Re: [sipx-users] Mobile or WEP interface?

ins > *Sent:* Tuesday, November 08, 2011 7:02 AM > > *To:* Discussion list for users of sipXecs software > *Subject:* Re: [sipx-users] Mobile or WEP interface? > > ** ** > > I just updates, I'll check it out when I get to the office. > > ** ** > > Rene Pa

Re: [sipx-users] Mobile or WEP interface?

eeping old value if no new is sent in the request... I have uploaded an update to android market. Would you please give it a try again? Regards René 2011/11/8 Rene Pankratz > Hi Nathaniel, > i never have recognized something like that. Please let me know if you can > reproduce this.

Re: [sipx-users] Mobile or WEP interface?

are >> Subject: Re: [sipx-users] Mobile or WEP interface? >> >> Hello Rene, >> >> It would be great if you would upload your app to the android market, i >> am very much interested in your app! >> >> Thanks a lot! >> >> Thomas >&

Re: [sipx-users] Mobile or WEP interface?

anks a lot! > > Thomas > > > Original-Nachricht ---- > > Datum: Sun, 11 Sep 2011 16:07:35 +0200 > > Von: Rene Pankratz > > An: Discussion list for users of sipXecs software > > > > Betreff: Re: [sipx-users] Mobile or WEP interface? > > >

Re: [sipx-users] Mobile or WEP interface?

Hi there, last year I have created a simple android app that lets you get and set up call forwardings in an easy-to-use interface. I may upload an old (beta-)revision to android market if someone is really interested on it. But as already said in this thread: The SOAP interface does not offer muc

Re: [sipx-users] polycom ring

I think it should be set here: SipX webinterface -> Devices -> Polycom Phone -> Sound Effects -> (show advanced...) -> External call ringtone You should be able to create a phone group if you want to change it for many phones. Please let me know if that did the trick. Regards

Re: [sipx-users] polycom ring

I thing it should be set here: Polycom Phone settings -> Sound Effects -> (show advanced...) -> External call ringtone Please let me know if that did the trick. Regards 2011/4/28 Douglas Hubler > On Thu, Apr 28, 2011 at 9:34 AM, Ben Goodfellow > wrote: > > I really like the new internal/exte

Re: [sipx-users] speaker for paging

I don't know the valcom speakers. But Snom PA1 should also work fine for you. http://www.snom.com/en/products/sip-paging/snom-pa1/ Regards 2011/4/27 Tony Graziano > Use the vakcom paving server and a speaker (ceiling tile replacement). I > use this in a couplebof different environments withou

Re: [sipx-users] Enabling 16kHz G.722 in 4.4

Could this become choosable via sipxconfig? If there might be an issue it should be disabled by default, but I'd like to be able to switch 16kHz on with just a click :) Regards 2011/3/30 Michael Scheidell > On 3/30/11 4:30 PM, Matthew Kitchin (public/usenet) wrote: > > I believe this was you

Re: [sipx-users] Lookup name for incoming call

AFAIK there is no way to do this with sipx. But there are a lot of telephones available that are able to do an Phonebook (e.g. LDAP) lookup on incomming calls and show caller's name in display. Maybe this is an option for you? 2011/2/22 Alistair J. Fenning > Hi guys, > > > > As far as I can s

Re: [sipx-users] browser soft phone

There is a flash based solution available from Adobe: Take a look here: http://labs.adobe.com/technologies/flashmedia_gateway/ Regards 2011/2/12 m...@grounded.net > Anyone know of a browser based softphone which is either open source or

Re: [sipx-users] reload configuration file

checks for "field that needs > reboot" but I think we need to check with other phones. > > thanks > Domenico Chierico > > --- Ven 10/12/10, Rene Pankratz ha scritto: > > Da: Rene Pankratz > Oggetto: Re: [sipx-users] reload configuration file > A: "Discussi

Re: [sipx-users] Unmanaged Phones

Also make sure you entered the correct VoIP-Domain name in the phone's config (It is shown on SipX webinterface: System configuration -> Domain name). I had a phone that did not support DNS correctly so I had to enter IP Addresses in the phone's config. This lead to exactly the same problem you des

Re: [sipx-users] Does SipX normalize MoH?

for your quick answer, tony Regards 2011/2/1 Tony Graziano > > > On Tue, Feb 1, 2011 at 2:26 AM, Rene Pankratz > wrote: > >> Hello, >> one of our customers complains about MoH that is too loud. >> >> After uploading another file with less volume MoH and r

[sipx-users] Does SipX normalize MoH?

