[sipx-users] SipXecs and PfSense

Hi, I'm planning to deploy sipxecs and pfsense altogether. As seen below: sipx server <---cisco<---pfsense firewall | INTERNET | --->pfsense --->cisco --> sipx server We will deploy IPSEC GRE tunnel on both sites. My question is: What ports needs to be opened in order to allow sipx-to-sipx to

Re: [sipx-users] sipX 4.2 and VMware?

There is an option on an IBM that is sold through IBM dealers as well. It's been a while since I actively sold the SCS version as a Nortel Distributor. Others might now on the list. Your point is a good one though. If it creates business opportunities then someone may have an interest

Re: [sipx-users] Cisco and sipX 4.2

I KNEW one day I'd see a car analogy on this list :-P Todd Hodgen wrote: > BTW, you can buy a Tom Tom GPS and use it in your Lexus. But, I wouldn't > expect to be able to integrate it into the dash like the one Toyota sells, > and I would not even consider going into Toyota dealerships and

Re: [sipx-users] Cisco and sipX 4.2

I think you guys are missing a really big point here. A company developed the support for Polycom so they could sell it in their commercial offering. That is why it is there, they created it and donated it back to the open source product, so they would have a product to offer to their customer

Re: [sipx-users] Cisco and sipX 4.2

What you are calling Cisco bashing is the cold reality that many have found trying to get a phone developed for a proprietary system to work as a phone using open standards. We could say the same thing about some of the Nortel Phones that were designed specifically for a proprietary phone system a

Re: [sipx-users] sipX 4.2 and VMware?

Clouds drop rain on parades.. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano Sent: Friday, April 23, 2010 1:16 PM To: mkitchin.pub...@gmail.com Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] sipX 4.2

Re: [sipx-users] sipX 4.2 and VMware?

So is it not available for customer installation at all? If not, is there any reason to think any effort would be spent toward making it more compatible with virtualization? Sent via BlackBerry from T-Mobile -Original Message- From: "Todd Hodgen" Date: Fri, 23 Apr 2010 19:26:37 To: ; '

Re: [sipx-users] sipX 4.2 and VMware?

Avaya ships the SCS on a DEll Optiplex. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of mkitchin.pub...@gmail.com Sent: Friday, April 23, 2010 1:10 PM To: Tony Graziano Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] sipX 4.

Re: [sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm

You can ensure the ntp is being sent to the polycom as part of its config file via sipxconfig. I prefer to specify ntp by ip address. Make sure the ntp source is reachable and working wherever it is. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Em

Re: [sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm

I've had this issue with one of my phones. what we had to do was go admin and reset all 3 options to completely wipe the phone. Then after the 3rd reboot, it downloaded the bootrom, sip app, and synced successfully. not sure if you have the same issue tho we were working with some old h

Re: [sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm

Thanks for the tips. I do control by group (POLY501) so it's weird that some phones pick it up correctly, and others (in the same group) don't. I disabled dhcp on sipx since I have another dhcp already that is setting the NTP server. I need to check to see if I disabled ntp on sipx. Any thoughts

Re: [sipx-users] Cannot establish a call from polycom 650 to cisco 7970g

Hi Everyone, After working with sipX for almost a month.. now I only have 2 remaining issues. 1. CIsco phone cannot call polycom 650 2. I cannot make outgoing calls to PSTN using audiocodes mp-118. Just so you all know. I'm still stuck with this problem. I know this thread goes too long and woul

Re: [sipx-users] Cisco and sipX 4.2

Many of the folks here have suffered the Cisco/ lock-in and gotten burned by it. These days it makes less and less business sense to stick with proprietary platforms and protocols because you end up running the planned obsolescence hamster wheel.  With ANY proprietary platform everything is all

Re: [sipx-users] Directly Call an extension via Site-to-Site

Mike Picher was right. The wiki was wrong. On Fri, Apr 23, 2010 at 8:19 PM, Rhon wrote: > Hi Everyone, > > Tony is right, changing the Address field to domain name fixed it. > > Closing this thread. > > Many thanks to Tony! > > Rgds, > > Rhon > > On Sat, Apr 24, 2010 at 1:07 AM, Picher, Michael

