all top-posting and it's a wonder anyone understands anything :-( _

Re: [sipx-users] Access to voicemail interface from an outside DID?

create a user with no phone, forward to VM (101), assign a DID. On Wed, Feb 10, 2010 at 10:07 PM, mkitchin.pub...@gmail.com < mkitchin.pub...@gmail.com> wrote: > * works. > > On 2/10/2010 9:00 PM, Nathaniel Watkins wrote: > > I'm pretty sure you can press # or * when voicemail picks up. > > > > N

Re: [sipx-users] Change in Network Configuration

Well, the ports are all the same whether it is NATted or not... The only thing that should change should be your IP address (assuming you have an internet-resolvable domain as your SIP domain). Mike > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- >