Hello, one of our customers complains about MoH that is too loud. After uploading another file with less volume MoH and restarting services, MoH seemed to be as loud as before. Does MoH server increase volume automatically? Regards René --

Re: [sipx-users] reload configuration file

Domenico, that sounds really interesting! I know that there are some settings that still require a reboot of the phone when they are changed. Does your patch handle that? How do the phones behave when changing such a setting and sending profile without rebooting? Regards René 2010/12/10 Domenic

Re: [sipx-users] Fwd: SOAP API help?

It might help to take a look here: http://wiki.voiceworks.pl/display/vwost/sipXworks+PHP+API Regards René 2010/11/26 Gmb > Yes George, > i'm sure, i'm connecting to sipx server :) and i can correctly access > from browser using > htt

Re: [sipx-users] Forwarding not working on Snom phone RESOLVED

Sorry I read your post too late. So I could not give you the answer you have found yourself in the meanwhile. If users use the webinterface of SipX for setting up call forwardings there is no need of setting call forwardings on the phone. Once there was a problem with SipX and snom phones that bro

Re: [sipx-users] wireless SIP phones - opinions

In general: +1 for the KIRK DECT System But if you are still looking for alternatives: There is also a Aastra VoIP-DECT system 2010/11/19 milosz > i have had an enormous amount of problems with the snom m3's. the > hardware is poor and the stack is abysmal. if you want an idea of > what you'l

Re: [sipx-users] We released our free ClickToDial Tool for SipX

m active window"? > > Thanks and regards, > Nikolay. > > > -- > *From:* sipx-users-boun...@list.sipfoundry.org [mailto: > sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Rene Pankratz > *Sent:* Thursday, November 11, 2010 1:24 PM > *To:* sipx-users &

[sipx-users] We released our free ClickToDial Tool for SipX

As already announced some time ago in another thread, we just published a free ClickToDial tool for SipXecs. With this tool you are able to dial out of the clipboard with a simple free configurable hotkey. It uses the SipX builtin click-to-dial mechanism and should work with any phone model. You

Re: [sipx-users] sipx as http server

Also worth to mention: It's not exactly the root of the webserver. You reach the folder there: http://SIPX-SERVER/phone/profile/docroot/ 2010/11/10 Douglas Hubler > On Wed, Nov 10, 2010 at 11:10 AM, Sven Evensen > wrote: > > We have a problem with the snom, for the version 320, we need an http

Re: [sipx-users] Not possible to change language on Snom phone

You should see the setting in the SipX Webinterface. If you select the phone you should find: Basic Settings -> web language And Basic Settings -> language Englisch should be default. Changing the language of Snom Phones may be tricky as the Languages are not included in the firmware. Which FW

Re: [sipx-users] Adding second route

Sorry I have been very busa at the end of last week. Yes indeed. That is what I meant. When you use this command: route add -net 172.16.30.0 netmask 255.255.255.0 gw 192.168.1.1 All Traffic sent by your sipx that is destined to the network 172.16.30.0 is sent over the gateway 192.168.1.1 all oth

Re: [sipx-users] Adding second route

Maybe you should tell us more about the network topology of the system. Why are there 2 Gateways in the Network of your sipx? Which nets are they connected to? If there is only one specific IP Network connected to the 192.168.1.1 then you can add a specific route only for this network and the def

Re: [sipx-users] yet another change to builds

ake it public. Currently at the doctor > because of a caraccident my wife had so slightly off work. > > Rene Pankratz schrieb: > > >Michal, > >I tried searching that page but couldn't find it in the wiki. > >Can you please provide us a link to that page?