Re: [sipx-users] Cisco and sipX 4.2

+1 for me. Best regards, Rhon On Sat, Apr 24, 2010 at 8:00 AM, gabriel wrote: > Nathan, I feel your pain, I got over it :) > > we should have a sipx-users-cisco list where we (the cisco users) can help > each other out without having to deal with the classic "trow it out buy > Policom" message

Re: [sipx-users] Directly Call an extension via Site-to-Site

Hi Everyone, Tony is right, changing the Address field to domain name fixed it. Closing this thread. Many thanks to Tony! Rgds, Rhon On Sat, Apr 24, 2010 at 1:07 AM, Picher, Michael wrote: > Ah, well, that never worked unless that is how the PBX is setup (not > using SRV records). > > >

Re: [sipx-users] Cisco and sipX 4.2

Nathan, I feel your pain, I got over it :) we should have a sipx-users-cisco list where we (the cisco users) can help each other out without having to deal with the classic "trow it out buy Policom" message ;) I think is pretty clear for everybody that policoms are "supported" and the cisco

Re: [sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm

There is a field in sipxconfig which you can control by group. If sipx was installed by iso, it has an ntp server and dhcp server. The phones will get that information pushed to them by dhcp. There is a "refresh" setting, how long the phone waits before rechecking and syncing time. I normally cha

Re: [sipx-users] Cisco and sipX 4.2

Cisco plays with SIP in a completely different space. Don't take it as bashing. Cisco was an early adopter and "pusher" of voip, albeit in a "hybrid" manner. As a result a lot of their offerings are really "analog" in most respects as it relates to signalling when they use MGCP. SIP is really an a

[sipx-users] Ploycom Phones - Time/Date is 12/31 4:00pm

Hi all: I have a problem with some polycom phones not picking up the correct date/time. I have a set of polycom 501s with half displaying the correct time and date, and half not showing the correct date/time (but with flashing 12/31 4:00pm instead). The phones have been reformatted so that means

Re: [sipx-users] Cisco and sipX 4.2

It's unfortunate that the Cisco bashing alone and not the lack of support or focus makes me want to stop using this product. In all honesty, it really doesn't even come close to Callmanager and Unity Connection as a replacement and I actually wanted to see it get there at one point. I guess I will

Re: [sipx-users] Custom Notifications

Question: If many of you already know or figured out how to get alternative notifications working, that must mean there is either more notes somewhere or that some of you are part of the dev team, or that you're just plain genius. I'm not having any luck getting this to work and short of begging

Re: [sipx-users] sipX 4.2 and VMware?

gimme a cloud version! On Fri, Apr 23, 2010 at 4:10 PM, wrote: > I was just throwing something out there for discussion. One of the major > components of Sipx appears to be working on becoming more compatible with > virtualization. Hopefully sipx as a whole can. I would think one of the > major

Re: [sipx-users] sipxbackup

S3 is an option for backup-manager, if you compile it locally, then run it from cron. [?] On Fri, Apr 23, 2010 at 3:35 PM, Tony Graziano wrote: > S3. > > > On Fri, Apr 23, 2010 at 3:31 PM, Gerald Drouillard < > gerryl...@drouillard.ca> wrote: > >> On 4/23/2010 2:59 PM, Tony Graziano wrote: >> >

Re: [sipx-users] sipX 4.2 and VMware?

I was just throwing something out there for discussion. One of the major components of Sipx appears to be working on becoming more compatible with virtualization. Hopefully sipx as a whole can. I would think one of the major questions would be how is avaya selling the paid version of the product

[sipx-users] anybody seen this error on 4.0.4 to 4.2.0 upgrade?

Trying to help out on an upgrade gone bad... Anybody seen this after an upgrade? HTTP ERROR: 404 /sipxconfig Not Found RequestURI=/sipxconfig Powered by Jetty:// I did get a backup from command line, reinstalled, I uninstalled sipX cleared the databas

Re: [sipx-users] sipX 4.2 and VMware?

