Re: [sipx-users] Enable and configure conference bridge between SipX and DimDim Webconference Service

Dim Dim integration is already in the 4.2 release. It's not a real tight integration, more of a link between the two system. Enter your dim dim account information in your conference bridge is basically all you do. It creates links you can send to your conference invitee's, you click on your lin

[sipx-users] Enable and configure conference bridge between SipX and DimDim Webconference Service

Hi, first I would like to know if it is even possible (now as of SipX 4.2) to combine a DimDim WebConference with a conference held via local SipX Server. I'm planning on also bringing together an audio and a multimedia conference with just audio available to users of IP phones. So please te

[sipx-users] Reset superadmin password failed

Hi, I tried to reset superadmin password as the guide of "sipxconfig.sh --database reset-superadmin". Whereas, after restarted the sipx and logged in the web, it did not prompt me by a blank password change. What was any step I missed then ? Regards,

Re: [sipx-users] Broadvox audio after 30 minutes...

Tony, I'm using a SIP trunk through Broadvox with the default settings as shown below in your email and have had no reported issues with dropped audio after 30 minutes. However they DID do some sort of tweak on their end back when we set the trunk up because incoming calls would get killed if you

[sipx-users] Registering Endpoints

Hi, I have a User defined to register a softphone. I see a REGISTER come in. The register msg gets forwarded to the domain (in the FQDN) defined in the Server--> Domain for the SIPx Server. ___ sipx-users mailing list sipx-users@list.sipfoundry.org

Re: [sipx-users] OpenFire SSL problem

On Mon, Aug 9, 2010 at 8:20 PM, Tony Graziano wrote: > > Content-Type: text/plain; >  charset="utf-8" > Content-Transfer-Encoding: 8bit > Organization: SipXecs Forum > In-Reply-To: > X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50140> > Message-ID: > > > > Confused. Why would one need an ssl ce

Re: [sipx-users] Broadvox audio after 30 minutes...

Thanks. As I was reading this: (page 6) http://www.voicesonic.com/panasonic/manuals/Panasonic-Tech-Notes/Panasonic-Other-Tech-Notes/KX-TDExxx_NCPxxx_BroadVox_SIP_CO_Line_Configuration.pdf

Re: [sipx-users] Broadvox audio after 30 minutes...

defaults - empty SIP message On Aug 9, 2010, Tony Graziano wrote: Anyone have any guidance using broadvox (no template yet).Using ip/auth, dead silence after 30 minutes (hold/resume restores it).Anyone know what rtp keepalive they want?Tony Graziano, ManagerTelephone:

Re: [sipx-users] Advice

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50141> Message-ID: My gut tells me to suggest you look at a different type of platform (FreeSwitch). sipXecs really excels at a lo

Re: [sipx-users] OpenFire SSL problem

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50140> Message-ID: Confused. Why would one need an ssl certificate for openfire anyway? sipxecs has a method for importing a stand

Re: [sipx-users] Move from Asterisk to SipX - Two hold backs

For the call-out annoy for voicemail you could add a feature request. This is something a lot of voicemail servers do. For call recording he could look at Oreka (http://www.orecx.com/web/). It just needs to set in the call path (mirror the gateway port or the PBX if using SIP trunks). Mike On

Re: [sipx-users] looking to change FROM line in voicemail

Been through this some time ago, hence a WIKI page was created many moons ago... http://wiki.sipfoundry.org/display/xecsuserV4r2/Voicemail-Email+Custom+Notifications On Mon, Aug 9, 2010 at 5:00 PM, Michael Scheidell < michael.scheid...@secnap.com> wrote: > > what do I need to restart for this to

Re: [sipx-users] SIP Trunk Setup questions

Essentially yes. It uses dns extensively, especially in HA mode, and dns must be configured correctly in order for it to work properly. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telephony/Security and Cont

Re: [sipx-users] SIP Trunk Setup questions

Thx for the help. Since I can change the listen port of asterisk , I just changed that to something else for now...to validate sipX. So interprocess communication on sipX will use DNS SRV instead of what is configured through the GUI? -Original Message- From: WORLEY, Dale R (Dale) [m

[sipx-users] Broadvox audio after 30 minutes...

