Re: [sipx-users] GXV3140?

On 12/07/2010, at 9:28 PM, Douglas Hubler wrote: > On Mon, Jul 12, 2010 at 1:18 AM, Graeme Allen > wrote: >> Has the Grandstream GXV3140 been added to the configuration/ >> provisioning >> interface of SipX, if not, are there plans to add it? > > As an initial test, have you tried configuring

Re: [sipx-users] IM group not distributed or not displayed by Bria 3.0

FWIW - http://www.counterpath.com/counterpath-updates-bria-for-mac-and-windows.html Says it will be available "mid-july". Counterpath is sometimes "vague" at best. If this is truly a day or few days away, it might help

Re: [sipx-users] IM group not distributed or not displayed by Bria 3.0

I'd like to resurface the issues that were raised by Tony and Paul regarding Bria 3.0 support Original can be found here http://article.gmane.org/gmane.comp.telephony.pbx.sipfoundry.general/25507 Bria 3.1 is still not out, but it might help improving the support in 3.0 and help those still gett

Re: [sipx-users] Actual call transfer, should it work? (polycom -> patton)

I'm thinking this is really a procedural thing. I believe there might be a step on the fxs device to complete a transfer successfully. It might also be possible to alter the transfer code in the fxs device but mostly I just use the fxo devices as pstn gateways. Tony Gra

Re: [sipx-users] Actual call transfer, should it work? (polycom -> patton)

I've never done this with an fxs port. The queatuion I have is when you do a trnasger initiated from an fxs device, how do you know it is referred or blind? A trace would be helpful. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myi

[sipx-users] Actual call transfer, should it work? (polycom -> patton)

[see the bottom for versions] When I receive a call on a subscribed FXS line on a patton 4114 I can flash and make a 2nd call. The patton manual states that if I then hang up on the second call the first call should 'transfer' to the destination. I've been unable to make this work between FXS p

Re: [sipx-users] Polycom off subject

looks like that was a fun one to ferret out! I too never use the 550. Pretty much just 33x & 650's. I have little use for the 45x and 55x phones. On an interesting note, I was working on a config doc for Voice Operator Panel and sipXecs and noticed the software is compatible with the Polycom CX

Re: [sipx-users] Polycom off subject

That sucks. I never order 550's, at some point it WILL affect those. Nice of them to recognize it and replace it though! On Mon, Jul 12, 2010 at 7:12 PM, Todd Hodgen wrote: > Having issues with headsets on Polycom 550 phones? A new technical > bulletin says there is an issue with any 550’s man

[sipx-users] Polycom off subject

Having issues with headsets on Polycom 550 phones? A new technical bulletin says there is an issue with any 550's manufactured between Jan 22, 2010 and March 31, 2010 and EHS capable headsets - Plantronics, Jabra, etc. Can't be fixed with software! http://knowledgebase.polycom.com/kb/search.do?cm

Re: [sipx-users] Call transfers

Perhaps... My point is itsp signalling is different than gateway signalling. How the gateway interprets it is one thing, what is being sent is another. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony

Re: [sipx-users] Call transfers

If I am correct the ITSP should never see internal extensions calling internal extensions, which is essentially what happens when you conference someone internally with an outside caller. Like I said before, the scenario the original poster explained works whenever using an Audiocodes Mediant 1

Re: [sipx-users] Call transfers

Er... Could be itsp signalling. I wouldn't put all my eggs in sipxbridge as the issue... 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 F

Re: [sipx-users] Call transfers

Being that sipXbridge is the only difference I would bet so. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/12/2010 3:57 PM, WORLEY, Dale R (Dale) wrote: > > From: sipx-users-boun...@list.sipfoundry.org > [sipx-use

Re: [sipx-users] SIP RTCP Summary Report - collector and report engine

Why don't you turn that on and log it with the polcom productivity license? 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: 434.984.8

Re: [sipx-users] doublechecking Pfsense and sipx?

Then its a lan rule, very simple. 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://w

Re: [sipx-users] Call transfers

I'll work on getting the snapshots put together. I don't have time today, but should tomorrow. Jake Ballamis Technical Support Manager p. 801-566-TECH (8324) f. 801-208-9317 jballa...@alliancetechsolutions.com This e-mail is intended solely for the person or entity to which it is addressed and

Re: [sipx-users] Call transfers

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Josh Patten [jpat...@co.brazos.tx.us] Dale I think this is a sipXbridge specific request because this functionality exists when using Audiocodes mediant 100

Re: [sipx-users] doublechecking Pfsense and sipx?

On 7/12/10 4:54 PM, Tony Graziano wrote: SMTP traffic and things like that are easier to dweal with than you think. Example: I have multiple ISP's and multiple Ip addresses. I create a LAN rule to send all SMTP traffic (tcp port 25) out via a specific gateway. I have to send smtp out 40 diffe

Re: [sipx-users] doublechecking Pfsense and sipx?