Yes, but you cant get that to stay (assuming it works) unless you have your own build with it stated. Once you make one change, it sends the configs to freeswitch (sixconfig does) like any other service, and will overwrite this "-vm" flag. If it honestly does something, it should be requested as an

Re: [sipx-users] Voicemail Misc Issues

> > > Thanks Dave, > > I didnt see the mailto: bit in the wireshark trace, that may > have been added by one of our browsers. Should the > improvement be able to pick up the divert number even if > there is no dial tag? i.e Cisco seems to format the diversion > hear with the directory numbe

Re: [sipx-users] sipxbackup

S3. On Fri, Apr 23, 2010 at 3:31 PM, Gerald Drouillard wrote: > On 4/23/2010 2:59 PM, Tony Graziano wrote: > >> /usr/bin/sipx-backup -d /backups -c >> >> Yeah, it is there. FTP is a little different though. You would have to >> use a FTP script. Both of these jobs (backup, ftp it off, move it to

Re: [sipx-users] sipX 4.2 and VMware?

I found at last a mention of the startup flag for freeswitch: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/054412.html -vm is the actual startup flag for freeswitch. I haven't been able to find a lot of info on exactly what it does. Having the same letters as the abbreviation

Re: [sipx-users] sipxbackup

On 4/23/2010 2:59 PM, Tony Graziano wrote: > /usr/bin/sipx-backup -d /backups -c > > Yeah, it is there. FTP is a little different though. You would have to > use a FTP script. Both of these jobs (backup, ftp it off, move it to an > archive, then prune the local archive) could all be run from cron t

Re: [sipx-users] Voicemail Misc Issues

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <6369cb70bfd88942b9705ac1e639a33821fd9bb...@dc-us1mbex4.global.avaya.com> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45421> Message-ID: Thanks Dave, I didnt see the mailto:

Re: [sipx-users] sipxbackup

/usr/bin/sipx-backup -d /backups -c Yeah, it is there. FTP is a little different though. You would have to use a FTP script. Both of these jobs (backup, ftp it off, move it to an archive, then prune the local archive) could all be run from cron though I think. On Fri, Apr 23, 2010 at 2:45 PM,

Re: [sipx-users] sipX 4.2 and VMware?

There is a whitepaper I saw (hat showed a more elaborate config to get timing, etc. Right for a cisco voice implementation on vmware. Very detailed. If I find it I'll shout. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartme

Re: [sipx-users] sipX 4.2 and VMware?

I can't find any info at the moment on it, but I believe there is an option in freeswitch to configure it to use something besides the cpu clock on virtual machines. I'm not sure if that would be applicable or help here. I'm all for anything that can be done to make sipx work properly on VMWare

Re: [sipx-users] Voicemail Misc Issues

> > Hi Dave, > > I have managed to get a wireshark trace of the divert to > voicemail. The key bit of the decode are: > > From: "6667912" > mailto:6667...@10.203.105.50>;tag=d19c5205-82bd-44fc-88c4 > -bf5d3c52feb5-37461274 > > Diversion: "6667912" > mailto:6667...@10.203.105.50>;reason=unc

Re: [sipx-users] sipxbackup

On 4/23/2010 2:36 PM, Picher, Michael wrote: > Is there a way to run backup without the GUI? > sipx-backup -n -c -d /tmp -- Regards -- Gerald Drouillard Technology Architect Drouillard & Associates, Inc. http://www.Drouillard.biz __

Re: [sipx-users] sipxbackup

[r...@sipx]# cd /usr/bin [r...@sipx bin]#./sipx-backup --help Backup script invoked at Fri Apr 23 18:41:46 UTC 2010, INTERACTIVE=yes. Usage: ./sipx-backup parameters Backup Configuration and/or Voicemail into archive(s.) If neither -c nor -v are specified, then both Configuration and Voic

[sipx-users] sipxbackup

Is there a way to run backup without the GUI? ___ 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 IP PBX --

Re: [sipx-users] Polycom different Ringtones

The ringers can be something other than uLaw or aLaw? Looks like I'll be resampling my ringers to 16kHz today. Eric Varsanyi wrote: If you want to actually choose the ringtones rather than use the default polycom builtin tones you'll need to put the wav files into an unmanaged device files sec

Re: [sipx-users] Voicemail Misc Issues

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <6369cb70bfd88942b9705ac1e639a33821fd957...@dc-us1mbex4.global.avaya.com> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45407> Message-ID: Hi Dave, I have managed to get a wir

Re: [sipx-users] sipX 4.2 and VMware?