Anyone have any guidance using broadvox (no template yet). Using ip/auth, dead silence after 30 minutes (hold/resume restores it). Anyone know what rtp keepalive they want? Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartmen

Re: [sipx-users] SIP Trunk Setup questions

From: Ujjval Karihaloo [ujj...@simplesignal.com] I was using port 5050 for SIP Proxy as I am also running asterisk on this server...I changed it back to port 5060, shutdown asterisk and it worked... Looks like the sipXBridge only talks to the SIPXProxy pr

Re: [sipx-users] SIP Trunk Setup questions

Yes AND no. The initial invite comes in from your BW switch at port 5080, ongoing signalling takes place on port 5060 once a connection has been ACK'd, which was an IMPORTANT detail you left out having * running and using port 5050, etc. The two both were trying to respond for requests on port 5060

Re: [sipx-users] SIP Trunk Setup questions

FYI.got it working. I was using port 5050 for SIP Proxy as I am also running asterisk on this server...I changed it back to port 5060, shutdown asterisk and it worked... Looks like the sipXBridge only talks to the SIPXProxy process over port 5060is that correct? Any way to use other por

Re: [sipx-users] SIP Trunk Setup questions

From: Ujjval Karihaloo [ujj...@simplesignal.com] I am not getting any ANSWER SECTION back for that. I will make sure its hostname is resolvable through DNS and then try What are my options if the hostname of the sipX server is not a FQDN...that maps

Re: [sipx-users] Move from Asterisk to SipX - Two hold backs

Message was in forum only, default reply does not include text. Okay, Ready to make the switch from Asterisk to SipX, but have two hold backs. 1. VMail notify - Asterisk currently has a vm-notify feature where a vmailbox can be setup to dial a telephone number to alert you that a message was le

Re: [sipx-users] MOH not working

(also if I recall, snom phones do not lookup SRV records in a normal fashion, so make sure you have an A record in DNS that points your domain to sipx as well). On Mon, Aug 9, 2010 at 3:03 PM, Tony Graziano wrote: > You need to tell me what is in the field for the MOH uri in your phone > config f

Re: [sipx-users] looking to change FROM line in voicemail

what do I need to restart for this to take effect? -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Security Corporation * Certified SNORT Integrator * 2008-9 Hot Company Award Winner, World Executive Alliance * Five-Star Partner Program

Re: [sipx-users] looking to change FROM line in voicemail

On Mon, Aug 9, 2010 at 3:39 PM, Michael Scheidell < michael.scheid...@secnap.com> wrote: > On 8/9/10 3:11 PM, Josh M. Patten wrote: > > > http://github.com/dhubler/sipxecs/blob/master/sipXivr/src/main/java/org/sipfoundry/voicemail/EmailFormats.properties > > > > fisheye isn’t working right now. >

Re: [sipx-users] SIP Trunk Setup questions

I am not getting any ANSWER SECTION back for that. I will make sure its hostname is resolvable through DNS and then try What are my options if the hostname of the sipX server is not a FQDN...that maps to an IP address..? Thx. -Original Message- From: WORLEY, Dale R (Dale) [mail

Re: [sipx-users] looking to change FROM line in voicemail

On 8/9/10 3:11 PM, Josh M. Patten wrote: http://github.com/dhubler/sipxecs/blob/master/sipXivr/src/main/java/org/sipfoundry/voicemail/EmailFormats.properties fisheye isn't working right now. Thanks, got it. what do I need to restart for this to take effect? -- Michael Scheidell, CTO o: 561

Re: [sipx-users] Question about removing DNS Forwarder from Sipx

I was taking the secondary 8.8.8.8 forwarder out, because it was using it and couldn't resolve any local hostnames. The named.conf was where I found it and removed it from. It fixed the issue as soon as I restarted named. Danny -Original Message- From: WORLEY, Dale R (Dale) [mailto:dw

Re: [sipx-users] Question about removing DNS Forwarder from Sipx

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Danny Shay [ds...@norlemtc.com] What is the best method for reconfiguring DNS on a sipx server? I'm not certain wh

Re: [sipx-users] SIP Trunk Setup questions

From: sipx-users-boun...@list.sipfoundry.org [sipx-users-boun...@list.sipfoundry.org] On Behalf Of Ujjval Karihaloo [ujj...@simplesignal.com] I am getting following error for the SIP Proxy Service….I am not using DNS SRV for now. * SIP route to SIPXC