Re: [sipx-users] tiny vpn device for remote workers

Some Snom phones (e.g. snom370) have a built in option for directly connecting through vpn. Maybe this is worth to look at. If this is working for you workers would only need the phone and no separate hardware. Regards 2010/10/28 m...@grounded.net > > With an "appliance" you would need IPSEC s

Re: [sipx-users] yet another change to builds

Michal, I tried searching that page but couldn't find it in the wiki. Can you please provide us a link to that page? Regards 2010/10/28 Michal Bielicki > We can add that but EDE is more or less EOL and most developers most hated. > There is a page on how to do it all very simply by installing t

Re: [sipx-users] t.38 testers please...

Sorry it is too early on saturday morning... I obviously I did not send FAX to a Patton GW but to a callweaver (asterisk) with T.38 enabled in the pcap. But as I said the reinvite was sent by the callweaver and not by faxback plugin. 2010/10/23 Rene Pankratz > Tony, > as you are

Re: [sipx-users] snom 370 and mwi

It can be set via sipxecs config: Select the phone -> Other settings -> show advanced settings -> Message LED other When this is unchecked the phon should use the LED for MWI only. Regards 2010/10/18 Norman Branitsky > On 10-10-17 11:01 AM, Roman Gelfand wrote: > > Never mind. It turns out

Re: [sipx-users] request for wiki edit rights: userid: herrold

May I jump on this thread with the same request? :-) userid: "pankratz" 2010/10/17 R P Herrold > > wiki / JIRA userid is: herrold > > thanks > > -- Russ herrold > ___ > sipx-users mailing list > sipx-users@list.sipfoundry.org > List Archive: http://li

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] 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] t.38 testers please...

No, this worked out of the box for me. 2010/10/12 Tony Graziano > Was there a trick to getting it to register directly to sipx? > > > On Tue, Oct 12, 2010 at 7:07 AM, Rene Pankratz > wrote: > >> Yes, this plugin registers as a SIP user directly at SipX. >> >

Re: [sipx-users] t.38 testers please...

Yes, this plugin registers as a SIP user directly at SipX. Be careful. I don't know if they updated the version since my install. I had to delete "Welcome.tif" in the installation folder as administrator under windows 7. Before doing that the Plugin never stopped sending me a "Welcome to Faxback"

Re: [sipx-users] t.38 testers please...

http://www.faxback.com/ I use theirt free T.38 Fax plugin for windows. It's really nice as it installs a FAX Printer driver... It works fine with our patton BRI Gateways. René 2010/10/11 Tony Graziano > Does anyone have a FXS adapter set for t.38 sitting behind sipx? If so, a > click to call s

Re: [sipx-users] SNOM 370 Problem Registering with sipx 4.2.1

/10/2010 09:22:08:Will try to reregister in 300 seconds > [2] 8/10/2010 09:27:08:Transport Error: Pending packet 104: generating > fake > [2] 8/10/2010 09:27:08:Registrar 2...@sipcomm.com refused with code 404 > [5] 8/10/2010 09:27:08:Will try to reregister in 300 seconds > [2]

Re: [sipx-users] SNOM 370 Configuration

We have several Snom's working with SipX (Best results with FW 7.3.14). René 2010/10/10 Roman Gelfand > Would anyone have a snom 370 configuration that works with 4.2.1? > Please, specify firmware viersion. > > > Thanks in advance > ___ > sipx-users m

Re: [sipx-users] Is there a response code that keeps SipX from trying the next gateway in the list in dialplan?

Thanks a lot for your answer... though it is somehow nonsatisfying... René 2010/10/7 Worley, Dale R (Dale) > > From: sipx-users-boun...@list.sipfoundry.org [ > sipx-users-boun...@list.sipfoundry.org] On Behalf Of Rene Pankratz [ > ren

[sipx-users] Is there a response code that keeps SipX from trying the next gateway in the list in dialplan?