Ok, thanks for the feedback. We'll have to tread carefully. Ken From: Picher, Michael [mailto:mpic...@cmctechgroup.com] Sent: Friday, April 23, 2010 11:52 AM To: Francis Tinio; Ken Fulmer Cc: sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] sipX 4.2 and VMware? You'll have to do

Re: [sipx-users] Polycom different Ringtones

If you want to actually choose the ringtones rather than use the default polycom builtin tones you'll need to put the wav files into an unmanaged device files section and activate it, then manually hack up /etc/sipxpbx/polycom/mac-address.d/sip.cfg.vm and add a section something like this near

Re: [sipx-users] Custom Notifications

Most happy to once I can get it working but have yet to figure out what I'm missing. Clues, hints, more information, most welcome :). On Fri, 23 Apr 2010 12:53:03 -0400, Picher, Michael wrote: > I don't know that anybody has volunteered yet.  Thanks for stepping >  > forward :-) !! >  >> -O

Re: [sipx-users] Directly Call an extension via Site-to-Site

Ah, well, that never worked unless that is how the PBX is setup (not using SRV records). > -Original Message- > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] > Sent: Friday, April 23, 2010 1:00 PM > To: Picher, Michael; xmlsc...@gmail.com > Cc: sipx-users@list.sipfoundry.org >

Re: [sipx-users] Polycom 3.2.3, shared lines seem to be working now

I followed the general directions on the wiki, the only trick might be that you have remember to resend profiles and reboot the phones and restart sipxproxy and registrar (?). If you're still stuck I can send you various config pages (but I didn't have to do anything weird). -Eric On Apr 22,

Re: [sipx-users] Directly Call an extension via Site-to-Site

The wiki had said to use the hostname, which did not work. I altered the wiki accordingly. 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

Re: [sipx-users] Directly Call an extension via Site-to-Site

How is that any different than before? From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Friday, April 23, 2010 12:39 PM To: Scott Lawrence Cc: Picher, Michael; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Directly Call an extension via Site-to-Site Tracker issue

Re: [sipx-users] Custom Notifications

I don't know that anybody has volunteered yet. Thanks for stepping forward :-) !! > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of m...@grounded.net > Sent: Friday, April 23, 2010 12:39 PM > To: sipx-users

Re: [sipx-users] sipX 4.2 and VMware?

You'll have to do some testing with load. With not much of a load they usually work fine... I've built many on KVM and on VMWare. I think with esx 4 at a minimum you'll want to dedicate 1 or two processors just to that virtual host for best operation. If you are going to operate an instal

Re: [sipx-users] Directly Call an extension via Site-to-Site

Tracker issue closed. Wiki amended with proper instructions. http://wiki.sipfoundry.org/display/xecsuserV4r0/sipXecs+to+sipXecs+Calling On Fri, Apr 23, 2010 at 12:15 PM, Scott Lawrence wrote: > On Fri, 2010-04-23 at 10:52 -0400, Tony Graziano wrote: > > Sounds like it is not just me: > > > > >

Re: [sipx-users] Custom Notifications

Is there a chance that someone is going to write a little howto for this? :). ___ 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

Re: [sipx-users] sipX 4.2 and VMware?

we're running some installations in vmware esxi. no issues so far. On Apr 23, 2010, at 12:28 PM, Ken Fulmer wrote: > Are there any known issues with running sipX 4.2 on VMware? As I understand > it, there were problems with earlier releases and VMware. > > Thanks, > > Ken Fulmer > > __

[sipx-users] sipX 4.2 and VMware?

Are there any known issues with running sipX 4.2 on VMware? As I understand it, there were problems with earlier releases and VMware. Thanks, Ken Fulmer ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfo

Re: [sipx-users] Outgoing call problem

Thanks for the response. To fix my issue, I had to delete my user and phone accounts, re-create and resent to phones to solve the problem. I did not have any required permissions applied to dial plans except 900 and international calling. In this case, no changes needed to be made to the exis

Re: [sipx-users] calling through gateway and rr. DNS records

On Fri, 2010-04-23 at 08:52 -0700, Johan Reinalda wrote: > All, > Apologies for a lenghty post. > > Here is the Scenario: test-lab fresh install of two 4.2.0 servers in > HA mode, some phones, and a gateway. Intra SIP, and inbound dialing > from campus PBX works fine (via AudioCodes M1K). > Tryin