Re: [sipx-users] looking to change FROM line in voicemail

http://github.com/dhubler/sipxecs/blob/master/sipXivr/src/main/java/org/sipfoundry/voicemail/EmailFormats.properties fisheye isn't working right now. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell Sent: Monday, August 0

Re: [sipx-users] looking to change FROM line in voicemail

The requested resource cannot be found. Path "main/sipXivr/src/main/java/org/sipfoundry/voicemail/EmailFormats.properties" does not exist. and in /etc/sipxpbx/sipxivr I only see: ls -lt total 16 -rw-r--r-- 1 sipxchange sipxchange 262 Apr 14 20:49 PersonalAttendan

Re: [sipx-users] MOH not working

You need to tell me what is in the field for the MOH uri in your phone config file. On Mon, Aug 9, 2010 at 2:44 PM, Claudio Succa wrote: > Il lunedì 09 agosto 2010 17:43:39 Tony Graziano ha scritto: > > Does it work on internal calls (user 1, to user 2) or none at all? > > > > MOH support depends

Re: [sipx-users] MOH not working

Il lunedì 09 agosto 2010 17:43:39 Tony Graziano ha scritto: > Does it work on internal calls (user 1, to user 2) or none at all? > > MOH support depends on several things working properly. Tony, I just trace the output of the server with tcpdump. I call the phone B from the phone A then press ho

Re: [sipx-users] Move from Asterisk to SipX - Two hold backs

Is it possible to see the questions ? Am 09.08.2010 um 20:09 schrieb Tony Graziano: > > Content-Type: text/plain; > charset="utf-8" > Content-Transfer-Encoding: 8bit > Organization: SipXecs Forum > In-Reply-To: > X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50111> > Message-ID: > > > > 1.

Re: [sipx-users] Awesome - We received a check today

On Mon, Aug 9, 2010 at 12:12 PM, Martin Steinmann wrote: I wanted to thank Jim Canfield and Emstar for a great contribution to > SIPfoundry, that we received today. This is the kind of collaboration we > were hoping for and it is very encouraging and motivating to see the > community getting enga

Re: [sipx-users] Move from Asterisk to SipX - Two hold backs

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50111> Message-ID: 1. This is not available as a system option, but could be added if someone wants to develop it. sipXecs also ha

Re: [sipx-users] SIP Trunk Setup questions

IF you can't get the sipviewer working, how about a Wireshark trace of the call coming into your network, or into the server. You can look at the capture from the VOIP call settings, it's very similar to sipviewer. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.si

Re: [sipx-users] Awesome - We received a check today

No argument from me. On Mon, Aug 9, 2010 at 1:42 PM, Philippe Laurent wrote: > Bingo. It would be nice for those of us who purposefully specialize in the > smaller customers. It would really be nice to have the backing of the > mothership. > > On Mon, Aug 9, 2010 at 1:17 PM, Todd Hodgen wrote:

Re: [sipx-users] SIP Trunk Setup questions

Sorry but this does not work http://sipxecs.sipfoundry.org/temp/sipXecs/sipviewer-install.jar Any other place I can get this viewer from? From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, August 09, 2010 11:36 AM To: Ujjval Karihaloo Cc: Michael Scheidell; sipx-users@list.

Re: [sipx-users] Awesome - We received a check today

Bingo. It would be nice for those of us who purposefully specialize in the smaller customers. It would really be nice to have the backing of the mothership. On Mon, Aug 9, 2010 at 1:17 PM, Todd Hodgen wrote: > EX Expert is a great idea, allowing small sipXecs firms to continue to > promote SIPf

Re: [sipx-users] SIP Trunk Setup questions

yes. On Mon, Aug 9, 2010 at 1:33 PM, Ujjval Karihaloo wrote: > Thx, > > > > Is this what I do to get a SIP trace that will help? > > > > > http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer > > > > > > > > > > *From:* Tony Graziano [mailto:tgrazi...@myitdepartment.n

Re: [sipx-users] SIP Trunk Setup questions

Thx, Is this what I do to get a SIP trace that will help? http://sipx-wiki.calivia.com/index.php/Display_SIP_message_flow_using_Sipviewer From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, August 09, 2010 11:33 AM To: Ujjval Karihaloo Cc: Michael Scheidell; sipx-users@li

