Re: [sipx-users] Vitelity SIP Registration Problems

On Thu, Mar 10, 2011 at 8:35 PM, Paul Herron wrote: > Tony is exactly right. > > > > If you log into the Vitelity user portal, under the Support tab>Generic SIP > subtab; Vitelity will tell you the proxy URL or IP address that you should > be using in SipX under Gateway>ITSP Account> ITSP Registr

Re: [sipx-users] Vitelity SIP Registration Problems

Tony is exactly right. If you log into the Vitelity user portal, under the Support tab>Generic SIP subtab; Vitelity will tell you the proxy URL or IP address that you should be using in SipX under Gateway>ITSP Account> ITSP Registrar Address. What I am also saying is that IF you have a p

Re: [sipx-users] Voicemail IVR Voice Stutters During Playback

The wiki states: http://wiki.sipfoundry.org/display/sipXecs/Managing+Device+Firmware Activating Device Files You can activate uploads from either the upload edit page or the manage uploads page. You can only activate one type of upload at a time and once activated, you cannot upload or delete fi

Re: [sipx-users] Voicemail IVR Voice Stutters During Playback

I figured out how to get the application and BootROM changed on the Polycom phone. First you have to deactivate the device files. Then you have to make the changes. Then you have to activate the device files. Then send the profiles to the phones. It even rolled back the BootROM to 4.2.2. Hooray

Re: [sipx-users] Voicemail IVR Voice Stutters During Playback

Tony, the system is a testing system in service for my home office with only a few phones. I've tried rolling the firmware back, but it won't download and change the version. When I delete the newer device file and then load the 3.1.3C file and send the profile to the phone or reboot it, the

Re: [sipx-users] Vitelity SIP Registration Problems

He is stating the sipxbridge approach is to have trunks register to the carrier and have inbound invites sent to the port (5080). Port 5060 (inbound invites) is used for remote users (registration/call signalling). He is also saying (I think) he removed dns lookup to register to vitelity, and use

Re: [sipx-users] Vitelity SIP Registration Problems

Paul Herron wrote: I encountered this problem with Vitelity on two occasions:   1)  When using register, occasionally there would be a timeout.  I found that this was often a timeout on the DNS lookup, not a problem on Vitelity’s end so; I changed to the static ip

Re: [sipx-users] Patton FXS gateway - how to register lines with sipxecs?

Er, no. #1 - simply set it up as a user in sipx. No gateway needed (its an unmanaged phone, but no phone device needs to be created in sipx). #2 - see the wiki for examples for fxs config for patton! Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Em

[sipx-users] Patton FXS gateway - how to register lines with sipxecs?

All I need some help setting up a Patton SmartNode 4114 (4 port fxs gateway) with sipXecs. As this is an FXS gateway, I'm assuming I need to set it up as an unmanaged phone (at least this is how the audiocodes fxs gateways work). So, assuming my logic is sound, I need to enter a username/passw

Re: [sipx-users] Ghost Calls

I was able to solve this via an asterisk media gateway by setting pedantic message examination on the peer. Sent from my iPhone On Mar 10, 2011, at 10:18 AM, Ly Tran wrote: > We have been running SipXecs for a little over a year now and had never > experienced this problem until recently. I

Re: [sipx-users] DIal By Name

I am not sure it matters. There is a JIRA on this, checking the tracker might give you more information. I'm sure it's also been discussed recently. http://track.sipfoundry.org/browse/XX-9346 On Thu, Mar 10, 2011 at 3:09 PM, Burden, Mike wrote: > Sorry for jumping in late, but are you sure tha

Re: [sipx-users] DIal By Name

Sorry for jumping in late, but are you sure that Gary is included in a group that has the "List in Directory" permission? -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Dan White Sent: Thursday, February 03,

Re: [sipx-users] Installation on CentOS4

I still plan to do a debian one, I dropped my solaris work though. Am 10.03.2011 um 17:46 schrieb Michael Scheidell: > On 3/10/11 11:27 AM, Michal Bielicki wrote: >> >> No >> good enough ? >> Centos 5 is required. It will not work on Centos 4 >> it will also not work on CP/M >> or on MSDOS >> >

Re: [sipx-users] Installation on CentOS4

On 3/10/11 11:27 AM, Michal Bielicki wrote: No good enough ? Centos 5 is required. It will not work on Centos 4 it will also not work on CP/M or on MSDOS my radio shake TRSDOS? Theos? (oasis?). but, I sure would like to see this on Freebsd! -- Michael Scheidell, CTO o: 561-999-5000 d: 561-9

Re: [sipx-users] Installation on CentOS4

I don't think you will find all the dependencies filled with centos4. I think you need centos 5.4 or later... Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Control Systems Helpdesk: Tel

Re: [sipx-users] Installation on CentOS4

No good enough ? Centos 5 is required. It will not work on Centos 4 it will also not work on CP/M or on MSDOS regards Michal Bielicki Am 10.03.2011 um 17:13 schrieb Gopalakrishnan A.N: > Hi, > > I have CentOS 4, and I am not able to find any repo for CentOS4 in the below > link, > http://down

[sipx-users] Ghost Calls

We have been running SipXecs for a little over a year now and had never experienced this problem until recently. It has happened twice now where an external cell phone is being called every 30 minutes with a caller id coming from our internal numbers. I found the following links what appears t