Re: [sipx-users] Directly Call an extension via Site-to-Site

On Fri, 2010-04-23 at 10:52 -0400, Tony Graziano wrote: > Sounds like it is not just me: > > > http://track.sipfoundry.org/browse/XX-8221 ... and you still have not attached the snapshots for both systems. ___ sipx-users mailing list sipx-users@list

[sipx-users] calling through gateway and rr. DNS records

All, Apologies for a lenghty post. Here is the Scenario: test-lab fresh install of two 4.2.0 servers in HA mode, some phones, and a gateway. Intra SIP, and inbound dialing from campus PBX works fine (via AudioCodes M1K). Trying outbound dialing, but getting the following : Phone sends INVITE,

Re: [sipx-users] Directly Call an extension via Site-to-Site

Rhon, change from hostname to sip domain name in the gateway address and restart services as prompted. Will you confirm your results here? On Fri, Apr 23, 2010 at 11:17 AM, Tony Graziano < tgrazi...@myitdepartment.net> wrote: > Right, but the description of the problem is identical to mine. Gatew

Re: [sipx-users] Remote Polycom phone update/configuration withouta VPN

It does work... yes it will be upgraded as well. > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Jermaine Pinder > Sent: Friday, April 23, 2010 11:22 AM > To: tgrazi...@myitdepartment.net > Cc: sipx-users@

Re: [sipx-users] Remote Polycom phone update/configuration without a VPN

I tried that but It seems to fail. What about the firmware, will that be upgraded to? -Original Message- From: "Tony Graziano" [tgrazi...@myitdepartment.net] Date: 04/23/2010 11:20 AM To: "Jermaine Pinder" CC: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Remote Polycom pho

Re: [sipx-users] Remote Polycom phone update/configuration without a VPN

Port 21, specify the public Ip of sipx and tell the phone to provision thusly (ftp, not tftp). Once it is provisioned, you can close the firewall port or make allowances to come from "specific ip addresses or networks", or depending upon your needs, leave it open. It does work. On Fri, Apr 23, 20

Re: [sipx-users] Directly Call an extension via Site-to-Site

Right, but the description of the problem is identical to mine. Gateway calls and is answered by AA instead of ringing user. Example: Prefix: 7 digits after: 3 Send "matched suffix" to opposite gateway. I can map a sip uri on the phone and call, but something has changed in 4.20 that keeps this

[sipx-users] Remote Polycom phone update/configuration without a VPN

Remote Polycom phone update/configuration without a VPN Question; has anyone done a setup that will automatically provision a remote Polycom phone with all the correct settings using FTP/TFTP? Here’s my concept: > Punch a hole in our firewall to allow tftp to the sipx server > When setting up a

Re: [sipx-users] Directly Call an extension via Site-to-Site

I don't think this is the same issue... Rhon has a VPN tunnel between sites... no NAT involved. From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Friday, April 23, 2010 10:53 AM To: Picher, Michael Cc: Rhon; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Directly Call

Re: [sipx-users] Directly Call an extension via Site-to-Site

Sounds like it is not just me: http://track.sipfoundry.org/browse/XX-8221 On Fri, Apr 23, 2010 at 10:27 AM, Picher, Michael wrote: > It can be. You just either have DNS messed up or your gateways > improperly defined. It’s one of those two things. > > > > Mike > > > > *From:* Rhon [mailto:c4r

Re: [sipx-users] Directly Call an extension via Site-to-Site

How are you able to actually call the remote AA. And did you define the remote gateway as the SIP DOMAIN NAME and not as the SIP pbx HOST NAME. You want to use the SIP DOMAIN NAME. Mike From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On B

Re: [sipx-users] Polycom different Ringtones

Yes it is. On each phone has multiple lines, in the phone config go into those lines on each of the phones and set the ring tone. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of cyrill.rei...@iscoord.com Sent: Friday, April 23, 2010

Re: [sipx-users] Directly Call an extension via Site-to-Site

It can be. You just either have DNS messed up or your gateways improperly defined. It's one of those two things. Mike From: Rhon [mailto:c4rdi...@gmail.com] Sent: Friday, April 23, 2010 8:47 AM To: Picher, Michael; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Directly Call an

Re: [sipx-users] Gateway recommendation

On Fri, Apr 23, 2010 at 5:40 AM, wrote: > Hi all, > > Because of ITSP compatibility issues, I'd like to use a gateway device to > connect via SIP to the ITSP. Therefore, I'll need a device that could be > gateway on one end (SipX-end) and B2BUA on the other side (ITSP-end). There > will be no mor

Re: [sipx-users] SipXbridge - "Refresher=uac" not accepted by SIP Trunk. Is there a way to change this parameter?