Re: [sipx-users] SIP Trunk Setup questions

Not sure what configurations have changed, so I would at least check the following - You state that you have sipxbridge set to operator for incoming calls. Do you have operator in the auto attendant? It defaults to being in there along with 0. In the sipxbridge, you can put any extensi

Re: [sipx-users] SIP Trunk Setup questions

I think a siptrace would be most useful first, I am not sure a full snapshot would be required at this time. I would send the trace to the list, a JIRA is premature. On Mon, Aug 9, 2010 at 1:23 PM, Ujjval Karihaloo wrote: > BTW, I am the ITSP and looking to test SipX as many of our Customers use

Re: [sipx-users] SIP Trunk Setup questions

BTW, I am the ITSP and looking to test SipX as many of our Customers use it. Should I create a Jira ticket or just send the snapshot to this list? From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, August 09, 2010 10:54 AM To: Ujjval Karihaloo Cc: Michael Scheidell; sipx-use

Re: [sipx-users] Awesome - We received a check today

EX Expert is a great idea, allowing small sipXecs firms to continue to promote SIPfoundry for accounts that fall outside of the eZuce niche, but offers the Tier 3 support required to be a reputable support firm. And, it grows the sipXecs name and reputation in the industry, while promoting SIPfoun

[sipx-users] Awesome - We received a check today

I wanted to thank Jim Canfield and Emstar for a great contribution to SIPfoundry, that we received today. This is the kind of collaboration we were hoping for and it is very encouraging and motivating to see the community getting engaged in this way. Along these lines, I'd like to circulate an

Re: [sipx-users] SIP Trunk Setup questions

You can also use a different Auto Attendant as a conference room Attendant, allowing them to dial the conference room they need to join, and giving them options to transfer to specific conference rooms. With this you can provide a specific voice recording with instructions pertaining to conference

Re: [sipx-users] SIP Trunk Setup questions

OK, I will do the snapshot, I have the logging levels at Debug from Diagnostic menu option on the Web UI.. I opened up the sipsproxy.log and saw only INFO log entries in there..though. From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, August 09, 2010 10:54 AM To: Ujjval Ka

Re: [sipx-users] SIP Trunk Setup questions

You are not providing much information. What is the UA? I see the ITSP is sending G722, is that for real? Who is the ITSP? I see they are sending to port 5080, which is good. Is the UA at a.b.c.161:5060? If so, what do the CDR logs show? I would want to see the sipproxy logs at debug for a call

Re: [sipx-users] SIP Trunk Setup questions

It maybe something else. Theoretically just matching a UserID shud work. I will make the user ID 4 digits later on. For now just want the SIPX to answer. I Know I may be doing something wrong, so please bear with me. To make things even simpler In the SBC Routes, I entered Operator in the Inco

Re: [sipx-users] SIP Trunk Setup questions

I don't understand. Your userID DOE NOT need to be identical. Example, user 200 has an ALIAS of 5625551000 Creating users with the same value of a DID really hampers you. Someone leaves the company and someone else needs to get their calls creates way too much work and quite a management headache

Re: [sipx-users] SIP Trunk Setup questions

you have 4 digit internal extensions? and, siptrace shows it going where? and are you sure the user answers? (before you try to fwd it to the conf bridge, make sure you got the right user) I think I would have a 'normal' 4 digit user, '1000' with an alias of 562*. also, watch 'job status' whe

Re: [sipx-users] SIP Trunk Setup questions

Thx, I am just trying to setup an inbound route to really anything right now to get it to work. I see the INVITE comin with 5625551000 in Req URI ; the user defined has the same userid...and I have call forwarding on that user to go to the Conf Bridge Extension 2000 - for a Conf I created... I

[sipx-users] track.sipfoundry.org up and down today

as I install balsamiq plugin and configure things. sorry for the late notice ___ sipx-users mailing list sipx-users@list.sipfoundry.org List Archive: http://list.sipfoundry.org/archive/sipx-users/

Re: [sipx-users] MOH not working

Il lunedì 09 agosto 2010 17:43:39 Tony Graziano ha scritto: > Does it work on internal calls (user 1, to user 2) or none at all? None at all. > MOH support depends on several things working properly. > > 1. Properly configured DNS, because the MOH uri is formed by the domain, so > SRV records nee