Hello, I have a dialplan rule with several gateways associated (a VoIP provider and a patton ISDN gateway). Now, if I call someone who desn't answer or is busy I see that sipx calls the callee twice. Even if the VoIP provider sends a 486 to sipx, sipx switches to the patton gateway for a second t

Re: [sipx-users] SNOM 370 Problem Registering with sipx 4.2.1

I also thought about creating a feature request in the tracker. It would be really helpful if this option would be set within the preconfigured dhcp server and I cannot see a problem with other devices in the network. René 2010/10/7 Douglas Hubler > On Thu, Oct 7, 2010 at 12:34 AM, R

Re: [sipx-users] SNOM 370 Problem Registering with sipx 4.2.1

Unfortunately Snom telephones do not run out of the box with sipx. You have to edit /etc/dhcpd.conf file and add the following option: option bootfile-name "phone/profile/docroot/{mac}.xml"; After doing that add you can simply add Snom370 telephones in the sipx config and they should get

Re: [sipx-users] iptables

Mike, this commands should do what you want: # Use this to clear all iptables rules and allow everything: # iptables --flush # allow 192.168.1.101 to 192.168.1.105 to access configserver iptables -A INPUT -p tcp --src 192.168.1.101 --dport 8443 -j ACCEPT iptables -A INPUT -p tcp --src 192.168.1.1

Re: [sipx-users] Does anyone use ACD in 4.2.1?

Yes, we also use ACD with 4.2.1. 2010/9/24 Jason Mitchell > Yes. I have 8 agents using 1 queue at the moment and it works great. The > only issue I have is when the agent logs in to their portal, the other agent > status don't show up. They can't tell who else is logged in. > > >>> "Matthew K

Re: [sipx-users] Non-route-able numbers

Wouldnt a Dial plan rule like: "1-9" (+ specific number of digits) catch all extensions of the specified length as long as there is no user for this extension? René 2010/9/23 Kyle Haefner > Thanks Tony, > > Is this something I should be doing in the gateway? I think > Audiocodes has a 404 A

Re: [sipx-users] Change sipxconfig menus.

Nearly the same question was about 30 minutes before on the list: http://list.sipfoundry.org/archive/sipx-users/msg29072.html 2010/9/23 Douglas Hubler > On Thu, Sep 23, 2010 at 11:19 AM, Sven Evensen > wrote: > > Is it possible to re

Re: [sipx-users] End user portal customization

If you want to go the way that tony described (build your own SipX) then this is really helpful: http://wiki.sipfoundry.org/display/xecsdev/How+to+create+a+simple+page+in+sipXconfig The page describes how to add

Re: [sipx-users] PHP install?

PHP works fine with the Centos ISO of SipX 4.2.1 Install php using yum: # yum install php php-xmlrpc (Maybe you need some more php packages for the Pawel Board?) In /etc/httpd/conf/httpd.conf and in /etc/httpd/conf.d/ssl.conf you must change the Listening ports for apache. After that apache wil

Re: [sipx-users] VOIP Panel...

Sorry for enthusing about VOP :-) Monitoring Park extensions is working fine. René 2010/9/21 Jim Canfield > On Mon, Sep 20, 2010 at 11:58 PM, Rene Pankratz < > rene.pankratz.l...@iant.de> wrote: > >> Jim, >> yes we Tested the current version of VOP and those featur

Re: [sipx-users] Ip address

Please tell us more about your network configuration. Does the SipX server also create the VPN tunnel with the static IP? SipX should only be used with one network interface enabled. So maybe you need to establish the tunnel with another device and connect your SipX server via LAN to the VPN. Ren

Re: [sipx-users] VOIP Panel...

Jim, yes we Tested the current version of VOP and those features are working fine with SipX 4.2.1. The only problem we see with VOP is that the CallerID is shown on the monitor but not on the phone that is ringing (You can use a telephone as "audiodevice" for VOP). This is really annoying when usi

Re: [sipx-users] Howto create users that are not reachable from outside with SipX?

round with branches and lost servers ;) René 2010/9/16 Douglas Hubler > On Thu, Sep 16, 2010 at 11:43 AM, Worley, Dale R (Dale) > wrote: > > > > From: sipx-users-boun...@list.sipfoundry.org [ > sipx-users-boun...@list.sipfoundry.org]

Re: [sipx-users] Howto create users that are not reachable from outside with SipX?