On Fri, Apr 23, 2010 at 7:48 AM, Rene Pankratz wrote: > Hello list members, > we are evaluating a VoIP provider that is used as SIP Trunk (www.qsc.de, the > product is named "IPFonie"). > Incoming calls are working without any problems. > > But when we are trying to place a call the INVITE sent by

[sipx-users] Polycom different Ringtones

Hello, We use sipXecs Version 4.0.4 in combination with Polycom IP 650 phones. Each phone are configured with multiple phone lines. For example pesonal number and office main number. Is it possible to have different ringtones for the different lines? Best regards Cyrill Reiser cyrill.rei...@is

Re: [sipx-users] Directly Call an extension via Site-to-Site

I followed the wiki and the Book Building Enterprise Ready Tel. System with SipXecs 4.0 on sipXecs to sipXecs calling. 1. I setup a Unmanaged gateway 2. Ensure DNS is working without problems. 3. Create a site-to-site dialplan on each sites. Dial 2 and 2 digits, send entire number to gateway on H

Re: [sipx-users] 4.2 Upgrade repos 5 change

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <1271952760.22195.68.ca...@localhost> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45366> Message-ID: Found MY mistake. I had created a backup file of sipxecs.repo as old.re

[sipx-users] limiting the number of registered users on SIPX

We are trying to limit the resources that SIPX uses. Is there a way to limit the total number of registered users on SIPX? or the total number of simultaneous calls? As far as I know, these things are doable on SCS500 through licensing. How about SIPX? any workarounds? Thanks _

Re: [sipx-users] IM presence, general questions

> When I call from user1 to user2 and they both have IM > accounts logged in and enabled, I can see user1 shows on the > phone to user two. I do not see any changes for user2. > > Is IM presence tied to phone events or simply the presence > server in sipx? The IM presence ties into the RLS ser

Re: [sipx-users] Cisco and sipX 4.2

On 4/23/10 6:22 AM, Picher, Michael wrote: "Cisco follows IETF standards for SIP" That right there is funny! Hahahaha Good one Nathan! they hired the same group from Microsoft that worked on the standards for their mail server and clients. -- Michael Scheidell, CTO Phone: 561-999-5000,

Re: [sipx-users] Directly Call an extension via Site-to-Site

On Fri, 2010-04-23 at 18:26 +0800, Rhon wrote: > Hi Tony, > > That's exactly what I did. > > As said we have site-to-site running already... it's just that people > have to call the AA first become connecting to the desired extension. > > What I'm looking at is to go directly to the desired ext

Re: [sipx-users] Directly Call an extension via Site-to-Site

Sorry just want to rephrase: "I have problem with establishing calls this way. I just prefer to call directly any extensions without going to the operator or dialing 200." It should be: "I have don't problem with establishing calls this way. I just prefer to call directly any extensions without go

Re: [sipx-users] Directly Call an extension via Site-to-Site

Hi Michael, Sorry for that. Both sites are connected via IPSEC GRE tunnel. To reach the remote site I dial 200 (i changed the default operator = 200), I will then dial the extension. I have problem with establishing calls this way. I just prefer to call directly any extensions without going to t

Re: [sipx-users] Directly Call an extension via Site-to-Site

How are you reaching the remote AA? You really aren't giving us much to go on. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rhon Sent: Friday, April 23, 2010 8:36 AM To: Tony Graziano; sipx-users@list.sipfoundry.org Subject: Re: [

Re: [sipx-users] Directly Call an extension via Site-to-Site

Hmn.. that's strange. I already have this options in placed but still can't call directly to extension without passing AA. Thank you for your patience. Rhon On Fri, Apr 23, 2010 at 6:33 PM, Tony Graziano wrote: > You dialplan needs to say, example: > > Prefix "3" and "2 digits, thesend the ent

[sipx-users] SipXbridge - "Refresher=uac" not accepted by SIP Trunk. Is there a way to change this parameter?