SMTP traffic and things like that are easier to dweal with than you think. Example: I have multiple ISP's and multiple Ip addresses. I create a LAN rule to send all SMTP traffic (tcp port 25) out via a specific gateway. It aint so hard, but I always suggest getting started somewhere. When you cre

[sipx-users] SIP RTCP Summary Report - collector and report engine

Caveat; This is likely a bit off topic, and if there is a better forum or place for it please point the way. I'm looking for a tool that collects the SIP RTCP Summary Report (i.e. one record per call) for the purpose of storing VoIP Call QOS detail for all calls over a fairly long period of tim

Re: [sipx-users] Call transfers

GOOD. It is important to state that in relation to phones, because assuming is not productive. On Mon, Jul 12, 2010 at 4:37 PM, Jake Ballamis < jballa...@alliancetechsolutions.com> wrote: > For the record, Tony, I **am** using 3.1.3 RevC. > > > > > > *Jake Ballamis* > > *Technical Support Manage

Re: [sipx-users] doublechecking Pfsense and sipx?

Why put everything on the pfsense firewall if that's what you're thinking. I use pfsense for sipx remote workers and ITSP's, leaving the rest of my traffic on my main firewall. On the sipx boxes, I just point them to their gateway of which ever pfsense they reside on. So far, seems to work very

Re: [sipx-users] Call transfers

For the record, Tony, I *am* using 3.1.3 RevC. Jake Ballamis Technical Support Manager p. 801-566-TECH (8324) f. 801-208-9317 jballa...@alliancetechsolutions.com This e-mail is intended solely for the person or entity to which it is addressed and may co

Re: [sipx-users] doublechecking Pfsense and sipx?

On 7/12/10 1:54 PM, Tony Graziano wrote: you could just donwload the sample config from my blog though... looks like the 'port forwarding' might not work for a whole network of different types of hosts behind the pfsense. this below looks like it will take anything and forwar it out the WAN ip

Re: [sipx-users] Call transfers

Does anyone feel like asking what firmware he is using on the phones? Also, the thread should be titled "adhoc conferencing on polycom phones with siptrunk". I'm not sure I would describe this as a transfer at all. I would hope he is using 3.1.3RevC firmware and not 3.2 stuff, which is known to h

Re: [sipx-users] Call transfers

Dale I think this is a sipXbridge specific request because this functionality exists when using Audiocodes mediant 1000 gateways (the helpdesk where I work uses it all the time) Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/12/2010 2:00 PM, WORLEY, Dale

Re: [sipx-users] Call transfers

From: Jake Ballamis [jballa...@alliancetechsolutions.com] So the behavior is expected, then, from what you're saying. The end result I am looking for is being able to do a 'warm' transfer, in that there is a brief conference between the three callers, then

Re: [sipx-users] Call transfers

I would guess not. I would suggest entering a ticket for sipXbridge at http://track.sipfoundry.org to create a workaround for sipXbridge to be able to handle BYE in this manner. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/12/2010 1:57 PM, Jake Ballami

Re: [sipx-users] Call transfers

So the behavior is expected, then, from what you're saying. The end result I am looking for is being able to do a 'warm' transfer, in that there is a brief conference between the three callers, then the middle caller drops out. Is this feature not something that sipx can do presently, or am I jus

Re: [sipx-users] Call transfers

When using an Audiocodes Mediant 1000 the two remaining conference parties stay on the phone together. This probably has something to do with sipXbridge tearing down the call when a BYE is received from any of the parties. Josh Patten Assistant Network Administrator Brazos County IT Dept. (979

Re: [sipx-users] Call transfers

Josh, My bad, I should have included that. We are using a bandwidth.com SIP trunk. Jake Ballamis Technical Support Manager p. 801-566-TECH (8324) f. 801-208-9317 jballa...@alliancetechsolutions.com This e-mail is intended solely for the person or enti

Re: [sipx-users] Call transfers

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jake Ballamis [jballa...@alliancetechsolutions.com] I have been doing some testing of transfers this morning and am seeing some behavior that I didn’t expe

Re: [sipx-users] Call transfers

What gateway are you using? Josh Patten Assistant Network Administrator Brazos County IT Dept. (979) 361-4676 On 7/12/2010 1:29 PM, Jake Ballamis wrote: All,   I have been doing some testing of transfers this morning and am seeing some behavior that I didn’t expect to find.