[sipx-users] Installation on CentOS4

Hi, I have CentOS 4, and I am not able to find any repo for CentOS4 in the below link, http://download.sipfoundry.org/pub/sipXecs/4.2.0/CentOS/ Is there any way I can install in CentOS4. Please advice. -- Thank you with regards, Gopalak

Re: [sipx-users] Tracing older calls

>>> On 3/10/2011 at 09:55 AM, in message <5e483bf5-a543-4910-b100-6ad6ca450...@thegilmores.net>, Jeff Gilmore wrote: > Hi all, > I often get reports of problems with a call that happened yesterday or > the > day before. Generally, this means that it was logged in one of the > less-tha

[sipx-users] Tracing older calls

Hi all, I often get reports of problems with a call that happened yesterday or the day before. Generally, this means that it was logged in one of the less-than-current versions of the log files (such as sipXproxy.log.1 or sipXproxy.2.gz). I have never figured out how to generate a trac

[sipx-users] Vitelity SIP Registration Problems

I encountered this problem with Vitelity on two occasions: 1) When using register, occasionally there would be a timeout. I found that this was often a timeout on the DNS lookup, not a problem on Vitelity’s end so; I changed to the static ip address of the registration server (though this

Re: [sipx-users] Get ongoing calls using callresolver soap service

On 3/10/11 8:00 AM, Michael Picher wrote: Mircea, To me it makes sense to authenticate/secure it... It might not be valuable information to some. But if I had a competing company and wanted to write something to know who a competitor might be talking to this might be quite valuable informat

Re: [sipx-users] Get ongoing calls using callresolver soap service

On Thu, Mar 10, 2011 at 3:00 PM, Michael Picher wrote: > Mircea, > > To me it makes sense to authenticate/secure it... It might not be valuable > information to some. But if I had a competing company and wanted to write > something to know who a competitor might be talking to this might be quit

Re: [sipx-users] Get ongoing calls using callresolver soap service

Mircea, To me it makes sense to authenticate/secure it... It might not be valuable information to some. But if I had a competing company and wanted to write something to know who a competitor might be talking to this might be quite valuable information. Mike On Thu, Mar 10, 2011 at 7:18 AM, Mi

Re: [sipx-users] Get ongoing calls using callresolver soap service

On Thu, Mar 10, 2011 at 1:57 PM, Michael Scheidell < michael.scheid...@secnap.com> wrote: > On 3/9/11 5:48 PM, Mircea Carasel wrote: > > This web server is not secured in any way, and basically anyone can write a > SOAP client and get ongoing calls: > http://wiki.sipfoundry.org/display/sipXecs/Co

Re: [sipx-users] Get ongoing calls using callresolver soap service

On Thu, Mar 10, 2011 at 6:57 AM, Michael Scheidell < michael.scheid...@secnap.com> wrote: > On 3/9/11 5:48 PM, Mircea Carasel wrote: > > This web server is not secured in any way, and basically anyone can write a > SOAP client and get ongoing calls: > http://wiki.sipfoundry.org/display/sipXecs >

Re: [sipx-users] Get ongoing calls using callresolver soap service

On 3/9/11 5:48 PM, Mircea Carasel wrote: This web server is not secured in any way, and basically anyone can write a SOAP client and get ongoing calls: http://wiki.sipfoundry.org/display/sipXecs /Configuration+SOAP+API I was thinking and wondering if this is a security issue; I know that for

Re: [sipx-users] Internal SBC Statistics - Active Calls are always more than zero.

Either the firewall is preventing the BYE from being seen at sipx, or the ITSP is not cleanly disconnecting. Either way, sipx is not seeing the BYE message in the signalling and that means the CDR doesn't know the call has disconnected. Usually its a firewall configuration. What is the firewall ty

Re: [sipx-users] Voicemail IVR Voice Stutters During Playback

Your ram setting is a bit small. I run 2GB in a lab environment with 2-3 phones for testing, I wouldn't in production run less than 4GB (64 bit OS preferred). Your vlan for voice (assuming you have one) needs to be prioritized over your other vlans, etc. There is nothing wrong with the wav files.

Re: [sipx-users] Vitelity SIP Registration Problems

On Wed, Mar 9, 2011 at 10:49 PM, Tim Ingalls wrote: > I've been having problems when my sipXecs 4.2.1 platform sends SIP > registration attempts to Vitelity. Periodically throughout the day the > authentication request gets rejected multiple times. Then sipXecs waits to > re-register for 10 minute

Re: [sipx-users] Internal SBC Statistics - Active Calls are always more than zero.

"CDR -Active " list is empty after restarting CDR service. "Internal SBC Statistics - Active Calls" is zero after restarting SIP trunk service. It is obvious. Point is I can find 10 "active calls" after some weeks uptime and I have no idea how to grep that 10 call from 2 weeks logs. My firewall ma

Re: [sipx-users] IVR OK but no audio with confernce call?

Thanks, Updated to new build and can confirm that the issue is now resolved. //Staffan On 7 mar 2011, at 14.13, George Niculae wrote: > Staffan, > > I reverted the commit that introduced this problem (for fixing conf > consultative transfer), please try on the next available build > > Thank