Hello list members, we are evaluating a VoIP provider that is used as SIP Trunk (www.qsc.de, the product is named "IPFonie"). Incoming calls are working without any problems. But when we are trying to place a call the INVITE sent by SipX contains the Session-expires header with the value "Session-

Re: [sipx-users] Gateway recommendation

The 5400 can but the 5200 isn't listed with that capability. > -Original Message- > From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] > Sent: Friday, April 23, 2010 7:20 AM > To: Picher, Michael; r.vanv...@raffel.nl; sipx- > us...@list.sipfoundry.org > Subject: Re: [sipx-users] G

Re: [sipx-users] Gateway recommendation

It mught be worth considering whether you need or want "transcoding" capabilities. The new patton sbc can transcode too. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Hel

Re: [sipx-users] Gateway recommendation

Well, Ingate Siparators are a known good solution for this. Would run around a couple thousand US$ with sip trunking options and remote user nat traversal options for the low-end 19 unit. I’ve also utilized Patton gateways for this functionality (SmartNode 45xx series has 2 ethernet interfa

Re: [sipx-users] Directly Call an extension via Site-to-Site

You dialplan needs to say, example: Prefix "3" and "2 digits, thesend the entire number On the "300-399" side, and vice versa. On 4/23/10, Rhon wrote: > Hi Tony, > > That's exactly what I did. > > As said we have site-to-site running already... it's just that people have > to call the AA first

Re: [sipx-users] Polycom 3.2.3, shared lines seem to be working now

This Wiki is your friend... http://wiki.sipfoundry.org/display/xecsuserV4r2/Bridged+Line+Appearance Works great. Mike > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Robert B > Sent: Thursday, April 22,

Re: [sipx-users] 4.2 Upgrade repos 5 change

There may be some centos dependencies needed to install for this upgrade I think. 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

Re: [sipx-users] Directly Call an extension via Site-to-Site

Hi Tony, That's exactly what I did. As said we have site-to-site running already... it's just that people have to call the AA first become connecting to the desired extension. What I'm looking at is to go directly to the desired extension without passing thru AA always. Rhon On Fri, Apr 23, 20

Re: [sipx-users] Directly Call an extension via Site-to-Site

You create a dial plan of type "site to site". You create an unmanaged gateway on each system with the hostname of the other system. You assign your dialplan for site to site calls to that gateway. Read the wiki. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.

Re: [sipx-users] Cisco and sipX 4.2

"Cisco follows IETF standards for SIP" That right there is funny! Hahahaha Good one Nathan! From: Nathan Nieblas [mailto:nathan.nieb...@sacatech.com] Sent: Thursday, April 22, 2010 3:25 PM To: Picher, Michael; Rhon; sipx-users@list.sipfoundry.org Subject: RE: [sipx-users] Cisco and sipX

Re: [sipx-users] 4.2 Upgrade repos 5 change

Make sure you only have the sipxecs.repo in that folder. Run a 'yum clean all'. The re-run your 'yum upgrade'. Mike > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users- > boun...@list.sipfoundry.org] On Behalf Of Scott Lawrence > Sent: Thursday, April

[sipx-users] Gateway recommendation

Hi all, Because of ITSP compatibility issues, I'd like to use a gateway device to connect via SIP to the ITSP. Therefore, I'll need a device that could be gateway on one end (SipX-end) and B2BUA on the other side (ITSP-end). There will be no more than 5 simultaneous calls (at most). Could any

[sipx-users] How to manage server side contacts

Hi, I am still trying to understand how to use "server side contacts" for Bria (provisioned via Sipx). The history: By default after upgrading from 4.0.4 to 4.2.0 the Bria softphone gets a complete list of all users, including superadmin (with side-effects, see below). After deleting the only

Re: [sipx-users] Cisco and sipX 4.2

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <038601cae291$cc9d32f0$65d798...@net> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <45345> Message-ID: I believe Cisco have stopped producing new SIP loads (at least for the 794