[sipx-users] Call transfers

All, I have been doing some testing of transfers this morning and am seeing some behavior that I didn't expect to find. I'd like to run this by everyone to see if I'm just off my rocker, or if we might have an issue with our sipx installation. We are currently on 4.2 with Polycom 550/650 p

Re: [sipx-users] doublechecking Pfsense and sipx?

you could just donwload the sample config from my blog though... On Mon, Jul 12, 2010 at 1:53 PM, Tony Graziano wrote: > You simply configure the firewall so it gets out, then you add the NAT > rules. Before you put any nat rules in you make sure > > Firewall: NAT: Outbound > * > * > > Has the d

Re: [sipx-users] doublechecking Pfsense and sipx?

You simply configure the firewall so it gets out, then you add the NAT rules. Before you put any nat rules in you make sure Firewall: NAT: Outbound * * Has the default allow any out, you can safely delete any other rules there it creates by default, then edit the default rule to use STATIC PORT N

Re: [sipx-users] doublechecking Pfsense and sipx?

On 7/12/10 1:34 PM, Tony Graziano wrote: No 1:1 is needed. Just use manual AON nat, and translate the ports using NAT and make sure it creates the associated outbound rule. Whatever you do, you should "stick" to a single public IP address when running one sipx server, one public ip for the

Re: [sipx-users] doublechecking Pfsense and sipx?

No 1:1 is needed. Just use manual AON nat, and translate the ports using NAT and make sure it creates the associated outbound rule. Whatever you do, you should "stick" to a single public IP address when running one sipx server, it doesn't matter if it is an IF ALIAS or the WAN address, it matters

[sipx-users] doublechecking Pfsense and sipx?

to make sure you have static natting, do you use the 1:1 natting with vircutal ip's and reflective rules? or do you turn off reflective rules, and AON and manually nat inbound and outbound? (my current firewall isn't fast enough for our traffic, and pfsense looks like it has LOTS of capabiliti

Re: [sipx-users] sipXecs 4.2.1 behind one-to-one NAT

The critical thing is that the sipXecs host must see DNS records that map its SIP domain to its *internal* address. Hosts outside the NAT should see DNS records that map the SIP domain to the *external* address. This sort of DNS configuration is commonly called "split DNS". Dale _

Re: [sipx-users] New System Planning Assistance Needed

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <48914> Message-ID: Thank you very much. ___ sipx-users mailing list sipx-users@list.si

[sipx-users] Who screwed with JIRA?

OK seriously WTF?? I browse to http://track.sipfoundry.org and a whole slew of tickets, one of which was being actively worked on by my coworker, are gone. http://track.sipfoundry.org/browse/XX-8474 click that and you'll see what I mean. Why did this happen? -- Josh Patten Assistant Network

[sipx-users] Who screwed with JIRA?

OK seriously WTF?? I browse to http://track.sipfoundry.org and a whole slew of tickets, one of which was being actively worked on by my coworker, are gone. http://track.sipfoundry.org/browse/XX-8474 click that and you'll see what I mean. Why did this happen? -- Josh Patten Assistant Network

Re: [sipx-users] GXV3140?

On Mon, Jul 12, 2010 at 1:18 AM, Graeme Allen wrote: > Has the Grandstream GXV3140 been added to the configuration/provisioning > interface of SipX, if not, are there plans to add it? As an initial test, have you tried configuring it as a GXV3000? Can you (or anyone reading this) get specs on any

Re: [sipx-users] Sipxecs behind NAT and one way audio. Strange ACK desitnation when a call is via VPN

I never understood why you had to use a vpn cpnection at all other than you had a security requirement to do so. Milkfish should be unnecessary as well. If the remote router turns off SPI and any sip alg, the remote user should be functional. Why you keep adding more stuff between the remote user a

[sipx-users] Fwd: Using Botnets to do SIP scanning

Keep in mind that its not just botnets. Previous discussions on the snort and emerging threats security list discussed amazon's cloud network being used for this. For SIPx users, I think the security is pretty tight and will prevent any overt attacks based on sipvicious but they may be looking to

Re: [sipx-users] Is there anyway to disable/stop openfire on sipxecs, so i can use standalone installation?

I found it. I just need to disable it from roles. On Mon, 2010-07-12 at 10:51 +0300, an...@iguanait.com wrote: > Hi, > > Is there anyway to disable openfire on sipxecs, so i can use standalone > installation? > > We are using centos 5. > > Thanks in advanced! _

[sipx-users] Is there anyway to disable/stop openfire on sipxecs, so i can use standalone installation?

Hi, Is there anyway to disable openfire on sipxecs, so i can use standalone installation? We are using centos 5. Thanks in advanced! ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users

Re: [sipx-users] Sipxecs behind NAT and one way audio. Strange ACK desitnation when a call is via VPN

Hi, We found another better solution for users that should use openvpn. They will set on their routers additional sip proxy that will help everything to work as expected ( milkfish ). This way they will not use openvpn and we can use sipxecs directly on host machine with the public ip address. I