Re: [sipx-users] SIP Trunk Setup questions

On 8/9/10 11:37 AM, Tony Graziano wrote: OR you can assign the DID to a separate Auto Attendant and let people choose the conference they want (1 for sales conf, 2 for management conf, etc.). I set it up so that 'special' users were given their own conference 'rooms', with a two digit prefix(70

Re: [sipx-users] SIP Trunk Setup questions

Thx a lot. I will take a look From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, August 09, 2010 9:38 AM To: Ujjval Karihaloo Cc: thod...@verizon.net; sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] SIP Trunk Setup questions You need to configure a dial plan to send

Re: [sipx-users] MOH not working

Does it work on internal calls (user 1, to user 2) or none at all? MOH support depends on several things working properly. 1. Properly configured DNS, because the MOH uri is formed by the domain, so SRV records need to function properly. 2. Phone configuration support, the phones must have a MOH

Re: [sipx-users] SIP Trunk Setup questions

You need to configure a dial plan to send calls out to the PSTN. There is adequate detail in the wiki, search for DIAL PLAN. You cannot assign a DID number to a conf bridge directly, but you can create a user without a phone, assign it the DID and forward all calls in the sipxconfig gui to the CON

Re: [sipx-users] Question about removing DNS Forwarder from Sipx

Thanks Tony... I'm a "less than" basic Linux person... heh heh. Danny From: Tony Graziano [mailto:tgrazi...@myitdepartment.net] Sent: Monday, August 09, 2010 10:33 AM To: Danny Shay Cc: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Question about removing DNS Forwarder from Sipx This

Re: [sipx-users] MOH not working

Il lunedì 09 agosto 2010 16:58:07 Tony Graziano ha scritto: > If I recall that is just a cosmetic warning and spidermonkey is not needed > for the sipxecs FS implementation. You can safely ignore that, though it > should be noted that the implementation should remove that so the warnings > are supp

Re: [sipx-users] Question about removing DNS Forwarder from Sipx

This is a basic linux querstion: Edit the /etc/named.conf file and restart named service. look in the section called 'forwarders'. On Mon, Aug 9, 2010 at 11:23 AM, Danny Shay wrote: > What is the best method for reconfiguring DNS on a sipx server? > > I need to remove the secondary forwarding D

Re: [sipx-users] SIP Trunk Setup questions

Got it...its registered now. What is the best place to start for inbound and outbound routing... Inbound to a Conf Bridge for example. I want to call into the SIP trunk from PSTN and hit a Conf Bridge. Outbound from registered endpoints and send the call out the SIP trunk. Any pointers to Doc

[sipx-users] Question about removing DNS Forwarder from Sipx

What is the best method for reconfiguring DNS on a sipx server? I need to remove the secondary forwarding DNS server from Sipx (build 4.2.1-018932 2010-06-25T10:43:40 build33) which I originally installed from the 4.2.1 iso. I originally set up our internal domain controller as the primary forw

Re: [sipx-users] one more request to use mp3 and not wav

I've noted in XX-8504 that mp3 licensing does not affect anything since the mp3's are note being mastered and disctributed outside of an organization. Have you look at configuring mod_shout and then looking at changing the default voicemail file type? http://wiki.freeswitch.org/wiki/Mod_voicemail

Re: [sipx-users] Blocking SIP URI Calls from the innternet

what firewall are you using? 2010/8/7 Matt White > I've thought about it, but they also support remote workers on home cable > modems. > > I've been thinking of moving them to vpn's. > > -M > > >>> "Todd Hodgen" 08/07/10 5:49 PM >>> > > How about a temporary fix by blocking all 5060 traffic tha

Re: [sipx-users] MOH not working

Il lunedì 09 agosto 2010 16:55:29 Michal Bielicki ha scritto: > I doubt that they can influence moh. Thanks Michal, now I know that have to investigate elsewere. -- Claudio Succa PERTEL - Torino - Italy +39-011-19826800 http://www.pertel.it http://www.uniassist.it __

Re: [sipx-users] MOH not working

If I recall that is just a cosmetic warning and spidermonkey is not needed for the sipxecs FS implementation. You can safely ignore that, though it should be noted that the implementation should remove that so the warnings are suppressed or silenced. I would assume the same about the others. What