Hehe, okay. I will keep this in mind fo the next time. Unfortunately the users have already extensions assigned within the range of all those others extensions. So I have to block them within the inbound gateway dialplan. Thanks René 2010/9/16 Tony Graziano > > Content-Type: text/plain; > ch

[sipx-users] Howto create users that are not reachable from outside with SipX?

Hello, I have a requierement for some users that shall not be able to call outside (no problem) and shall not be reachable from outside (problem?). One way I can reach this is of course in the dialplan of the PSTN gateway. I can do this but this is not really friendly for administration. Is there

Re: [sipx-users] Trademark "SipXecs"? What do I need to know?

to trouble when using the name in context with our application. Though i think it is just a positive fact that there are tools developed for the usage with SipX. René 2010/9/8 Rene Pankratz > Hi, > as we want to publish a simple free ClickToDial tool for sipx on our > website I wonder if

Re: [sipx-users] Redirection / 302 Moved Temporarily

There is some information about that: http://sipx-wiki.calivia.com/index.php/How_to_configure_User_Call_Forwarding 2010/9/7 Worley, Dale R (Dale) > > From: sipx-users-boun...@list.sipfoundry.org [ > sipx-users-boun...@list.sipfoundry.org] On Behalf Of Jea

[sipx-users] Trademark "SipXecs"? What do I need to know?

Hi, as we want to publish a simple free ClickToDial tool for sipx on our website I wonder if I have to take care about a trademark or something? Are there some simple rules that I may follow? If there are some rules it would be a great idea to put them into the (developer-) wiki. Of course we wan

Re: [sipx-users] Click to Call - IE8 - Script?

could send keys to the active window and put it in the clipboard – > > seems like the clipboard is an extra step? > > > > At any rate – let us know when you are able to publish the app. > > > > Nathaniel > > > > From: sipx-users-boun...@list.sipfoundry.o

Re: [sipx-users] Click to Call - IE8 - Script?

> extra step? > > > > At any rate – let us know when you are able to publish the app. > > > > Nathaniel > > > > *From:* sipx-users-boun...@list.sipfoundry.org [mailto: > sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Rene Pankratz > *Sent:*

Re: [sipx-users] Click to Call - IE8 - Script?

We have created a simple tool for doing click to dial out of any application using the clipboard with sipx. The steps are: - Copy number to clipboard - press a hotkey - a small window opens where you can modify the number that shall be dialed - press okay and click to dial is started We planned to

Re: [sipx-users] Sometimes SipXmaster & SipXredundant (4.0.4) are not responding network requests until reboot

nding. Maybe I should add an Intel NIC and deactivate the realtec cards? (Just an idea that the kernel module for the realtec chipset does not work properly) René 2010/8/31 Rene Pankratz > > As I just had the problem again I could test that. > Even if I ping numeric I see this happen

Re: [sipx-users] Sometimes SipXmaster & SipXredundant (4.0.4) are not responding network requests until reboot

As I just had the problem again I could test that. Even if I ping numeric I see this happen: 64 bytes from 192.168.10.71: icmp_seq=1 ttl=64 time=0.077 ms 64 bytes from 192.168.10.71: icmp_seq=2 ttl=64 time=0.074 ms 64 bytes from 192.168.10.71: icmp_seq=3 ttl=64 time=0.076 ms 64 bytes from 192.168.

[sipx-users] Sometimes SipXmaster & SipXredundant (4.0.4) are not responding network requests until reboot

Hello, we have a SipX 4.0.4 System that sometimes gets into a state where Master & Slave do not answer to network requests. Or at least do not answer at once, but after a few seconds they answer to all reuqests at once. For example if I ping one of the machines i don't get an answer for about 10 se

Re: [sipx-users] Snom 320 autoconfigure only works after disabling PnP

Hi, there are several ways to provision Snom telephones. Take a look here: http://wiki.snom.com/Category:Auto_Provisioning It seems the Epygi IP PBX replies to the SUBSCRIBE broadcast a Snom phone sends on startup by default. If that request is answred the snom phone ignores the options supplied

Re: [sipx-users] Did the content type of the resource list server change from 4.0.4 to 4.2.1?