Re: [sipx-users] MOH not working

Hi Claudio, I am working on redoing the whole freeswitch package for 4.3. The errors you see are caused by those modules being in the modules.conf.xml file and not being found by freeswitch on startup. I doubt that they can influence moh. sipdermonkey is for javascript support lua is for lua sup

Re: [sipx-users] Launched a Voicemail Service on 4.2

I am glad I am not the only humorist on here. On Mon, Aug 9, 2010 at 10:40 AM, Josh M. Patten wrote: > That’s about as graceful as a rhino on ice skates. > > > > > > *From:* sipx-users-boun...@list.sipfoundry.org [mailto: > sipx-users-boun...@list.sipfoundry.org] *On Behalf Of *Michael Scheidell

[sipx-users] MOH not working

Hi, moh is no more working with last 4.2.1 and snom phones. Trying to solve the problem I browsed through the logs and saw some critical errors in freeswitch.log: *** 2010-08-09 16:16:10.630427 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswit

Re: [sipx-users] Launched a Voicemail Service on 4.2

That's about as graceful as a rhino on ice skates. From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Michael Scheidell Sent: Monday, August 09, 2010 7:57 AM To: sipx-users@list.sipfoundry.org Subject: Re: [sipx-users] Launched a Voicemail S

Re: [sipx-users] Polycom speed dial entries

How many users do you have on your system? If it's over about 150 or so then it's possible you're experiencing http://track.sipfoundry.org/browse/XX-8474 Try restarting the "Presence" service in sipXconfig. -Original Message- From: sipx-users-boun...@list.sipfoundry.org [mailto:sipx-us

Re: [sipx-users] one more request to use mp3 and not wav

The jira issue I created indicates it is not a legal issue to encode these as mp3's using mod_shout in FS. I still think it should be a configurable feature. Tony Graziano, Manager Telephone: 434.984.8430 Fax: 434.984.8431 Email: tgrazi...@myitdepartment.net LAN/Telep

Re: [sipx-users] Launched a Voicemail Service on 4.2

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <4c5ffb25.30...@secnap.com> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50062> Message-ID: Mike, Not sure if can add in the specific user extension, as the operator number i

Re: [sipx-users] one more request to use mp3 and not wav

Redhat as well as centos miss that feature in sox for legal reasons afair. But I can do that in freeswitch without problems, just currently working out how to translate the voicemail to lua and than make those settings a simple option. Am 09.08.2010 um 15:03 schrieb Joe Micciche: > -BEGIN P

Re: [sipx-users] one more request to use mp3 and not wav

-BEGIN PGP SIGNED MESSAGE- Hash: SHA1 yum install sox man sox You can pass a wav file for sox processing into mp3. As far as the code to grab the wav, convert it, and email the mp3 to the user, if you get that sorted let us all know! - -- ===

Re: [sipx-users] Launched a Voicemail Service on 4.2

On 8/9/10 7:42 AM, Abdul Mayat wrote: Find a more 'graceful' way of disabling the operator - XX-4856 put the users extension in as operator and if they hit 0 they just them selves back again. -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SECNAP Network Secu

Re: [sipx-users] sipXconfig on mobile device

I thought it would be pretty cool if end users can access the portal from mobile device. Would be useful for superadmin to be able to monitor things from anywhere. Will try another browser. Michael, I had to set up Opera to connect with html rather than default socket. Then it worked! sven

Re: [sipx-users] MWI on Cisco phones

Thanks Mike, it looks 8238 describe MWI works. I will test myself here and send patch in. Thanks, Sen From: Michael Scheidell [mailto:michael.scheid...@secnap.com] Sent: Monday, August 09, 2010 11:51 AM To: Douglas Hubler Cc: Sen Heng; sipx-users@list.sipfoundry.org Subject: Re: [sip

Re: [sipx-users] looking to change FROM line in voicemail

>From an old thread that discussed it: "Place the following file: http://code.sipfoundry.org/browse/~raw,r=17072/sipXecs/main/sipXivr/src/main/java/org/sipfoundry/voicemail/ EmailFormats.Properties

[sipx-users] Launched a Voicemail Service on 4.2

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50056> Message-ID: Hi All, Seen as I have been posting a few questions asking for help on the forum, I thought it may be helpful to others to s