Opened a ticket and attached requested Traces: http://track.sipfoundry.org/browse/XX-8652 René 2010/8/16 Douglas Hubler > On Thu, Aug 12, 2010 at 1:28 AM, Rene Pankratz > wrote: > > Indeed polycom phones subscribe the list without an error. > > But the problem still persist

Re: [sipx-users] Did the content type of the resource list server change from 4.0.4 to 4.2.1?

Indeed polycom phones subscribe the list without an error. But the problem still persists with snom phones. Maybe I should switch to sipx-dev and discuss if I should re-open the ticket. René 2010/8/12 Rene Pankratz > I am using 4.2.1-018930 (clean iso install) > The description e

Re: [sipx-users] Did the content type of the resource list server change from 4.0.4 to 4.2.1?

am > running 4.0.4 and 4.2.0 through 4.2.1 without seeing this (polycom). What > version of sipxecs are you using? > > On Wed, Aug 11, 2010 at 3:23 PM, Rene Pankratz > wrote: > >> I figured out: >> http://track.sipfoundry.org/browse/XX-7885 >> Seems to me I am sufferin

Re: [sipx-users] sipXconfig on mobile device

Using a HTC Desire (Android 2.2) with the default browser I don't see a play button when looking for voicemails. But it seems to me that the layout is not displayed correctly as the text of "duration" is cut in the middle. It seems the width of the display is to small for showing the whole layout.

Re: [sipx-users] Did the content type of the resource list server change from 4.0.4 to 4.2.1?

I figured out: http://track.sipfoundry.org/browse/XX-7885 Seems to me I am suffering from this? I will debug sipxrls tomorrow and post a snapshot. René 2010/8/11 Rene Pankratz > Hello list members, > after updating my PBX and my snom telephones cannot subscribe resource > list

[sipx-users] Did the content type of the resource list server change from 4.0.4 to 4.2.1?

Hello list members, after updating my PBX and my snom telephones cannot subscribe resource lists anymore. My telephone tries to subscribe: >SUBSCRIBE sip:~~rl~f~...@voip.ikt-bs.de SIP/2.0 >[...] >Event: dialog >Accept: application/dialog-info+xml,multipart/related >Supported: eventlist >User-Agent

Re: [sipx-users] Voicemail does not work after update - nullpointer exception in SipxIVR after update to 4.2.1

id your update to 4.2.1, did you make sure VM worked, or that it > was simply running as a service without errors? > > On Tue, Aug 10, 2010 at 3:51 AM, Rene Pankratz > wrote: > >> Hello List members. >> >> I did the following steps in my update: updated my old SipX

Re: [sipx-users] Voicemail does not work after update - nullpointer exception in SipxIVR after update to 4.2.1

Picher > Clean install from ISO? > > On Tue, Aug 10, 2010 at 3:51 AM, Rene Pankratz > wrote: > >> Hello List members. >> >> I did the following steps in my update: updated my old SipX 4.0.4 to 4.2.1 >> and archived the Config & Vm via webinterface. >>

[sipx-users] Voicemail does not work after update - nullpointer exception in SipxIVR after update to 4.2.1

Hello List members. I did the following steps in my update: updated my old SipX 4.0.4 to 4.2.1 and archived the Config & Vm via webinterface. Then i did a clean install of Sipx 4.2.1-018930 and restored Config & VM. Unfortunately the Voicemail system does not work any more. I can login but when I

Re: [sipx-users] Asterisk as a gateway -> attended transfer does not work

it's an Asterisk problem (I've > experienced this before as well). FreeSWITCH has the same issue currently. > I'd report this to Digium but don't expect it to get fixed. > > Josh Patten > Assistant Network Administrator > Brazos County IT Dept. > (979) 36