Re: [sipx-users] looking to change FROM line in voicemail

On 8/9/10 7:05 AM, Abdul Mayat wrote: een able to customise the from address for voicemail notification (EmailFormats.properties) and alarms... I this what your trying to achieve? yes, where is that located? -- Michael Scheidell, CTO o: 561-999-5000 d: 561-948-2259 ISN: 1259*1300 > *| *SE

Re: [sipx-users] sipXconfig on mobile device

i can't even get it to log in (admin) to 4.2.0 with Opera on Windows Mobile device. ideally, there will be a specific users .mobi type interface. I can't see any possibility of trying to administer sipx with a mobile phone. The system admin had better ALWAYS have access to his laptop. -- Mi

Re: [sipx-users] SIP Trunk Setup questions

On 8/8/10 10:56 PM, Ujjval Karihaloo wrote: How do I register my SIPX with a ITSP. I do not see any way to put in a username and passwd...only address (FQDN). delete and start over. its TRICKY and if you don't do it EXACTLY right, the options never show up. as soon as you enter gateway ->a

[sipx-users] Updating localization package

Hi, I want to upgrade my localization package in my production system. I'm guessing if uploading it will reset the dial plan. If so, is there a smart upgrading procedure that will not reset the dial plan to defaults? Max -- Massimo Vignone UniMORE - Servizi Informatici - Reti e Sistemi Pho

Re: [sipx-users] sipXconfig on mobile device

On Mon, Aug 9, 2010 at 5:25 AM, Sven Evensen wrote: > Is there any setting or config so that sipXconfig will work on a mobile > device. I have tried on > > an Android device using Opera, sipXconfig comes up, I can log on, but the > problem is the main menu > > (Users, Devices, Features etc) are ja

Re: [sipx-users] Polycom speed dial entries

Did you try restarting presence server? 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: ht

Re: [sipx-users] SIP Trunk Setup questions

You need to make sure you choose sbc route/sipxbrige1 to get these options. 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] looking to change FROM line in voicemail

Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: 8bit Organization: SipXecs Forum In-Reply-To: <4c5dd9fc.7090...@secnap.com> X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <50047> Message-ID: Maybe i have misinterpreted the question, but in 4.2 I have been able to customise

Re: [sipx-users] MWI on Cisco phones

Bug already created. right after 4.2.0 came out and MWI stopped working on the Cisco's. hint: its in unified messaging, advanced. wasn't needed in 4.0.4. hint BUG: if you do this, and its a hunt group, everyone gets MIT light. and you should program to latest firmware anyway. 8-12. Three cisc

Re: [sipx-users] MWI on Cisco phones

Can I get email address for Matt please or if he could give us more details where I can put user ID into SEP profile, it would be great. I can test at here immediately. Thanks, Sen -Original Message- From: Douglas Hubler [mailto:dhub...@ezuce.com] Sent: Monday, August 09, 2010 11:09 A

Re: [sipx-users] Polycom speed dial entries

Sorry to resurrect this thread but we seem to have a similar problem. Most (if not all) of our users have recently lost the speed dials on their phones (Polycom 650's). I can see the users speed dials on the SipX config server but when I 'Save and Update Phones' they don't reappear on the handse

Re: [sipx-users] MWI on Cisco phones

On Mon, Aug 9, 2010 at 4:25 AM, Sen Heng wrote: > Has anyone got MWI working on Cisco phones please give me a shout. Matt Burdick told me at cluecon that you have to specify the user id in the MWI value, only I cannot find where you set that at the moment. If you find it, then let me know, we sh

[sipx-users] sipXconfig on mobile device

Hi, Is there any setting or config so that sipXconfig will work on a mobile device. I have tried on an Android device using Opera, sipXconfig comes up, I can log on, but the problem is the main menu (Users, Devices, Features etc) are javascript or something similar and do not show up on the

[sipx-users] MWI on Cisco phones

Hi Mike, I have tested latest Sipx with Cisco 7945 and 7970 phones. The Phone can register straightway. I compiled Sipx from source code and latest version is 4.3.0-018884. The vm working ok, the MWI works for xlite but not for Cisco phones. I have tried change firmware from 8.3(5) up to 8.5(4