Re: [sipx-users] Asterisk as a gateway -> attended transfer does not work

Hi, Adding asterisk was quite simple. The peer config can be seen in my initial post of the thread. I simply had to add a user in sipx (for giving asterisk the right of placing external calls over other gateways) and added asterisk as an unmanaged gateway. Dialplan configuration debends on your sy

[sipx-users] Asterisk as a gateway -> attended transfer does not work

Hello list members, we have successfully connected an asterisk as a gateway to our sipx installation. This gateway is only used for outbound calls and everything seems to be working fine. The only problem we figured out is the attended transfer While blind transfer works without any problems the a

Re: [sipx-users] Remove "New software package update found. For details click: here" message in Webinterface

Thank you! 2010/5/17 Tony Graziano > Disable the repo > > 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: 434.984.84

[sipx-users] Remove "New software package update found. For details click: here" message in Webinterface

Hello, is there a way to remove this info message? I already deactivated the alarm so that I do not get the E-Mail, but I also would like to remove the message in Webinterface. René ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive

Re: [sipx-users] Auth required for a call from unmanaged GW to unmanaged GW (with no perm. configured)?

ed without authentication. I will try to reproduce this on SipX 4.2. If I see there the same behaviour I will open a Ticket for this. René 2010/5/7 Scott Lawrence > On Fri, 2010-05-07 at 12:35 +0200, Rene Pankratz wrote: > > Hello, > > I have a PSTN GW and another unmanaged

Re: [sipx-users] Auth required for a call from unmanaged GW tounmanaged GW (with no perm. configured)?

y on two rules… > > > > But this is expected behavior….Research multiple dial plans matching > same dial string. Discussed over, and over, and over… > > > > > > > > *From:* sipx-users-boun...@list.sipfoundry.org [mailto: > sipx-users-boun...@list.sipfoundry.

Re: [sipx-users] Auth required for a call from unmanaged GW to unmanaged GW (with no perm. configured)?

cannot add an unmanaged GW twice to the same SipX config, right? So how to solve this problem? René 2010/5/7 Rene Pankratz > Hello, > I have a PSTN GW and another unmanaged GW connected to the SipX (4.0.4). > > Incoming calls from PSTN that match an extension (e.g exact match o

[sipx-users] Auth required for a call from unmanaged GW to unmanaged GW (with no perm. configured)?

Hello, I have a PSTN GW and another unmanaged GW connected to the SipX (4.0.4). Incoming calls from PSTN that match an extension (e.g exact match on ext "298") shall be routed to the other unmanaged GW. But unfortunately the SipX wants the PSTN GW to be authenticated and sends a 407. No permissio

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

> > Well, if there is enough demand, I can make the refresher choice > dynamic. The other possibility is to suppress Session timer altogether > and simply rely on periodic re-INVITE to check for liveness of the > session. > Is it possible to achieve one of these possibilities without needing anot

[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] Sipxecs and openvpn server on the same machine

You might switch from TUN to TAP interface bridged with the ETH0 of the SipX machine. This would result in a VPN tunnel directly into the LAN and VPN client will be able to connect to SipX with the "normal" IP address. But also all other devices in LAN will become available to VPN client... René

Re: [sipx-users] Longer lists options: Users

Back to topic: +1 2010/4/14 Tony Graziano > I don't prefer beer in my coffee. Something stronger and darker from a > barrel out of kentucky. > > On Wed, Apr 14, 2010 at 4:08 PM, m...@grounded.net > wrote: > > Must be something in my coffee. I think I need beer now. > > > > > > On Wed, 14 Apr 20

[sipx-users] Polycom KIRK 300 and special Features (Presence / PickUp)

Hallo list members, I know that there are already several posts about Polycom KIRK stations but I but those features are never mentioned. Is it possible to observe a KIRK handset's presence (for BLF) with another VoIP Telephone? http://www.polycom.com/global/documents/support/sales_marketing/prod

Re: [sipx-users] Need pointer to doc for enhancing auto config information

look here for compiling SipXecs (you can modify and recompile quite easy with the EDE). http://sipx-wiki.calivia.com/index.php/Express_Development_Environment_Setup As far as I can see you will find the code for grandstream autoconf in: sipXconfig\neoconf\src\org\sipfoundry\sipxconfig\phone\grands

[sipx-users] Direct call forward without ringing the called extension

Hello list members, I got a very simple question. I want to forward a call to an extension directly so that the called extension does not ring. I know I can reach this when activating DND on telephone and configuring a call forward in SipX user interface. But would'nt it make sense if I could conf

Re: [sipx-users] Is a user able to diable VM himself? - Feature request for this purpose.

urations to end users will lead to wanting users to be > able to remove themselves from the directory, etc. This may be a slippery > slope to start down. > > -Original Message- > From: sipx-users-boun...@list.sipfoundry.org > [mailto:sipx-users-boun...@list.sipfoundry.org] On Beha

[sipx-users] Is a user able to diable VM himself? - Feature request for this purpose.

Hello List members, I often get the Request of enabling/disabling VM for users. As far as I can see the only way to disable VM for a user is to remove the permission as administrator. Or did I miss another way to achieve this setting as a user? It would be a great feature to control VM within use

Re: [sipx-users] When server is added to fresh installation the creation of mappingrules.xml fails

-5723>DNS > server on master is not reloaded after a redundant proxy is added.). > > Thanks, > Sathya > > Rene Pankratz wrote: > > Sorry my fault when writing the steps. > Of course I add the second server first in Web UI and then enter the data > in setup screen on

Re: [sipx-users] When server is added to fresh installation the creation of mappingrules.xml fails

installed SipX as a Redundant system i wonder if anything has changed since 4.0.0 release. When I add the second server. Should I first create profiles for both servers or directly enter the data in Redundant-server-setup? 2009/12/9 Scott Lawrence > On Wed, 2009-12-09 at 13:36 +0100, Rene Pankratz wr

Re: [sipx-users] When server is added to fresh installation the creation of mappingrules.xml fails

René 2009/12/7 Scott Lawrence > On Mon, 2009-12-07 at 14:20 +0100, Rene Pankratz wrote: > > Ok, I solved the problem myself. > > > > After adding a 2 A-Records for the VoIP domain in DNS that point to > > the servers everything works fine. But I am wondering why s

Re: [sipx-users] When server is added to fresh installation the creation of mappingrules.xml fails

Ok, I solved the problem myself. After adding a 2 A-Records for the VoIP domain in DNS that point to the servers everything works fine. But I am wondering why sipx-dns did not create those entries René 2009/12/7 Rene Pankratz > Hello, > after installing from the current SipX 4.0.4

Re: [sipx-users] Get registration status of phones via SOAP/REST oranything else

This might be simple enough. I will have a try with the subscription. I figured out that I also might parse the /var/sipxdata/sipdb/registration.xml by a simple script if I want to check if any registration expires. Or would that be a bad idea? Thanks a lot René 2009/10/5 Robert Joly > > Hel

Re: [sipx-users] Recording Calls

As voice data normally does not flow through the PBX there is no way of recording calls. René 2009/10/5 James Johnson > Is it possible to record calls with sipxecs and if so how? > > .. > James > > > ___ > sipx-users mailing list sipx-users@list.sipfo

[sipx-users] Get registration status of phones via SOAP/REST or anything else

Hello list members, as we want to observe registration status of phones by an external application I just wanted to ask if there is a (simple) way to poll registration status of all phones on SipX Registrar. René ___ sipx-users mailing list sipx-users@li

Re: [sipx-users] ldap attributes pulldown menus?

I hope I understood you question the right way. SipX (4.0.1) does a query for the attributes of the user object you selected before. As we are using a self-created non-standard object type for users we can still select any attribute we defined for our object class in LDAP. René 2009/9/2 Jiann-Mi

Re: [sipx-users] Where to find the Pick-up Settings?

P Registrar > > At present my Linksys SPA942 see the *78 as the DND code so its not working > for me. > > Hope this helps, > > Christopher Coleman > > On Wed, Sep 2, 2009 at 4:02 AM, Rene Pankratz wrote: > >> Hello list members, >> I got a simple question: W

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